Speech and Language Processing Guide
Speech and Language Processing Guide
Daniel Jurafsky
Stanford University
James H. Martin
University of Colorado at Boulder
2
Contents
I Large Language Models 1
1 Introduction 3
4 Logistic Regression 61
4.1 Machine learning and classification . . . . . . . . . . . . . . . . . 62
4.2 The sigmoid function . . . . . . . . . . . . . . . . . . . . . . . . 64
4.3 Classification with Logistic Regression . . . . . . . . . . . . . . . 65
4.4 Multinomial logistic regression . . . . . . . . . . . . . . . . . . . 69
4.5 Learning in Logistic Regression . . . . . . . . . . . . . . . . . . . 72
4.6 The cross-entropy loss function . . . . . . . . . . . . . . . . . . . 73
4.7 Gradient Descent . . . . . . . . . . . . . . . . . . . . . . . . . . 74
4.8 Learning in Multinomial Logistic Regression . . . . . . . . . . . . 80
4.9 Evaluation: Precision, Recall, F-measure . . . . . . . . . . . . . . 81
4.10 Test sets and Cross-validation . . . . . . . . . . . . . . . . . . . . 84
4.11 Statistical Significance Testing . . . . . . . . . . . . . . . . . . . 85
4.12 Avoiding Harms in Classification . . . . . . . . . . . . . . . . . . 88
4.13 Interpreting models . . . . . . . . . . . . . . . . . . . . . . . . . 89
4.14 Advanced: Regularization . . . . . . . . . . . . . . . . . . . . . . 90
4.15 Advanced: Deriving the Gradient Equation . . . . . . . . . . . . . 92
4.16 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Historical Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
3
4 C ONTENTS
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
5 Embeddings 95
5.1 Lexical Semantics . . . . . . . . . . . . . . . . . . . . . . . . . . 96
5.2 Vector Semantics: The Intuition . . . . . . . . . . . . . . . . . . . 98
5.3 Simple count-based embeddings . . . . . . . . . . . . . . . . . . 100
5.4 Cosine for measuring similarity . . . . . . . . . . . . . . . . . . . 102
5.5 Word2vec . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 104
5.6 Visualizing Embeddings . . . . . . . . . . . . . . . . . . . . . . . 110
5.7 Semantic properties of embeddings . . . . . . . . . . . . . . . . . 111
5.8 Bias and Embeddings . . . . . . . . . . . . . . . . . . . . . . . . 113
5.9 Evaluating Vector Models . . . . . . . . . . . . . . . . . . . . . . 114
5.10 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115
Historical Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 118
8 Transformers 171
8.1 Attention . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172
8.2 Transformer Blocks . . . . . . . . . . . . . . . . . . . . . . . . . 177
8.3 Parallelizing computation using a single matrix X . . . . . . . . . 181
8.4 The input: embeddings for token and position . . . . . . . . . . . 184
8.5 The Language Modeling Head . . . . . . . . . . . . . . . . . . . 186
8.6 More on Sampling . . . . . . . . . . . . . . . . . . . . . . . . . . 188
8.7 Training . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189
8.8 Dealing with Scale . . . . . . . . . . . . . . . . . . . . . . . . . . 190
8.9 Interpreting the Transformer . . . . . . . . . . . . . . . . . . . . . 193
8.10 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
Historical Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 195
16 Text-to-Speech 361
16.1 TTS overview . . . . . . . . . . . . . . . . . . . . . . . . . . . . 362
16.2 Using a codec to learn discrete audio tokens . . . . . . . . . . . . 363
16.3 VALL-E: Generating audio with 2-stage LM . . . . . . . . . . . . 369
16.4 TTS Evaluation . . . . . . . . . . . . . . . . . . . . . . . . . . . 371
16.5 Other speech tasks . . . . . . . . . . . . . . . . . . . . . . . . . . 372
16.6 Spoken Language Models . . . . . . . . . . . . . . . . . . . . . . 372
16.7 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 373
Historical Notes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 373
Exercises . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 374
Bibliography 577
Subject Index 607
Volume I
LARGE LANGUAGE MODELS
In the first part of the book we introduce the fundamental suite of algorithmic
and linguistic tools that make up the modern neural large language model. We begin
with tokenization and preprocessing, including Unicode, and then proceed to intro-
duce many basic language modeling ideas using simple n-gram language models, we
then introduce the algorithms which are the components of large language models:
logistic regression, embeddings, and feedforward networks. Next we are ready to
introduce the principles of large language modeling, encoder, decoders and pretrain-
ing, then the fundamental transformer architecture, then masked language model
and other architectures like RNNs and LSTMs, information retrieval and retrieval-
based algorithms like RAG, machine translation and the encoder-decoder model,
and finally spoken language modeling including both ASR and TTS.
CHAPTER
1 Introduction
La dernière chose qu’on trouve en faisant un ouvrage est de savoir celle qu’il faut
mettre la première.
[The last thing you figure out in writing a book is what to put first.]
Pascal
3
4 C HAPTER 2 • W ORDS AND T OKENS
CHAPTER
2.1 Words
How many words are in the following sentence?
They picnicked by the pool, then lay back on the grass and
looked at the stars.
This sentence has 16 words if we don’t count punctuation as words, 18 if we
count punctuation. Whether we treat period (“.”), comma (“,”), and so on as words
depends on the task. Punctuation is critical for finding boundaries of things (com-
mas, periods, colons) and for identifying some aspects of meaning (question marks,
exclamation marks, quotation marks). Large language models generally count punc-
tuation as separate words.
Spoken language introduces other complications with regard to defining words.
utterance What about this utterance from a spoken conversation? (Utterance is the technical
linguistic term for the spoken correlate of a sentence).
I do uh main- mainly business data processing
disfluency This utterance has two kinds of disfluencies. The broken-off word main- is
fragment called a fragment. Words like uh and um are called fillers or filled pauses. Should
filled pause we consider these to be words? Again, it depends on the application. If we are
building a speech transcription system, we might want to eventually strip out the
disfluencies. But we also sometimes keep disfluencies around. Disfluencies like uh
or um are actually helpful in speech recognition in predicting the upcoming word,
because they may signal that the speaker is restarting the clause or idea, and so for
speech recognition they are treated as regular words. Because different people use
different disfluencies they can also be a cue to speaker identification. In fact Clark
and Fox Tree (2002) showed that uh and um have different meanings in English.
What do you think they are?
Perhaps most important, in thinking about what is a word, we need to distinguish
word type two ways of talking about words that will be useful throughout the book. Word types
are the number of distinct words in a corpus; if the set of words in the vocabulary
word instance is V , the number of types is the vocabulary size |V |. Word instances are the total
number N of running words.1 If we ignore punctuation, the picnic sentence has 14
types and 16 instances:
They picnicked by the pool, then lay back on the grass and
looked at the stars.
We still have decisions to make! For example, should we consider a capitalized
string (like They) and one that is uncapitalized (like they) to be the same word type?
The answer is that it depends on the task! They and they might be lumped together as
the same type in some tasks where we care less about the formatting, while for other
tasks, capitalization is a useful feature and is retained. Sometimes we keep around
two versions of a particular NLP model, one with capitalization and one without
capitalization.
So far we have been talking about orthographic words: words based on our
English writing system. But there are many other possible ways to define words.
For example, while orthographically I’m is one word, grammatically it functions as
two words: the subject pronoun I and the verb ’m, short for am.
1 In earlier tradition, and occasionally still, you might see word instances referred to as word tokens, but
we now try to reserve the word token instead to mean the output of subword tokenization algorithms.
6 C HAPTER 2 • W ORDS AND T OKENS
The distinctions get even harder to make once we start to think about other lan-
guages. For example the writing systems of languages like Chinese, Japanese, and
Thai simply don’t have orthographic words at all! That is, they don’t use spaces to
mark potential word-boundaries. In Chinese, for example, words are composed of
hanzi characters (called hanzi in Chinese). Each character generally represents a single
unit of meaning (called a morpheme, introduced below) and is pronounceable as a
single syllable. Words are about 2.4 characters long on average. But since Chinese
has no orthographic words, deciding what counts as a word in Chinese is complex.
For example, consider the following sentence:
(2.1) 姚明进入总决赛 yáo mı́ng jı̀n rù zǒng jué sài
“Yao Ming reaches the finals”
As Chen et al. (2017b) point out, this could be treated as 3 words (a definition of
words called the ‘Chinese Treebank’ definition, in which Chinese names (family
name followed by personal names) are treated as a single word):
(2.2) 姚明 进入 总决赛
YaoMing reaches finals
But the same sentence could be treated as 5 words (‘Peking University’ standard),
in which names are separated into their own units and some adjectives appear as
distinct words:
(2.3) 姚 明 进入 总 决赛
Yao Ming reaches overall finals
Finally, it is possible in Chinese simply to ignore words altogether and use characters
as the basic elements, treating the sentence as a series of 7 characters, which works
pretty well for Chinese since characters are at a reasonable semantic level for most
applications (Li et al., 2019):
(2.4) 姚 明 进 入 总 决 赛
Yao Ming enter enter overall decision game
But that method doesn’t work for Japanese and Thai, where the individual character
is too small a unit.
These issues with defining words makes it hard to use words as the basis for
tokenizing text in NLP across languages.
But there’s another problem with words. There are too many of them!!! How
many words are there in English? When we speak about the number of words in
the language, we are generally referring to word types. Fig. 2.1 shows the rough
numbers of types and instances computed from some English corpora.
You will notice that the larger the corpora we look at, the more word types we
find! That suggests that there is not a clear answer to how many words there are;
the answer keeps growing as we see more data! We can see this fact mathematically
2.1 • W ORDS 7
because the relationship between the number of types |V | and number of instances
Herdan’s Law N is called Herdan’s Law (Herdan, 1960) or Heaps’ Law (Heaps, 1978) after its
Heaps’ Law discoverers (in linguistics and information retrieval respectively). It is shown in
Eq. 2.5, where k and β are positive constants, and 0 < β < 1.
|V | = kN β (2.5)
The value of β depends on the corpus size and the genre; numbers from 0.44 to 0.56
or even higher have often been reported. Roughly we can say that the vocabulary
size for a text goes up a little faster than the square root of its length in words.
There are also variants of the law, which capture the fact that we can distinguish
function words roughly two classes of words. One is function words, the grammatical words like
English a and of, that tend not to grow indefinitely (a language tends to have a fixed
content words number of these). The other is content words: nouns, adjectives and verbs that tend
to have meanings about people and places and events. Nouns, and especially partic-
ular nouns like names and technical terms do tend to grow indefinitely. So models
that are sensitive to this difference between function words and content words have
one value of β for the initial part of the corpus where all words are still appearing,
and then a second β afterwords for when only the content words are still appearing.
Fig. 2.2 shows an example from Tria et al. (2018) showing two values of β for Heaps
law computed on the Gutenberg corpus of books.
Entropy 2018, 20, 752 4 of 19
Figure 2.2
Figure Vocabulary
2. Growth size asofa distinct
of the number function of text
words length,oncomputed
computed on the
the Gutenberg Gutenberg
corpus corpus
of texts [15].
of publicly available
The position of texts books. Figure
in the corpus from Tria
is chosen et al. (2018).
at random. In this case g ' 0.44. Similar behaviours are
observed in many other systems.
The fact that words grow without end leads to a problem for any computational
2.3. Zipf’s vs. Heaps’ Laws
model. No matter how big our vocabulary, we will never have a vocabulary that
In this section
captures all thewe comparewords
possible the twothat
lawsmight
just observed,
occur! Zipf’s
That law for the
means frequencies
that of occurrence
our computational
of the elements in a system and Heaps’ law for their temporal appearance. It has often been claimed that
model will constantly see unknown words: words that it has never seen before.
Heaps’ and Zipf’s law are trivially related and that one can derive Heaps’s law once the Zipf’s is known.
This is a huge problem for machine learning models.
This is not true in general. It turns out to be true only under the specific hypothesis of random-sampling
Because
as follows. of these
Suppose two problems
the existence (first, that
of a strict power-law many languages
behaviour don’t have
of the frequency-rank ortho-
distribution,
f graphic
( R) ⇠ R words, and defining
a , and construct them
a sequence of post-hoc
elements byisrandomly
challenging and from
sampling second, that distribution
this Zipf the num-
f ber
( R). of wordsthis
Through grows without
procedure, onebound),
recovers alanguage
Heaps’ law models
with theand other NLP
functional form D models
(t) ⇠ tg don’t
[23,24]
tend to use words as their unit of processing. Instead, they use smaller units
with g = 1/a. In order to do that we need to consider the correct expression for f ( R ) that called
includes the
normalisation factor, whose expression can be derived through
subwords that can be recombined to model new words that our model has never the following approximated integral:
seen before. To think about defining Z Rmaxsubwords, we first need to talk about units that
f ( R̃)d R̃ = 1 . (3)
are smaller than words; morphemes 1 and characters.
1 a a
f ( R) = R . (4)
R1maxa 1
The first dimension is the number of morphemes per word. In some languages,
like Vietnamese and Cantonese, each word on average has just over one morpheme.
isolating We call languages at this end of the scale isolating languages. For example each
word in the following Cantonese sentence has one morpheme (and one syllable):
(2.8) keoi5 waa6 cyun4 gwok3 zeoi3 daai6 gaan1 uk1 hai6 ni1 gaan1
he say entire country most big building house is this building
“He said the biggest house in the country was this one”
Alternatively, in languages like Koryak, a Chukotko-Kamchatkan language spo-
ken in the northern part of the Kamchatka peninsula in Russia, a single word may
have very many morphemes, corresponding to a whole sentence in English (Arkadiev,
synthetic 2020; Kurebito, 2017). We call languages toward this end of the scale synthetic lan-
polysynthetic guages, and the very end of the scale polysynthetic languages.
(2.9) t-@-nk’e-mejN-@-jetem@-nni-k
[Link]-E-sew-1SG.S[PFV]
“I sewed a lot of yurt covers in the middle of a night.”
(Koryak, Chukotko-Kamchatkan, Russia; Kurebito (2017, 844))
Fig. 2.3 shows an early computation of morphemes per words on a few languages
by the linguistic typologist Joseph Greenberg (1960).
e sh ic
es gli t a nd
am si ish n
E t ili kri l
n )
et
n r gl l d aku wah ans ee uit
Vi Fa En O Y S S Gr (In
1.1 1.5 1.7 2.1 2.2 2.5 2.6 3.7
Figure 2.3 An early estimate of morphemes per word by Joseph Greenberg (1960).
The second dimension is the degree to which morphemes are easily segmentable,
agglutinative ranging from agglutinative languages like Turkish, in which morphemes have rel-
fusion atively clean boundaries, to fusion languages like Russian, in which a single affix
may conflate multiple morphemes, like -om in the word stolom (table-SG-INSTR-
DECL 1), which fuses the distinct morphological categories instrumental, singular,
and first declension.
The English -s suffix in She reads the article is an example of fusion, since the
suffix means both third person singular but also means present tense, and there’s no
way to divide up the meaning to different parts of the -s.
Although we have loosely talked about these properties (analytic, polysynthetic,
fusional, agglutinative) as if they are properties of languages, in fact languages can
make use of different morphological systems so it would be more accurate to talk
about these as general tendencies.
Nonetheless, the fact morphemes can be hard to define, and that many languages
can have complex morphemes that aren’t easy to break up into pieces makes it very
difficult to use morphemes as a standard for tokenization cross-lingually.
10 C HAPTER 2 • W ORDS AND T OKENS
2.3 Unicode
Another option we could consider for tokenization is the level of the individual char-
acter. How do we even represent characters across languages and writing system?
Unicode The Unicode standard is a method for representing text written using any character
in any script of the languages of the world (including dead languages like Sumerian
cuneiform, and invented languages like Klingon).
Let’s start with a brief historical note about an English-specific subset of Unicode
(technically called ‘Basic Latin’ in Unicode, and commonly referred to as ASCII).
Starting in the 1960s, the Latin characters used to write English (like the ones used
ASCII in this sentence), were represented with a code called ASCII (American Standard
Code for Information Interchange). ASCII represented each character with a single
byte. A byte can represent 256 different characters, but ASCII only used 127 of
them; the high-order bit of ASCII bytes is always set to 0. (Actually it only used 95
of them and the rest were control codes for an obsolete machine called a teletype).
Here’s a few ASCII characters with their representation in hex and decimal:
But ASCII is of course insufficient since there are lots of other characters in the
world’s writing systems! Even for scripts that use Latin characters, there are many
more than the 95 in ASCII. For example, this Spanish phrase (meaning “Sir, replied
Sancho”) has two non-ASCII characters, ñ and ó:
(2.10) Señor- respondió Sancho-
Devanagari And lots of languages aren’t based on Latin characters at all! The Devanagari
script is used for 120 languages (including Hindi, Marathi, Nepali, Sindhi, and San-
skrit). Here’s a Devanagari example from the Hindi text of the Universal Declaration
of Human Rights:
00 00 00 68 00 00 00 65 00 00 00 6C 00 00 00 6C 00 00 00 6F
But we don’t use this encoding (which is technically called UTF-32) because it
makes every file 4 times longer than it would have been in ASCII, making files really
big and full of zeros. Also those zeros cause another problem: it turns out that having
any byte that is completely zero messes things up for backwards compatibility for
ASCII-based systems that historically used a 0 byte as an end-of-string marker.
UTF-8 Instead, the most common encoding standard is UTF-8 (Unicode Transforma-
tion Format 8), which represents characters efficiently (using fewer bytes on av-
erage) by writing some characters using fewer bytes and some using more bytes.
variable-length
encoding UTF-8 is thus a variable-length encoding.
For some characters (the first 127 code points, i.e. the set of ASCII characters),
UTF-8 encodes them as a single byte, so the UTF-8 encoding of hello is :
68 65 6C 6C 6F
This conveniently means that files encoded in ASCII are also valid UTF-8 en-
codings!
But UTF-8 is a variable length encoding, meaning that code points ≥128 are
encoded as a sequence of two, three, or four bytes. Each of these bytes are between
128 and 255, so they won’t be confused with ASCII, and each byte indicates in the
first few bits whether it’s a 2-byte, 3-byte, or 4-byte encoding.
Code Points UTF-8 Encoding
From - To Bit Value Byte 1 Byte 2 Byte 3 Byte 4
U+0000-U+007F 0xxxxxxx xxxxxxxx
U+0080-U+07FF 00000yyy yyxxxxxx 110yyyyy 10xxxxxx
U+0800-U+FFFF zzzzyyyy yyxxxxxx 1110zzzz 10yyyyyy 10xxxxxx
U+010000-U+10FFFF 000uuuuu zzzzyyyy yyxxxxxx 11110uuu 10uuzzzz 10yyyyyy 10xxxxxx
Figure 2.5 Mapping from Unicode code point to the variable length UTF-8 encoding. For a given code point
in the From-To range, the bit value in column 2 is packed into 1, 2, 3, or 4 bytes. Figure adapted from Unicode
16.0 Core Spec Chapter 3 Table 3-6.
Fig. 2.5 shows how this mapping occurs. For example these rules explain how
the character ñ, which has code point U+00F1, or bit sequence 00000000 11110001,
(where blue indicates the sequence yyyyy and red the sequence xxxxxx) is encoded
into to the two-byte bit sequence 11000011 10110001 or 0xC3B1. As a result of
these rules, the first 127 characters (ASCII) are mapped to one byte, most remain-
ing characters in European, Middle Eastern, and African scripts map to two bytes,
most Chinese, Japanese, and Korean characters map to three bytes, and rarer CJKV
characters and emojis and some symbols map to 4 bytes.
UTF-8 has a number of advantages. It’s relatively efficient, using fewer bytes for
commonly-encountered characters, it doesn’t use zero bytes (except when literally
representing the NULL character which is U+0000), it’s backwards compatible with
ASCII, and it’s self-synchronizing, meaning that if a file is corrupted, it’s always
possible to find the start of the next or prior character just by moving up to 3 bytes
left or right.
Unicode and Python: Starting with Python 3, all Python strings are stored in-
ternally as Unicode, each string a sequence of Unicode code points. Thus string
functions and regular expressions all apply natively to code points. For example,
functions like len() of a string return its length in characters, i.e., code points, not
its length in bytes.
When reading or writing from a file, however, the code points need to be encoded
and decoding using a method like UTF-8. That is, every file is encoded in some
2.4 • S UBWORD T OKENIZATION : B YTE -PAIR E NCODING 13
encoding. If it’s not UTF-8, it’s an older encoding method like ASCII or Latin-1
(iso 8859 1). There is no such thing as a text file without an encoding. The encoding
method is specified in Python when opening a file for reading and writing.
2 The SentencePiece library includes implementations of both of these (Kudo and Richardson, 2018a),
and people sometimes use the name SentencePiece to simply mean ULM tokenization.
14 C HAPTER 2 • W ORDS AND T OKENS
(usually roughly pre-separated into words, for example by whitespace) and induce
a vocabulary, a set of tokens. Then a token encoder take a raw test sentence and
encodes it into the tokens in the vocabulary that were learned in training.
The BPE training algorithm first counts all pairs of adjacent symbols: the most
frequent is the pair n e because it occurs in new (frequency of 2) and renew (fre-
quency of 2) for a total of 4 occurrences. We then merge these symbols, treating ne
as one symbol, and count again:
corpus vocabulary
2 ne w , e, n, r, s, t, w, ne
2 r e ne w
1 s e t
1 r e s e t
Now the most frequent pair is ne w (total count=4), which we merge.
corpus vocabulary
2 new , e, n, r, s, t, w, ne, new
2 r e new
1 s e t
1 r e s e t
Next r (total count of 3) get merged to r, and then r e (total count 3) gets
merged to re. The system has essentially induced that there is a word-initial prefix
re-:
corpus vocabulary
2 new , e, n, r, s, t, w, ne, new, r, re
2 re new
1 s e t
1 re s e t
If we continue, the next merges are:
merge current vocabulary
( , new) , e, n, r, s, t, w, ne, new, r, re, new
( re, new) , e, n, r, s, t, w, ne, new, r, re, new, renew
(s, e) , e, n, r, s, t, w, ne, new, r, re, new, renew, se
(se, t) , e, n, r, s, t, w, ne, new, r, re, new, renew, se, set
Figure 2.6 The training part of the BPE algorithm for taking a corpus broken up into in-
dividual characters or bytes, and learning a vocabulary by iteratively merging tokens. Figure
adapted from Bostrom and Durrett (2020).
the training data. It runs them greedily, in the order we learned them. (Thus the
frequencies in the test data don’t play a role, just the frequencies in the training
data). So first we segment each test sentence word into characters. Then we apply
the first rule: replace every instance of n e in the test corpus with ne, and then the
second rule: replace every instance of ne w in the test corpus with new, and so on.
By the end of course many of the merges simple recreated words in the training
set. But the merges also created knowledge of morphemes like the re- prefix (that
might appear in perhaps unseen combinations like revisit or rearrange), or the
morpheme new without an initial space (hence word-internal) that might appear at
the start of sentences or in words unseen in training like anew.
Of course in real settings BPE is run with tens of thousands of merges on a very
large input corpus, to produce vocabulary sizes of 50,000, 100,000, or even 200,000
tokens. The result is that most words can be represented as single tokens, and only
the rarer words (and unknown words) will have to be represented by multiple tokens.
At least for English. For multilingual systems, the tokens can be dominated by
English, leaving fewer tokens for other languages, as we’ll discuss below.
The visualization shows colors to separate out words, but of course the true out-
put of the tokenizer is simply a sequence of unique token ids. (In case you’re in-
terested, they were the following 13 tokens: 11865, 8923, 11, 31211, 6177, 23919,
885, 220, 19427, 7633, 18887, 147065, 0)
Notice that most words are their own token, usually including the leading space.
Clitics like ’s are segmented off when they appear on proper nouns like Jane, but
are counted as part of a word for frequent words like she’s. Numbers tend to be
segmented into chunks of 3 digits. And some words (like anyhow) are segmented
differently if they appear capitalized sentence-initially (two tokens, Any and how),
then if they appear after a space, lower case (one token anyhow).
Some of these are related to preprocessing steps. As we mentioned briefly above,
pretokenization language models usually create their tokens in a pretokenization stage that first seg-
ments the input using regular expressions, for example breaking the input at spaces
and punctuation, stripping off clitics, and breaking numbers into sets of 3 digits.
2.5 • RULE - BASED TOKENIZATION 17
Figure 2.7 The SuperBPE algorithm creating larger tokens by allowing a second stage of
merging across spaces. Figure from Liu et al. (2025).
Many of the tokenizers used in practice for large language models are multilin-
gual, trained on many languages. But because the training data for large language
models is vastly dominated by English text, these multilingual BPE tokenizers tend
to use most of the tokens for English, leaving fewer of them for other languages. The
result is that they do a better job of tokenizing English, and the other languages tend
to get their words split up into shorter tokens. For example let’s look at a Spanish
sentence from a recipe for plantains, together with an English translation.
The English has 18 tokens; each of the 14 words is a token (none of the words
are split into Figure
multiple tokens):
1: SuperBPE tokenizers encode text much more efficiently than BPE, and the
gap grows with larger vocabulary size. Encoding efficiency (y-axis) is measured with
bytes-per-token, the number of bytes encoded per token on average over a large corpus of text.
In the above text with 40 bytes, SuperBPE uses 7 tokens and BPE uses 13, so the methods’
efficiencies are 40/7 = 5.7 and 40/13 = 3.1 bytes-per-token, respectively. In the graph,
the encoding efficiency of BPE plateaus early due to exhausting the valuable whitespace-
delimited words in the training data. In fact, it is bounded above by the gray dotted line,
which shows the maximum achievable encoding efficiency with BPE, if every whitespace-
By contrast, the original
delimited word were 16 words
in the in Spanish
vocabulary. have
On the other been
hand, encoded
SuperBPE into 33 tokens,
has dramatically
a much largerbetter
number.
encodingNotice
efficiencythat
that many basic
continues words
to improve have
with been
increased broken
vocabulary
it can continue to add common word sequences to treat as tokens to the vocabulary. The
intoas pieces.
size,
stream processing
Notably, of the language.
SuperBPE Asmuch
tokenizers scale willbetter
become more size—while
with vocabulary clear once BPEwe introduce
quickly
hits a point of diminishing returns and begins adding increasingly rare subwords to the
transformer models
vocabulary,in Chapter
SuperBPE can 8, suchtofragmentation
continue discover common word cansequences
lead to poor
to treat representa-
as single
tokens and improve encoding efficiency (see Figure 1).
tions of meaning, the need for longer contexts, and higher costs to train models
In our main experiments, we pretrain English LMs at 8B scale from scratch. When fixing the
(Rust et al., 2021; Ahia et al.,size,
model size, vocabulary 2023).
and training compute—varying only the algorithm for learning
the vocabulary—we find that models trained with SuperBPE tokenizers consistently and
significantly improve over counterparts trained with a BPE tokenizer, while also being 27–
33% more efficient at inference time. Our best SuperBPE model achieves an average +4.0%
While data-based tokenization like BPE is the most common way of doing tokeniza-
tion, there are also situations where we want to constrain our tokens to be words and
not subwords. This might be useful if we are running parsing algorithms for English
where the parser might need grammatical words as input. Or it can be useful for
any linguistic application where we have some a prior definition of the token that we
18 C HAPTER 2 • W ORDS AND T OKENS
are interested in studying. Or it can be useful for social science applications where
orthographic words are useful domains of study.
In rule-based tokenization, we pre-define a standard and implement rules to im-
plement that kind of tokenization. Let’s explore this for English word tokenization.
We have some desiderata for English. We often want to break off punctua-
tion as a separate token; commas are a useful piece of information for parsers,
and periods help indicate sentence boundaries. But we’ll often want to keep the
punctuation that occurs word internally, in examples like m.p.h., Ph.D., AT&T, and
cap’n. Special characters and numbers will need to be kept in prices ($45.55) and
dates (01/02/06); we don’t want to segment that price into separate tokens of “45”
and “55”. And there are URLs ([Link] Twitter hashtags
(#nlproc), or email addresses (someone@[Link]).
Number expressions introduce complications; in addition to appearing at word
boundaries, commas appear inside numbers in English, every three digits: 555,500.50.
Tokenization differs by language; languages like Spanish, French, and German, for
example, use a comma to mark the decimal point, and spaces (or sometimes periods)
where English puts commas, for example, 555 500,50.
clitic A rule-based tokenizer can also be used to expand clitic contractions that are
marked by apostrophes, converting what’re to the two tokens what are, and we’re
to we are. A clitic is a part of a word that can’t stand on its own, and can only oc-
cur when it is attached to another word. Such contractions occur in other alphabetic
languages, including French pronouns (j’ai and articles l’homme).
Depending on the application, tokenization algorithms may also tokenize mul-
tiword expressions like New York or rock ’n’ roll as a single token, which re-
quires a multiword expression dictionary of some sort. Rule-based tokenization is
thus intimately tied up with named entity recognition, the task of detecting names,
dates, and organizations (Chapter 17).
One commonly used tokenization standard is known as the Penn Treebank to-
Penn Treebank kenization standard, used for the parsed corpora (treebanks) released by the Lin-
tokenization
guistic Data Consortium (LDC), the source of many useful datasets. This standard
separates out clitics (doesn’t becomes does plus n’t), keeps hyphenated words to-
gether, and separates out all punctuation (to save space we’re showing visible spaces
‘ ’ between tokens, although newlines is a more common output):
2.6 Corpora
Words don’t appear out of nowhere. Any particular piece of text that we study
is produced by one or more specific speakers or writers, in a specific dialect of a
specific language, at a specific time, in a specific place, for a specific function.
20 C HAPTER 2 • W ORDS AND T OKENS
Perhaps the most important dimension of variation is the language. NLP algo-
rithms are most useful when they apply across many languages. The world has 7097
languages at the time of this writing, according to the online Ethnologue catalog
(Simons and Fennig, 2018). It is important to test algorithms on more than one lan-
guage, and particularly on languages with different properties; by contrast there is
an unfortunate current tendency for NLP algorithms to be developed or tested just on
English (Bender, 2019). Even when algorithms are developed beyond English, they
tend to be developed for the official languages of large industrialized nations (Chi-
nese, Spanish, Japanese, German etc.), but we don’t want to limit tools to just these
few languages. Furthermore, most languages also have multiple varieties, often spo-
ken in different regions or by different social groups. Thus, for example, if we’re
AAE processing text that uses features of African American English (AAE) or African
American Vernacular English (AAVE)—the variations of English that can be used
by millions of people in African American communities (King 2020)—we must use
NLP tools that function with features of those varieties. Twitter posts might use fea-
tures often used by speakers of African American English, such as constructions like
MAE iont (I don’t in Mainstream American English (MAE)), or talmbout corresponding
to MAE talking about, both examples that influence word segmentation (Blodgett
et al. 2016, Jones 2015).
It’s also quite common for speakers or writers to use multiple languages in a sin-
code switching gle utterance, a phenomenon called code switching. Code switching is enormously
common across the world; here are examples showing Spanish and (transliterated)
Hindi code switching with English (Solorio et al. 2014, Jurgens et al. 2017):
(2.12) Por primera vez veo a @username actually being hateful! it was beautiful:)
[For the first time I get to see @username actually being hateful! it was
beautiful:) ]
(2.13) dost tha or ra- hega ... dont wory ... but dherya rakhe
[“he was and will remain a friend ... don’t worry ... but have faith”]
Another dimension of variation is the genre. The text that our algorithms must
process might come from newswire, fiction or non-fiction books, scientific articles,
Wikipedia, or religious texts. It might come from spoken genres like telephone
conversations, business meetings, police body-worn cameras, medical interviews,
or transcripts of television shows or movies. It might come from work situations
like doctors’ notes, legal text, or parliamentary or congressional proceedings.
Text also reflects the demographic characteristics of the writer (or speaker): their
age, gender, race, socioeconomic class can all influence the linguistic properties of
the text we are processing.
And finally, time matters too. Language changes over time, and for some lan-
guages we have good corpora of texts from different historical periods.
Because language is so situated, when developing computational models for lan-
guage processing from a corpus, it’s important to consider who produced the lan-
guage, in what context, for what purpose. How can a user of a dataset know all these
datasheet details? The best way is for the corpus creator to build a datasheet (Gebru et al.,
2020) or data statement (Bender et al., 2021) for each corpus. A datasheet specifies
properties of a dataset like:
Motivation: Why was the corpus collected, by whom, and who funded it?
Situation: When and in what situation was the text written/spoken? For example,
was there a task? Was the language originally spoken conversation, edited
text, social media communication, monologue vs. dialogue?
2.7 • R EGULAR E XPRESSIONS 21
Language variety: What language (including dialect/region) was the corpus in?
Speaker demographics: What was, e.g., the age or gender of the text’s authors?
Collection process: How big is the data? If it is a subsample how was it sampled?
Was the data collected with consent? How was the data pre-processed, and
what metadata is available?
Annotation process: What are the annotations, what are the demographics of the
annotators, how were they trained, how was the data annotated?
Distribution: Are there copyright or other intellectual property restrictions?
The regular expression r"[1234567890]" specifies any single digit. This can
get awkward (imagine typing r"[ABCDEFGHIJKLMNOPQRSTUVWXYZ]" to mean an
uppercase letter) so the brackets can also be used with a dash (-) to specify any one
range character in a range. The pattern r"[2-5]" specifies any one of the characters 2, 3,
4, or 5. The pattern r"[b-g]" specifies one of the characters b, c, d, e, f, or g. Some
other examples are shown in Fig. 2.10.
The square braces can also be used to specify what a single character cannot be,
by use of the caret ˆ. If the caret ˆ is the first symbol after the open square brace
[, the resulting pattern is negated. For example, the pattern r"[ˆa]" matches any
single character (including special characters) except a. This is only true when the
caret is the first symbol after the open square brace. If it occurs anywhere else, it
usually stands for a caret; Fig. 2.11 shows some examples.
a. That’s because Kleene star means “zero or more occurrences”. Instead, for the
sheep language we’ll want r"baaa*", meaning b followed by aa followed by zero
or more additional as. More complex patterns can also be repeated. So r"[ab]*"
means “zero or more a’s or b’s” (not “zero or more right square braces”). This will
match strings like aaaa or ababab or bbbb, as well as the empty string. For speci-
fying an integer (a string of digits) we can use r"[0-9][0-9]*". (Why isn’t it just
r"[0-9]*"?)
There is a slightly shorter way to specify “at least one” of some character: the
Kleene + Kleene +, which means “one or more occurrences of the immediately preceding
character or regular expression”. So r"[0-9]+" is the normal way to specify “a
sequence of digits”, and we could also specify the sheep language as r"baa+!".
Besides the Kleene * and Kleene + we can also use explicit numbers as coun-
ters, by enclosing them in curly brackets. The operator r"{3}" means “exactly 3
occurrences of the previous character or expression”. So r"ax{10}z" will match a
followed by exactly 10 x’s followed by z.
period An important special character is the period (r"."), a wildcard expression that
matches any single character (except a newline).
The wildcard is often used together with the Kleene star to mean “any string
of characters”. For example, suppose we want to find any line in which a particu-
lar word, for example, rose, appears twice. We can specify this with the regular
expression r"rose.*rose", meaning two roses, with a sequence of zero or more
characters (of any kind) between them. Fig. 2.12 summarizes.
Regex Match
* zero or more occurrences of the previous char or expression
+ one or more occurrences of the previous char or expression
? zero or one occurrence of the previous char or expression
{n} exactly n occurrences of the previous char or expression
. any single char
.* any string of zero or more chars
Figure 2.12 Counting and wildcards.
Regex Match
ˆ start of line
$ end of line
\b word boundary
\B non-word boundary
Figure 2.13 Anchors in regular expressions.
Parenthesis ()
Counters * + ? {}
Sequences and anchors the ˆmy end$
Disjunction |
r"the" (2.14)
One problem is that this pattern will miss the word when it begins a sentence and
hence is capitalized (i.e., The). This might lead us to the following pattern:
r"[tT]he" (2.15)
But we will still overgeneralize, incorrectly return texts with the embedded in other
words (e.g., other or there). So we need to specify that we want instances with a
word boundary on both sides:
r"\b[tT]he\b" (2.16)
The simple process we just went through was based on fixing two kinds of errors:
false positives false positives, strings that we incorrectly matched like other or there, and false
false negatives negatives, strings that we incorrectly missed, like The. Addressing these two kinds
of errors comes up again and again in language processing. Reducing the overall
error rate for an application thus involves two antagonistic efforts:
• Increasing precision (minimizing false positives)
• Increasing recall (minimizing false negatives)
We’ll come back to precision and recall with more precise definitions in Chapter 4.
r"(?:\d\d/\d\d/\d\d\d\d\s+){14}(\d\d/\d\d/\d\d\d\d)" (2.17)
Substitutions and capture groups are also useful for implementing historically
important chatbots like ELIZA (Weizenbaum, 1966). Recall that ELIZA simulates
a Rogerian psychologist by carrying on conversations like the following:
User2 : They’re always bugging us about something or other.
ELIZA2 : CAN YOU THINK OF A SPECIFIC EXAMPLE
User3 : Well, my boyfriend made me come here.
ELIZA3 : YOUR BOYFRIEND MADE YOU COME HERE
User4 : He says I’m depressed much of the time.
ELIZA4 : I AM SORRY TO HEAR YOU ARE DEPRESSED
r"ˆ(?![tT])(\w+)\b" (2.18)
The first negative lookahead says that the line must not start with a t or T, but
matches the empty string, not moving the match pointer. Then the capture group
captures the first word.
Now that there is one word per line, we can sort the lines, and pass them to uniq
-c which will collapse and count them:
tr -sc ’A-Za-z’ ’\n’ < [Link] | sort | uniq -c
with the following output:
1945 A
72 AARON
19 ABBESS
25 Aaron
6 Abate
1 Abates
...
Alternatively, we can collapse all the upper case to lower case:
tr -sc ’A-Za-z’ ’\n’ < [Link] | tr A-Z a-z | sort | uniq -c
whose output is
14725 a
97 aaron
1 abaissiez
10 abandon
2 abandoned
2 abase
1 abash
14 abate
...
Now we can sort again to find the frequent words. The -n option to sort means
to sort numerically rather than alphabetically, and the -r option means to sort in
reverse order (highest-to-lowest):
tr -sc ’A-Za-z’ ’\n’ < [Link] | tr A-Z a-z | sort | uniq -c | sort -n -r
The results show that the most frequent words in Shakespeare, as in any other
corpus, are the short function words like articles, pronouns, prepositions:
27378 the
26084 and
22538 i
19771 to
17481 of
14725 a
13826 you
...
Unix tools of this sort can be very handy in building quick word count statistics
for any corpus in English. For anything more complex, we generally turn to the
more sophisticated tokenization algorithms we’ve discussed above.
recognition or machine translation, where we want to know how similar the sequence
of words is to some reference sequence of words.
Edit distance gives us a way to quantify these intuitions about string similarity.
minimum edit More formally, the minimum edit distance between two strings is defined as the
distance
minimum number of editing operations (operations like insertion, deletion, substitu-
tion) needed to transform one string into another. In this section we’ll introduce edit
distance for single words, but the algorithm applies equally to entire strings.
The gap between intention and execution, for example, is 5 (delete an i, substi-
tute e for n, substitute x for t, insert c, substitute u for n). It’s much easier to see
alignment this by looking at the most important visualization for string distances, an alignment
between the two strings, shown in Fig. 2.16. Given two sequences, an alignment is
a correspondence between substrings of the two sequences. Thus, we say I aligns
with the empty string, N with E, and so on. Beneath the aligned strings is another
representation; a series of symbols expressing an operation list for converting the
top string into the bottom string: d for deletion, s for substitution, i for insertion.
INTE*NTION
| | | | | | | | | |
*EXECUTION
d s s i s
Figure 2.16 Representing the minimum edit distance between two strings as an alignment.
The final row gives the operation list for converting the top string into the bottom string: d for
deletion, s for substitution, i for insertion.
We can also assign a particular cost or weight to each of these operations. The
Levenshtein distance between two sequences is the simplest weighting factor in
which each of the three operations has a cost of 1 (Levenshtein, 1966)—we assume
that the substitution of a letter for itself, for example, t for t, has zero cost. The Lev-
enshtein distance between intention and execution is 5. Levenshtein also proposed
an alternative version of his metric in which each insertion or deletion has a cost of
1 and substitutions are not allowed. (This is equivalent to allowing substitution, but
giving each substitution a cost of 2 since any substitution can be represented by one
insertion and one deletion). Using this version, the Levenshtein distance between
intention and execution is 8.
i n t e n t i o n
n t e n t i o n i n t e c n t i o n i n x e n t i o n
Figure 2.17 Finding the edit distance viewed as a search problem
The space of all possible edits is enormous, so we can’t search naively. However,
lots of distinct edit paths will end up in the same state (string), so rather than recom-
2.9 • M INIMUM E DIT D ISTANCE 31
puting all those paths, we could just remember the shortest path to a state each time
dynamic
programming we saw it. We can do this by using dynamic programming. Dynamic programming
is the name for a class of algorithms, first introduced by Bellman (1957), that apply
a table-driven method to solve problems by combining solutions to subproblems.
Some of the most commonly used algorithms in natural language processing make
use of dynamic programming, such as the Viterbi algorithm (Chapter 17) and the
CKY algorithm for parsing (Chapter 18).
The intuition of a dynamic programming problem is that a large problem can
be solved by properly combining the solutions to various subproblems. Consider
the shortest path of transformed words that represents the minimum edit distance
between the strings intention and execution shown in Fig. 2.18.
i n t e n t i o n
delete i
n t e n t i o n
substitute n by e
e t e n t i o n
substitute t by x
e x e n t i o n
insert u
e x e n u t i o n
substitute n by c
e x e c u t i o n
Figure 2.18 Path from intention to execution.
Imagine some string (perhaps it is exention) that is in this optimal path (whatever
it is). The intuition of dynamic programming is that if exention is in the optimal
operation list, then the optimal sequence must also include the optimal path from
intention to exention. Why? If there were a shorter path from intention to exention,
then we could use it instead, resulting in a shorter overall path, and the optimal
minimum edit
sequence wouldn’t be optimal, thus leading to a contradiction.
distance The minimum edit distance algorithm was named by Wagner and Fischer
algorithm
(1974) but independently discovered by many people (see the Historical Notes sec-
tion of Chapter 17).
Let’s first define the minimum edit distance between two strings. Given two
strings, the source string X of length n, and target string Y of length m, we’ll define
D[i, j] as the edit distance between X[1..i] and Y [1.. j], i.e., the first i characters of X
and the first j characters of Y . The edit distance between X and Y is thus D[n, m].
We’ll use dynamic programming to compute D[n, m] bottom up, combining so-
lutions to subproblems. In the base case, with a source substring of length i but an
empty target string, going from i characters to 0 requires i deletes. With a target
substring of length j but an empty source going from 0 characters to j characters
requires j inserts. Having computed D[i, j] for small i, j we then compute larger
D[i, j] based on previously computed smaller values. The value of D[i, j] is com-
puted by taking the minimum of the three possible paths through the matrix which
arrive there:
D[i − 1, j] + del-cost(source[i])
D[i, j] = min D[i, j − 1] + ins-cost(target[ j]) (2.19)
D[i − 1, j − 1] + sub-cost(source[i], target[ j])
We mentioned above two versions of Levenshtein distance, one in which substitu-
tions cost 1 and one in which substitutions cost 2 (i.e., are equivalent to an insertion
plus a deletion). Let’s here use that second version of Levenshtein distance in which
32 C HAPTER 2 • W ORDS AND T OKENS
the insertions and deletions each have a cost of 1 (ins-cost(·) = del-cost(·) = 1), and
substitutions have a cost of 2 (except substitution of identical letters has zero cost).
Under this version of Levenshtein, the computation for D[i, j] becomes:
D[i − 1, j] + 1
D[i, j − 1] + 1
D[i, j] = min (2.20)
2; if source[i] 6= target[ j]
D[i − 1, j − 1] +
0; if source[i] = target[ j]
The algorithm is summarized in Fig. 2.19; Fig. 2.20 shows the results of applying
the algorithm to the distance between intention and execution with the version of
Levenshtein in Eq. 2.20.
n ← L ENGTH(source)
m ← L ENGTH(target)
Create a distance matrix D[n+1,m+1]
# Initialization: the zeroth row and column is the distance from the empty string
D[0,0] = 0
for each row i from 1 to n do
D[i,0] ← D[i-1,0] + del-cost(source[i])
for each column j from 1 to m do
D[0,j] ← D[0, j-1] + ins-cost(target[j])
# Recurrence relation:
for each row i from 1 to n do
for each column j from 1 to m do
D[i, j] ← M IN( D[i−1, j] + del-cost(source[i]),
D[i−1, j−1] + sub-cost(source[i], target[j]),
D[i, j−1] + ins-cost(target[j]))
# Termination
return D[n,m]
Figure 2.19 The minimum edit distance algorithm, an example of the class of dynamic
programming algorithms. The various costs can either be fixed (e.g., ∀x, ins-cost(x) = 1)
or can be specific to the letter (to model the fact that some letters are more likely to be in-
serted than others). We assume that there is no cost for substituting a letter for itself (i.e.,
sub-cost(x, x) = 0).
Alignment Knowing the minimum edit distance is useful for algorithms like find-
ing potential spelling error corrections. But the edit distance algorithm is important
in another way; with a small change, it can also provide the minimum cost align-
ment between two strings. Aligning two strings is useful throughout speech and
language processing. In speech recognition, minimum edit distance alignment is
used to compute the word error rate (Chapter 15). Alignment plays a role in ma-
chine translation, in which sentences in a parallel corpus (a corpus with a text in two
languages) need to be matched to each other.
To extend the edit distance algorithm to produce an alignment, we can start by
visualizing an alignment as a path through the edit distance matrix. Figure 2.21
shows this path with boldfaced cells. Each boldfaced cell represents an alignment
of a pair of letters in the two strings. If two boldfaced cells occur in the same row,
2.9 • M INIMUM E DIT D ISTANCE 33
Src\Tar # e x e c u t i o n
# 0 1 2 3 4 5 6 7 8 9
i 1 2 3 4 5 6 7 6 7 8
n 2 3 4 5 6 7 8 7 8 7
t 3 4 5 6 7 8 7 8 9 8
e 4 3 4 5 6 7 8 9 10 9
n 5 4 5 6 7 8 9 10 11 10
t 6 5 6 7 8 9 8 9 10 11
i 7 6 7 8 9 10 9 8 9 10
o 8 7 8 9 10 11 10 9 8 9
n 9 8 9 10 11 12 11 10 9 8
Figure 2.20 Computation of minimum edit distance between intention and execution with
the algorithm of Fig. 2.19, using Levenshtein distance with cost of 1 for insertions or dele-
tions, 2 for substitutions.
there will be an insertion in going from the source to the target; two boldfaced cells
in the same column indicate a deletion.
Figure 2.21 also shows the intuition of how to compute this alignment path. The
computation proceeds in two steps. In the first step, we augment the minimum edit
distance algorithm to store backpointers in each cell. The backpointer from a cell
points to the previous cell (or cells) that we came from in entering the current cell.
We’ve shown a schematic of these backpointers in Fig. 2.21. Some cells have mul-
tiple backpointers because the minimum extension could have come from multiple
backtrace previous cells. In the second step, we perform a backtrace. In a backtrace, we start
from the last cell (at the final row and column), and follow the pointers back through
the dynamic programming matrix. Each complete path between the final cell and the
initial cell is a minimum distance alignment. Exercise 2.7 asks you to modify the
minimum edit distance algorithm to store the pointers and compute the backtrace to
output an alignment.
# e x e c u t i o n
# 0 ←1 ← 2 3
← 4
← 5 ← ← 6 ← 7 ← 8 ← 9
i ↑1 -←↑ 2 -←↑ 3 -←↑ 4 -←↑ 5 -←↑ 6 -←↑ 7 -6 ←7 ←8
n ↑2 -←↑ 3 -←↑ 4 -←↑ 5 -←↑ 6 -←↑ 7 -←↑ 8 ↑7 -←↑ 8 -7
t ↑3 -←↑ 4 -←↑ 5 -←↑ 6 -←↑ 7 -←↑ 8 -7 ←↑ 8 -←↑ 9 ↑8
e ↑4 -3 ←4 -← 5 ←6 ←7 ←↑ 8 -←↑ 9 -←↑ 10 ↑9
n ↑5 ↑4 -←↑ 5 -←↑ 6 -←↑ 7 -←↑ 8 -←↑ 9 -←↑ 10 -←↑ 11 -↑ 10
t ↑6 ↑5 -←↑ 6 -←↑ 7 -←↑ 8 -←↑ 9 -8 ←9 ← 10 ←↑ 11
i ↑7 ↑6 -←↑ 7 -←↑ 8 -←↑ 9 -←↑ 10 ↑9 -8 ←9 ← 10
o ↑8 ↑7 -←↑ 8 -←↑ 9 -←↑ 10 -←↑ 11 ↑ 10 ↑9 -8 ←9
n ↑9 ↑8 -←↑ 9 -←↑ 10 -←↑ 11 -←↑ 12 ↑ 11 ↑ 10 ↑9 -8
Figure 2.21 When entering a value in each cell, we mark which of the three neighboring
cells we came from with up to three arrows. After the table is full we compute an alignment
(minimum edit path) by using a backtrace, starting at the 8 in the lower-right corner and
following the arrows back. The sequence of bold cells represents one possible minimum
cost alignment between the two strings, again using Levenshtein distance with cost of 1 for
insertions or deletions, 2 for substitutions. Diagram design after Gusfield (1997).
While we worked our example with simple Levenshtein distance, the algorithm
in Fig. 2.19 allows arbitrary weights on the operations. For spelling correction, for
example, substitutions are more likely to happen between letters that are next to
34 C HAPTER 2 • W ORDS AND T OKENS
2.10 Summary
This chapter introduced the fundamental concepts of tokens and tokenization in lan-
guage processing. We discussed the linguistic levels of words, morphemes, and
characters, introduced Unicode code points and the UTF-8 encoding, introduced
the BPE algorithm for tokenization, and introduced the regular expression and the
minimum edit distance algorithm for comparing strings. Here’s a summary of the
main points we covered about these ideas:
• Words and morphemes are useful units of representation, but difficult to define
formally.
• Unicode is a system for representing characters in the many scripts used to
write the languages of the world.
• Each character is represented internally with a unique id called a code point,
and can be encoded in a file via encoding methods like UTF-8, which is a
variable-length encoding.
• Byte-Pair Encoding or BPE is the standard way to induce tokens in a data-
driven way. It is the first step in most large language models.
• BPE tokens are often roughly word or morpheme-sized, although they can be
as small as single characters.
• The regular expression language is a powerful tool for pattern-matching.
• Basic operations in regular expressions include disjunction of symbols ([],
|), counters (*, +, and {n,m}), anchors (ˆ, $), capture groups ((,)), and
substitutions.
• The minimum edit distance between two strings is the minimum number of
operations it takes to edit one into the other. Minimum edit distance can be
computed by dynamic programming, which also results in an alignment of
the two strings.
Historical Notes
For more on Herdan’s law and Heaps’ Law, see Herdan (1960, p. 28), Heaps (1978),
Egghe (2007) and Baayen (2001);
Unicode drew on ASCII and ISO character encoding standards. Early drafts
were worked out in discussions between engineers from Xerox and Apple. An early
draft standard was published in 1988, with a more formal release of the Unicode
Stanford in 1991. What became UTF-8 began with ISO drafts in 1989, with various
extensions. The self-synchronizing aspects were famously outlined on a placemat in
a New Jersey dinner in 1992 by Ken Thompson.
Word tokenization and other text normalization algorithms have been applied
since the beginning of the field. This include stemming, like the widely used stem-
mer of Lovins (1968), and applications to the digital humanities like those of by
Packard (1973), who built an affix-stripping morphological parser for Ancient Greek.
E XERCISES 35
BPE, originally a text compression method proposed by Gage (1994), was applied
to subword tokenization in the context of early neural machine translation by Sen-
nrich et al. (2016). It was then taken up in OpenAI’s GPT-2 (Radford et al., 2019)
as the default tokenization method, and also included in the open-source Sentence-
Piece library (Kudo and Richardson, 2018b). There is a nice a public implemen-
tation, minbpe, [Link] by Andrej Karpathy,
who also has a popular lecture introducing BPE ([Link]
watch?v=zduSFxRajkE).
Kleene 1951; 1956 first defined regular expressions and the finite automaton,
based on the McCulloch-Pitts neuron. Ken Thompson was one of the first to build
regular expressions compilers into editors for text searching (Thompson, 1968). His
editor ed included a command “g/regular expression/p”, or Global Regular Expres-
sion Print, which later became the Unix grep utility.
NLTK is an essential tool that offers both useful Python libraries (https://
[Link]) and textbook descriptions (Bird et al., 2009) of many algorithms
including text normalization and corpus interfaces.
For more on edit distance, see Gusfield (1997). Our example measuring the edit
distance from ‘intention’ to ‘execution’ was adapted from Kruskal (1983). There are
various publicly available packages to compute edit distance, including Unix diff
and the NIST sclite program (NIST, 2005).
In his autobiography Bellman (1984) explains how he originally came up with
the term dynamic programming:
“...The 1950s were not good years for mathematical research. [the]
Secretary of Defense ...had a pathological fear and hatred of the word,
research... I decided therefore to use the word, “programming”. I
wanted to get across the idea that this was dynamic, this was multi-
stage... I thought, let’s ... take a word that has an absolutely precise
meaning, namely dynamic... it’s impossible to use the word, dynamic,
in a pejorative sense. Try thinking of some combination that will pos-
sibly give it a pejorative meaning. It’s impossible. Thus, I thought
dynamic programming was a good name. It was something not even a
Congressman could object to.”
Exercises
2.1 Write regular expressions for the following languages.
1. the set of all alphabetic strings;
2. the set of all lower case alphabetic strings ending in a b;
3. the set of all strings from the alphabet a, b such that each a is immedi-
ately preceded by and immediately followed by a b;
2.2 Write regular expressions for the following languages. By “word”, we mean
an alphabetic string separated from other words by whitespace, any relevant
punctuation, line breaks, and so forth.
1. the set of all strings with two consecutive repeated words (e.g., “Hum-
bert Humbert” and “the the” but not “the bug” or “the big bug”);
2. all strings that start at the beginning of the line with an integer and that
end at the end of the line with a word;
36 C HAPTER 2 • W ORDS AND T OKENS
3. all strings that have both the word grotto and the word raven in them
(but not, e.g., words like grottos that merely contain the word grotto);
4. write a pattern that places the first word of an English sentence in a
register. Deal with punctuation.
2.3 Implement an ELIZA-like program, using substitutions such as those described
on page 27. You might want to choose a different domain than a Rogerian psy-
chologist, although keep in mind that you would need a domain in which your
program can legitimately engage in a lot of simple repetition.
2.4 Compute the edit distance (using insertion cost 1, deletion cost 1, substitution
cost 1) of “leda” to “deal”. Show your work (using the edit distance grid).
2.5 Figure out whether drive is closer to brief or to divers and what the edit dis-
tance is to each. You may use any version of distance that you like.
2.6 Now implement a minimum edit distance algorithm and use your hand-computed
results to check your code.
2.7 Augment the minimum edit distance algorithm to output an alignment; you
will need to store pointers and add a stage to compute the backtrace.
CHAPTER
“You are uniformly charming!” cried he, with a smile of associating and now
and then I bowed and they perceived a chaise and four to wish for.
Random sentence generated from a Jane Austen trigram model
Predicting is difficult—especially about the future, as the old quip goes. But how
about predicting something that seems much easier, like the next word someone is
going to say? What word, for example, is likely to follow
The water of Walden Pond is so beautifully ...
You might conclude that a likely word is blue, or green, or clear, but probably
not refrigerator nor this. In this chapter we formalize this intuition by intro-
language model ducing n-gram language models or LMs. A language model is a machine learning
LM model that predicts upcoming words. More formally, a language model assigns a
probability to each possible next word, or equivalently gives a probability distribu-
tion over possible next words. Language models can also assign a probability to an
entire sentence. Thus an LM could tell us that the following sequence has a much
higher probability of appearing in a text:
all of a sudden I notice three guys standing on the sidewalk
Why would we want to predict upcoming words? The main reason is that large
language models are built just by training them to predict words!! As we’ll see
in chapters 5-10, large language models learn an enormous amount about language
solely from being trained to predict upcoming words from neighboring words.
This probabilistic knowledge can be very practical. Consider correcting gram-
mar or spelling errors like Their are two midterms, in which There was mistyped
as Their, or Everything has improve, in which improve should have been
improved. The phrase There are is more probable than Their are, and has
improved than has improve, so a language model can help users select the more
grammatical variant.
Or for a speech system to recognize that you said I will be back soonish
and not I will be bassoon dish, it helps to know that back soonish is a more
probable sequence. Language models can also help in augmentative and alterna-
AAC tive communication (Trnka et al. 2007, Kane et al. 2017). People can use AAC
systems if they are physically unable to speak or sign but can instead use eye gaze
or other movements to select words from a menu. Word prediction can be used to
suggest likely words for the menu.
n-gram In this chapter we introduce the simplest kind of language model: the n-gram
38 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
3.1 N-Grams
Let’s begin with the task of computing P(w|h), the probability of a word w given
some history h. Suppose the history h is “The water of Walden Pond is so
beautifully ” and we want to know the probability that the next word is blue:
One way to estimate this probability is directly from relative frequency counts: take a
very large corpus, count the number of times we see The water of Walden Pond
is so beautifully, and count the number of times this is followed by blue. This
would be answering the question “Out of the times we saw the history h, how many
times was it followed by the word w”, as follows:
If we had a large enough corpus, we could compute these two counts and estimate
the probability from Eq. 3.2. But even the entire web isn’t big enough to give us
good estimates for counts of entire sentences. This is because language is creative;
new sentences are invented all the time, and we can’t expect to get accurate counts
for such large objects as entire sentences. For this reason, we’ll need more clever
ways to estimate the probability of a word w given a history h, or the probability of
an entire word sequence W .
Let’s start with some notation. First, throughout this chapter we’ll continue to
refer to words, although in practice we usually compute language models over to-
kens like the BPE tokens of page 13. To represent the probability of a particular
random variable Xi taking on the value “the”, or P(Xi = “the”), we will use the
simplification P(the). We’ll represent a sequence of n words either as w1 . . . wn or
w1:n . Thus the expression w1:n−1 means the string w1 , w2 , ..., wn−1 , but we’ll also
be using the equivalent notation w<n , which can be read as “all the elements of w
from w1 up to and including wn−1 ”. For the joint probability of each word in a se-
quence having a particular value P(X1 = w1 , X2 = w2 , X3 = w3 , ..., Xn = wn ) we’ll
use P(w1 , w2 , ..., wn ).
Now, how can we compute probabilities of entire sequences like P(w1 , w2 , ..., wn )?
One thing we can do is decompose this probability using the chain rule of proba-
3.1 • N-G RAMS 39
bility:
The chain rule shows the link between computing the joint probability of a sequence
and computing the conditional probability of a word given previous words. Equa-
tion 3.4 suggests that we could estimate the joint probability of an entire sequence of
words by multiplying together a number of conditional probabilities. But using the
chain rule doesn’t really seem to help us! We don’t know any way to compute the
exact probability of a word given a long sequence of preceding words, P(wn |w1:n−1 ).
As we said above, we can’t just estimate by counting the number of times every word
occurs following every long string in some corpus, because language is creative and
any particular context might have never occurred before!
P(blue|beautifully) (3.6)
When we use a bigram model to predict the conditional probability of the next word,
we are thus making the following approximation:
The assumption that the probability of a word depends only on the previous word is
Markov called a Markov assumption. Markov models are the class of probabilistic models
that assume we can predict the probability of some future unit without looking too
far into the past. We can generalize the bigram (which looks one word into the past)
n-gram to the trigram (which looks two words into the past) and thus to the n-gram (which
looks n − 1 words into the past).
Let’s see a general equation for this n-gram approximation to the conditional
probability of the next word in a sequence. We’ll use N here to mean the n-gram
40 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
Given the bigram assumption for the probability of an individual word, we can com-
pute the probability of a complete word sequence by substituting Eq. 3.7 into Eq. 3.4:
n
Y
P(w1:n ) ≈ P(wk |wk−1 ) (3.9)
k=1
C(wn−1 wn )
P(wn |wn−1 ) = P (3.10)
w C(wn−1 w)
We can simplify this equation, since the sum of all bigram counts that start with
a given word wn−1 must be equal to the unigram count for that word wn−1 (the reader
should take a moment to be convinced of this):
C(wn−1 wn )
P(wn |wn−1 ) = (3.11)
C(wn−1 )
Let’s work through an example using a mini-corpus of three sentences. We’ll
first need to augment each sentence with a special symbol <s> at the beginning
of the sentence, to give us the bigram context of the first word. We’ll also need a
special end-symbol </s>.1
<s> I am Sam </s>
<s> Sam I am </s>
<s> I do not like green eggs and ham </s>
Here are the calculations for some of the bigram probabilities from this corpus
2 1 2
P(I|<s>) = 3 = 0.67 P(Sam|<s>) = 3 = 0.33 P(am|I) = 3 = 0.67
1 1 1
P(</s>|Sam) = 2 = 0.5 P(Sam|am) = 2 = 0.5 P(do|I) = 3 = 0.33
For the general case of MLE n-gram parameter estimation:
C(wn−N+1:n−1 wn )
P(wn |wn−N+1:n−1 ) = (3.12)
C(wn−N+1:n−1 )
1 We need the end-symbol to make the bigram grammar a true probability distribution. Without an end-
symbol, instead of the sentence probabilities of all sentences summing to one, the sentence probabilities
for all sentences of a given length would sum to one. This model would define an infinite set of probability
distributions, with one distribution per sentence length. See Exercise 3.5.
3.1 • N-G RAMS 41
Equation 3.12 (like Eq. 3.11) estimates the n-gram probability by dividing the
observed frequency of a particular sequence by the observed frequency of a prefix.
relative
frequency This ratio is called a relative frequency. We said above that this use of relative
frequencies as a way to estimate probabilities is an example of maximum likelihood
estimation or MLE. In MLE, the resulting parameter set maximizes the likelihood of
the training set T given the model M (i.e., P(T |M)). For example, suppose the word
Chinese occurs 400 times in a corpus of a million words. What is the probability
that a random word selected from some other text of, say, a million words will be the
400
word Chinese? The MLE of its probability is 1000000 or 0.0004. Now 0.0004 is not
the best possible estimate of the probability of Chinese occurring in all situations; it
might turn out that in some other corpus or context Chinese is a very unlikely word.
But it is the probability that makes it most likely that Chinese will occur 400 times
in a million-word corpus. We present ways to modify the MLE estimates slightly to
get better probability estimates in Section 3.6.
Let’s move on to some examples from a real but tiny corpus, drawn from the
now-defunct Berkeley Restaurant Project, a dialogue system from the last century
that answered questions about a database of restaurants in Berkeley, California (Ju-
rafsky et al., 1994). Here are some sample user queries (text-normalized, by lower
casing and with punctuation striped) (a sample of 9332 sentences is on the website):
can you tell me about any good cantonese restaurants close by
tell me about chez panisse
i’m looking for a good place to eat breakfast
when is caffe venezia open during the day
Figure 3.1 shows the bigram counts from part of a bigram grammar from text-
normalized Berkeley Restaurant Project sentences. Note that the majority of the
values are zero. In fact, we have chosen the sample words to cohere with each other;
a matrix selected from a random set of eight words would be even more sparse.
Figure 3.2 shows the bigram probabilities after normalization (dividing each cell
in Fig. 3.1 by the appropriate unigram for its row, taken from the following set of
unigram counts):
In practice throughout this book, we’ll use log to mean natural log (ln) when the
base is not specified.
3.2 • E VALUATING L ANGUAGE M ODELS : T RAINING AND T EST S ETS 43
Longer context Although for pedagogical purposes we have only described bi-
trigram gram models, when there is sufficient training data we use trigram models, which
4-gram condition on the previous two words, or 4-gram or 5-gram models. For these larger
5-gram n-grams, we’ll need to assume extra contexts to the left and right of the sentence end.
For example, to compute trigram probabilities at the very beginning of the sentence,
we use two pseudo-words for the first trigram (i.e., P(I|<s><s>).
Some large n-gram datasets have been created, like the million most frequent
n-grams drawn from the Corpus of Contemporary American English (COCA), a
curated 1 billion word corpus of American English (Davies, 2020), Google’s Web
5-gram corpus from 1 trillion words of English web text (Franz and Brants, 2006),
or the Google Books Ngrams corpora (800 billion tokens from Chinese, English,
French, German, Hebrew, Italian, Russian, and Spanish) (Lin et al., 2012a)).
It’s even possible to use extremely long-range n-gram context. The infini-gram
(∞-gram) project (Liu et al., 2024) allows n-grams of any length. Their idea is to
avoid the expensive (in space and time) pre-computation of huge n-gram count ta-
bles. Instead, n-gram probabilities with arbitrary n are computed quickly at inference
time by using an efficient representation called suffix arrays. This allows computing
of n-grams of every length for enormous corpora of 5 trillion tokens.
Efficiency considerations are important when building large n-gram language
models. It is standard to quantize the probabilities using only 4-8 bits (instead of
8-byte floats), store the word strings on disk and represent them in memory only as
a 64-bit hash, and represent n-grams in special data structures like ‘reverse tries’.
It is also common to prune n-gram language models, for example by only keeping
n-grams with counts greater than some threshold or using entropy to prune less-
important n-grams (Stolcke, 1998). Efficient language model toolkits like KenLM
(Heafield 2011, Heafield et al. 2013) use sorted arrays and use merge sorts to effi-
ciently build the probability tables in a minimal number of passes through a large
corpus.
The training set is the data we use to learn the parameters of our model; for
simple n-gram language models it’s the corpus from which we get the counts that
we normalize into the probabilities of the n-gram language model.
The test set is a different, held-out set of data, not overlapping with the training
set, that we use to evaluate the model. We need a separate test set to give us an
unbiased estimate of how well the model we trained can generalize when we apply
it to some new unknown dataset. A machine learning model that perfectly captured
the training data, but performed terribly on any other data, wouldn’t be much use
when it comes time to apply it to any new data or problem! We thus measure the
quality of an n-gram model by its performance on this unseen test set or test corpus.
How should we choose a training and test set? The test set should reflect the
language we want to use the model for. If we’re going to use our language model
for speech recognition of chemistry lectures, the test set should be text of chemistry
lectures. If we’re going to use it as part of a system for translating hotel booking re-
quests from Chinese to English, the test set should be text of hotel booking requests.
If we want our language model to be general purpose, then the test set should be
drawn from a wide variety of texts. In such cases we might collect a lot of texts
from different sources, and then divide it up into a training set and a test set. It’s
important to do the dividing carefully; if we’re building a general purpose model,
we don’t want the test set to consist of only text from one document, or one author,
since that wouldn’t be a good measure of general performance.
Thus if we are given a corpus of text and want to compare the performance of
two different n-gram models, we divide the data into training and test sets, and train
the parameters of both models on the training set. We can then compare how well
the two trained models fit the test set.
But what does it mean to “fit the test set”? The standard answer is simple:
whichever language model assigns a higher probability to the test set—which
means it more accurately predicts the test set—is a better model. Given two proba-
bilistic models, the better model is the one that better predicts the details of the test
data, and hence will assign a higher probability to the test data.
Since our evaluation metric is based on test set probability, it’s important not
to let the test sentences into the training set. Suppose we are trying to compute
the probability of a particular “test” sentence. If our test sentence is part of the
training corpus, we will mistakenly assign it an artificially high probability when
it occurs in the test set. We call this situation training on the test set or also data
data contamination. Training on the test set introduces a bias that makes the probabilities
contamination
all look too high, and causes huge inaccuracies in perplexity, the probability-based
metric we introduce below.
Even if we don’t train on the test set, if we test our language model on the
test set many times after making different changes, we might implicitly tune to its
characteristics, by noticing which changes seem to make the model better. For this
reason, we only want to run our model on the test set once, or a very few number of
times, once we are sure our model is ready.
development For this reason we normally instead have a third dataset called a development
test
test set or, devset. We do all our testing on this dataset until the very end, and then
we test on the test set once to see how good our model is.
How do we divide our data into training, development, and test sets? We want
our test set to be as large as possible, since a small test set may be accidentally un-
representative, but we also want as much training data as possible. At the minimum,
we would want to pick the smallest test set that gives us enough statistical power
3.3 • E VALUATING L ANGUAGE M ODELS : P ERPLEXITY 45
Note that because of the inverse in Eq. 3.15, the higher the probability of the word
sequence, the lower the perplexity. Thus the the lower the perplexity of a model on
the data, the better the model. Minimizing perplexity is equivalent to maximizing
the test set probability according to the language model. Why does perplexity use
the inverse probability? It turns out the inverse arises from the original definition
of perplexity from cross-entropy rate in information theory; for those interested, the
explanation is in the advanced Section 3.7. Meanwhile, we just have to remember
that perplexity has an inverse relationship with probability.
The details of computing the perplexity of a test set W depends on which lan-
guage model we use. Here’s the perplexity of W with a unigram language model
(just the geometric mean of the inverse of the unigram probabilities):
v
uN
uY 1
perplexity(W ) = t N
(3.16)
P(wi )
i=1
46 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
What we generally use for word sequence in Eq. 3.15 or Eq. 3.17 is the entire
sequence of words in some test set. Since this sequence will cross many sentence
boundaries, if our vocabulary includes a between-sentence token <EOS> or separate
begin- and end-sentence markers <s> and </s> then we can include them in the
probability computation. If we do, then we also include one token per sentence in
the total count of word tokens N.2
We mentioned above that perplexity is a function of both the text and the lan-
guage model: given a text W , different language models will have different perplex-
ities. Because of this, perplexity can be used to compare different language models.
For example, here we trained unigram, bigram, and trigram models on 38 million
words from the Wall Street Journal newspaper. We then computed the perplexity of
each of these models on a WSJ test set using Eq. 3.16 for unigrams, Eq. 3.17 for
bigrams, and the corresponding equation for trigrams. The table below shows the
perplexity of the 1.5 million word test set according to each of the language models.
Unigram Bigram Trigram
Perplexity 962 170 109
As we see above, the more information the n-gram gives us about the word
sequence, the higher the probability the n-gram will assign to the string. A trigram
model is less surprised than a unigram model because it has a better idea of what
words might come next, and so it assigns them a higher probability. And the higher
the probability, the lower the perplexity (since as Eq. 3.15 showed, perplexity is
related inversely to the probability of the test sequence according to the model). So
a lower perplexity tells us that a language model is a better predictor of the test set.
Note that in computing perplexities, the language model must be constructed
without any knowledge of the test set, or else the perplexity will be artificially low.
And the perplexity of two language models is only comparable if they use identical
vocabularies.
An (intrinsic) improvement in perplexity does not guarantee an (extrinsic) im-
provement in the performance of a language processing task like speech recognition
or machine translation. Nonetheless, because perplexity usually correlates with task
improvements, it is commonly used as a convenient evaluation metric. Still, when
possible a model’s improvement in perplexity should be confirmed by an end-to-end
evaluation on a real task.
language that is deterministic (no probabilities), any word can follow any word, and
whose vocabulary consists of only three colors:
We should expect the perplexity of the same test set red red red red blue for
language model B to be lower since most of the time the next color will be red, which
is very predictable, i.e. has a high probability. So the probability of the test set will
be higher, and since perplexity is inversely related to probability, the perplexity will
be lower. Thus, although the branching factor is still 3, the perplexity or weighted
branching factor is smaller:
polyphonic
p=0.0000018
however
the of a to in (p=0.0003)
Figure 3.3 A visualization of the sampling distribution for sampling sentences by repeat-
edly sampling unigrams. The blue bar represents the relative frequency of each word (we’ve
ordered them from most frequent to least frequent, but the choice of order is arbitrary). The
number line shows the cumulative probabilities. If we choose a random number between 0
and 1, it will fall in an interval corresponding to some word. The expectation for the random
number to fall in the larger intervals of one of the frequent words (the, of, a) is much higher
than in the smaller interval of one of the rare words (polyphonic).
–To him swallowed confess hear both. Which. Of save on trail for are ay device and
1
gram
rote life have
–Hill he late speaks; or! a more to leg less first you enter
–Why dost stand forth thy canopy, forsooth; he is this palpable hit the King Henry. Live
2
gram
king. Follow.
–What means, sir. I confess she? then all sorts, he is trim, captain.
–Fly, and will rid me these news of price. Therefore the sadness of parting, as they say,
3
gram
’tis done.
–This shall forbid it should be branded, if renown made it empty.
–King Henry. What! I will go seek the traitor Gloucester. Exeunt some of the watch. A
4
gram
great banquet serv’d in;
–It cannot be but so.
Figure 3.4 Eight sentences randomly generated from four n-gram models computed from Shakespeare’s
works. All characters were mapped to lower-case and punctuation marks were treated as words. Output is
hand-corrected for capitalization to improve readability.
and the WSJ are both English, so we might have expected some overlap between our
n-grams for the two genres. Fig. 3.5 shows sentences generated by unigram, bigram,
and trigram models trained on 40 million words from WSJ.
1
gram
Months the my and issue of year foreign new exchange’s september
were recession exchange new endorsed a acquire to six executives
Last December through the way to preserve the Hudson corporation N.
2
gram
B. E. C. Taylor would seem to complete the major central planners one
point five percent of U. S. E. has already old M. X. corporation of living
on information such as more frequently fishing to keep her
They also point to ninety nine point six billion dollars from two hundred
3
gram
four oh six three percent of the rates of interest stores as Mexico and
Brazil on market conditions
Figure 3.5 Three sentences randomly generated from three n-gram models computed from
40 million words of the Wall Street Journal, lower-casing all characters and treating punctua-
tion as words. Output was then hand-corrected for capitalization to improve readability.
Compare these examples to the pseudo-Shakespeare in Fig. 3.4. While they both
model “English-like sentences”, there is no overlap in the generated sentences, and
little overlap even in small phrases. Statistical models are pretty useless as predictors
if the training sets and the test sets are as different as Shakespeare and the WSJ.
How should we deal with this problem when we build n-gram models? One step
is to be sure to use a training corpus that has a similar genre to whatever task we are
trying to accomplish. To build a language model for translating legal documents,
we need a training corpus of legal documents. To build a language model for a
question-answering system, we need a training corpus of questions.
It is equally important to get training data in the appropriate dialect or variety,
especially when processing social media posts or spoken transcripts. For exam-
ple some tweets will use features of African American English (AAE)— the name
for the many variations of language used in African American communities (King,
2020). Such features can include words like finna—an auxiliary verb that marks
immediate future tense —that don’t occur in other varieties, or spellings like den for
then, in tweets like this one (Blodgett and O’Connor, 2017):
50 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
of the word wi is its count ci normalized by the total number of word tokens N:
ci
P(wi ) =
N
Laplace smoothing merely adds one to each count (hence its alternate name add-
add-one one smoothing). Since there are V words in the vocabulary and each one was in-
cremented, we also need to adjust the denominator to take into account the extra V
observations. (What happens to our P values if we don’t increase the denominator?)
ci + 1
PLaplace (wi ) = (3.24)
N +V
Now that we have the intuition for the unigram case, let’s smooth our Berkeley
Restaurant Project bigrams. Figure 3.6 shows the add-one smoothed counts for the
bigrams in Fig. 3.1.
Figure 3.7 shows the add-one smoothed probabilities for the bigrams in Fig. 3.2,
computed by Eq. 3.26 below. Recall that normal bigram probabilities are computed
by normalizing each row of counts by the unigram count:
C(wn−1 wn )
PMLE (wn |wn−1 ) = (3.25)
C(wn−1 )
For add-one smoothed bigram counts, we need to augment the unigram count in the
denominator by the number of total word types in the vocabulary V . We can see
why this is in the following equation, which makes it explicit that the unigram count
in the denominator is really the sum over all the bigrams that start with wn−1 . Since
we add one to each of these, and there are V of them, we add a total of V to the
denominator:
C(wn−1 wn ) + 1 C(wn−1 wn ) + 1
PLaplace (wn |wn−1 ) = P = (3.26)
w (C(wn−1 w) + 1) C(wn−1 ) +V
Thus, each of the unigram counts given on page 41 will need to be augmented by V =
1446. The result, using Eq. 3.26, is the smoothed bigram probabilities in Fig. 3.7.
One useful visualization technique is to reconstruct an adjusted count matrix
so we can see how much a smoothing algorithm has changed the original counts.
This adjusted count C∗ is the count that, if divided by C(wn−1 ), would result in
the smoothed probability. This adjusted count is easier to compare directly with
the MLE counts. That is, the Laplace probability can equally be expressed as the
adjusted count divided by the (non-smoothed) denominator from Eq. 3.25:
C(wn−1 wn ) + 1 C∗ (wn−1 wn )
PLaplace (wn |wn−1 ) = =
C(wn−1 ) +V C(wn−1 )
52 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
[C(wn−1 wn ) + 1] ×C(wn−1 )
C∗ (wn−1 wn ) = (3.27)
C(wn−1 ) +V
Figure 3.8 shows the reconstructed counts, computed by Eq. 3.27.
i want to eat chinese food lunch spend
i 3.8 527 0.64 6.4 0.64 0.64 0.64 1.9
want 1.2 0.39 238 0.78 2.7 2.7 2.3 0.78
to 1.9 0.63 3.1 430 1.9 0.63 4.4 133
eat 0.34 0.34 1 0.34 5.8 1 15 0.34
chinese 0.2 0.098 0.098 0.098 0.098 8.2 0.2 0.098
food 6.9 0.43 6.9 0.43 0.86 2.2 0.43 0.43
lunch 0.57 0.19 0.19 0.19 0.19 0.38 0.19 0.19
spend 0.32 0.16 0.32 0.16 0.16 0.16 0.16 0.16
Figure 3.8 Add-one reconstituted counts for eight words (of V = 1446) in the BeRP corpus
of 9332 sentences, computed by Eq. 3.27. Previously-zero counts are in gray.
Note that add-one smoothing has made a very big change to the counts. Com-
paring Fig. 3.8 to the original counts in Fig. 3.1, we can see that C(want to) changed
from 608 to 238. We can see this in probability space as well: P(to|want) decreases
from 0.66 in the unsmoothed case to 0.26 in the smoothed case. Looking at the dis-
count d, defined as the ratio between new and old counts, shows us how strikingly
the counts for each prefix word have been reduced; the discount for the bigram want
to is 0.39, while the discount for Chinese food is 0.10, a factor of 10. The sharp
change occurs because too much probability mass is moved to all the zeros.
∗ C(wn−1 wn ) + k
PAdd-k (wn |wn−1 ) = (3.28)
C(wn−1 ) + kV
Add-k smoothing requires that we have a method for choosing k; this can be
done, for example, by optimizing on a devset. Although add-k is useful for some
tasks (including text classification), it turns out that it still doesn’t work well for
3.6 • S MOOTHING , I NTERPOLATION , AND BACKOFF 53
language modeling, generating counts with poor variances and often inappropriate
discounts (Gale and Church, 1994).
How are these λ values set? Both the simple interpolation and conditional interpo-
held-out lation λ s are learned from a held-out corpus. A held-out corpus is an additional
training corpus, so-called because we hold it out from the training data, that we use
to set these λ values.3 We do so by choosing the λ values that maximize the likeli-
hood of the held-out corpus. That is, we fix the n-gram probabilities and then search
for the λ values that—when plugged into Eq. 3.29—give us the highest probability
of the held-out set. There are various ways to find this optimal set of λ s. One way
is to use the EM algorithm, an iterative learning algorithm that converges on locally
optimal λ s (Jelinek and Mercer, 1980).
count(w)
The backoff terminates in the unigram, which has score S(w) = N . Brants et al.
(2007) find that a value of 0.4 worked well for λ .
The log can, in principle, be computed in any base. If we use log base 2, the
resulting value of entropy will be measured in bits.
One intuitive way to think about entropy is as a lower bound on the number of
bits it would take to encode a certain decision or piece of information in the optimal
coding scheme. Consider an example from the standard information theory textbook
Cover and Thomas (1991). Imagine that we want to place a bet on a horse race but
it is too far to go all the way to Yonkers Racetrack, so we’d like to send a short
message to the bookie to tell him which of the eight horses to bet on. One way to
encode this message is just to use the binary representation of the horse’s number
as the code; thus, horse 1 would be 001, horse 2 010, horse 3 011, and so on, with
horse 8 coded as 000. If we spend the whole day betting and each horse is coded
with 3 bits, on average we would be sending 3 bits per race.
Can we do better? Suppose that the spread is the actual distribution of the bets
placed and that we represent it as the prior probability of each horse as follows:
3.7 • A DVANCED : P ERPLEXITY ’ S R ELATION TO E NTROPY 55
1 1
Horse 1 2 Horse 5 64
1 1
Horse 2 4 Horse 6 64
1 1
Horse 3 8 Horse 7 64
1 1
Horse 4 16 Horse 8 64
The entropy of the random variable X that ranges over horses gives us a lower
bound on the number of bits and is
i=8
X
H(X) = − p(i) log2 p(i)
i=1
= 1 log 1 −4( 1 log 1 )
− 12 log2 12 − 14 log2 41 − 18 log2 18 − 16 2 16 64 2 64
= 2 bits (3.33)
A code that averages 2 bits per race can be built with short encodings for more
probable horses, and longer encodings for less probable horses. For example, we
could encode the most likely horse with the code 0, and the remaining horses as 10,
then 110, 1110, 111100, 111101, 111110, and 111111.
What if the horses are equally likely? We saw above that if we used an equal-
length binary code for the horse numbers, each horse took 3 bits to code, so the
average was 3. Is the entropy the same? In this case each horse would have a
probability of 18 . The entropy of the choice of horses is then
i=8
X 1 1 1
H(X) = − log2 = − log2 = 3 bits (3.34)
8 8 8
i=1
Until now we have been computing the entropy of a single variable. But most of
what we will use entropy for involves sequences. For a grammar, for example, we
will be computing the entropy of some sequence of words W = {w1 , w2 , . . . , wn }.
One way to do this is to have a variable that ranges over sequences of words. For
example we can compute the entropy of a random variable that ranges over all se-
quences of words of length n in some language L as follows:
X
H(w1 , w2 , . . . , wn ) = − p(w1 : n ) log p(w1 : n ) (3.35)
w1 : n ∈L
entropy rate We could define the entropy rate (we could also think of this as the per-word
entropy) as the entropy of this sequence divided by the number of words:
1 1 X
H(w1 : n ) = − p(w1 : n ) log p(w1 : n ) (3.36)
n n
w1 : n ∈L
1
H(L) = lim H(w1 : n )
n
n→∞
1X
= − lim p(w1 : n ) log p(w1 : n ) (3.37)
n→∞ n
W ∈L
56 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
The Shannon-McMillan-Breiman theorem (Algoet and Cover 1988, Cover and Thomas
1991) states that if the language is regular in certain ways (to be exact, if it is both
stationary and ergodic),
1
H(L) = lim − log p(w1 : n ) (3.38)
n→∞ n
That is, we can take a single sequence that is long enough instead of summing over
all possible sequences. The intuition of the Shannon-McMillan-Breiman theorem
is that a long-enough sequence of words will contain in it many other shorter se-
quences and that each of these shorter sequences will reoccur in the longer sequence
according to their probabilities.
Stationary A stochastic process is said to be stationary if the probabilities it assigns to a
sequence are invariant with respect to shifts in the time index. In other words, the
probability distribution for words at time t is the same as the probability distribution
at time t + 1. Markov models, and hence n-grams, are stationary. For example, in
a bigram, Pi is dependent only on Pi−1 . So if we shift our time index by x, Pi+x is
still dependent on Pi+x−1 . But natural language is not stationary, since as we show
in Appendix D, the probability of upcoming words can be dependent on events that
were arbitrarily distant and time dependent. Thus, our statistical models only give
an approximation to the correct distributions and entropies of natural language.
To summarize, by making some incorrect but convenient simplifying assump-
tions, we can compute the entropy of some stochastic process by taking a very long
sample of the output and computing its average log probability.
cross-entropy Now we are ready to introduce cross-entropy. The cross-entropy is useful when
we don’t know the actual probability distribution p that generated some data. It
allows us to use some m, which is a model of p (i.e., an approximation to p). The
cross-entropy of m on p is defined by
1X
H(p, m) = lim − p(w1 , . . . , wn ) log m(w1 , . . . , wn ) (3.39)
n→∞ n
W ∈L
That is, we draw sequences according to the probability distribution p, but sum the
log of their probabilities according to m.
Again, following the Shannon-McMillan-Breiman theorem, for a stationary er-
godic process:
1
H(p, m) = lim − log m(w1 w2 . . . wn ) (3.40)
n→∞ n
This means that, as for entropy, we can estimate the cross-entropy of a model m
on some distribution p by taking a single sequence that is long enough instead of
summing over all possible sequences.
What makes the cross-entropy useful is that the cross-entropy H(p, m) is an up-
per bound on the entropy H(p). For any model m:
This means that we can use some simplified model m to help estimate the true en-
tropy of a sequence of symbols drawn according to probability p. The more accurate
m is, the closer the cross-entropy H(p, m) will be to the true entropy H(p). Thus,
the difference between H(p, m) and H(p) is a measure of how accurate a model is.
Between two models m1 and m2 , the more accurate model will be the one with the
3.8 • S UMMARY 57
lower cross-entropy. (The cross-entropy can never be lower than the true entropy, so
a model cannot err by underestimating the true entropy.)
We are finally ready to see the relation between perplexity and cross-entropy
as we saw it in Eq. 3.40. Cross-entropy is defined in the limit as the length of the
observed word sequence goes to infinity. We approximate this cross-entropy by
relying on a (sufficiently long) sequence of fixed length. This approximation to the
cross-entropy of a model M = P(wi |wi−N+1 : i−1 ) on a sequence of words W is
1
H(W ) = − log P(w1 w2 . . . wN ) (3.42)
N
perplexity The perplexity of a model P on a sequence of words W is now formally defined as
2 raised to the power of this cross-entropy:
Perplexity(W ) = 2H(W )
1
= P(w1 w2 . . . wN )− N
s
1
= N
P(w1 w2 . . . wN )
3.8 Summary
This chapter introduced language modeling via the n-gram model, a classic model
that allows us to introduce many of the basic concepts in language modeling.
• Language models offer a way to assign a probability to a sentence or other
sequence of words or tokens, and to predict a word or token from preceding
words or tokens.
• N-grams are perhaps the simplest kind of language model. They are Markov
models that estimate words from a fixed window of previous words. N-gram
models can be trained by counting in a training corpus and normalizing the
counts (the maximum likelihood estimate).
• N-gram language models can be evaluated on a test set using perplexity.
• The perplexity of a test set according to a language model is a function of
the probability of the test set: the inverse test set probability according to the
model, normalized by the length.
• Sampling from a language model means to generate some sentences, choos-
ing each sentence according to its likelihood as defined by the model.
• Smoothing algorithms provide a way to estimate probabilities for events that
were unseen in training. Commonly used smoothing algorithms for n-grams
include add-1 smoothing, or rely on lower-order n-gram counts through inter-
polation.
Historical Notes
The underlying mathematics of the n-gram was first proposed by Markov (1913),
who used what are now called Markov chains (bigrams and trigrams) to predict
whether an upcoming letter in Pushkin’s Eugene Onegin would be a vowel or a con-
sonant. Markov classified 20,000 letters as V or C and computed the bigram and
58 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
trigram probability that a given letter would be a vowel given the previous one or
two letters. Shannon (1948) applied n-grams to compute approximations to English
word sequences. Based on Shannon’s work, Markov models were commonly used in
engineering, linguistic, and psychological work on modeling word sequences by the
1950s. In a series of extremely influential papers starting with Chomsky (1956) and
including Chomsky (1957) and Miller and Chomsky (1963), Noam Chomsky argued
that “finite-state Markov processes”, while a possibly useful engineering heuristic,
were incapable of being a complete cognitive model of human grammatical knowl-
edge. These arguments led many linguists and computational linguists to ignore
work in statistical modeling for decades.
The resurgence of n-gram language models came from Fred Jelinek and col-
leagues at the IBM Thomas J. Watson Research Center, who were influenced by
Shannon, and James Baker at CMU, who was influenced by the prior, classified
work of Leonard Baum and colleagues on these topics at labs like the US Institute
for Defense Analyses (IDA) after they were declassified. Independently these two
labs successfully used n-grams in their speech recognition systems at the same time
(Baker 1975b, Jelinek et al. 1975, Baker 1975a, Bahl et al. 1983, Jelinek 1990). The
terms “language model” and “perplexity” were first used for this technology by the
IBM group. Jelinek and his colleagues used the term language model in a pretty
modern way, to mean the entire set of linguistic influences on word sequence prob-
abilities, including grammar, semantics, discourse, and even speaker characteristics,
rather than just the particular n-gram model itself.
Add-one smoothing derives from Laplace’s 1812 law of succession and was first
applied as an engineering solution to the zero frequency problem by Jeffreys (1948)
based on an earlier Add-K suggestion by Johnson (1932). Problems with the add-
one algorithm are summarized in Gale and Church (1994).
A wide variety of different language modeling and smoothing techniques were
proposed in the 80s and 90s, including Good-Turing discounting—first applied to the
n-gram smoothing at IBM by Katz (Nádas 1984, Church and Gale 1991)— Witten-
class-based
n-gram Bell discounting (Witten and Bell, 1991), and varieties of class-based n-gram mod-
els that used information about word classes. Starting in the late 1990s, Chen and
Goodman performed a number of carefully controlled experiments comparing dif-
ferent algorithms and parameters (Chen and Goodman 1999, Goodman 2006, inter
alia). They showed the advantages of Modified Interpolated Kneser-Ney, which
became the standard baseline for n-gram language modeling around the turn of the
century, especially because they showed that caches and class-based models pro-
vided only minor additional improvement. SRILM (Stolcke, 2002) and KenLM
(Heafield 2011, Heafield et al. 2013) are publicly available toolkits for building n-
gram language models.
Large language models are based on neural networks rather than n-grams, en-
abling them to solve the two major problems with n-grams: (1) the number of param-
eters increases exponentially as the n-gram order increases, and (2) n-grams have no
way to generalize from training examples to test set examples unless they use iden-
tical words. Neural language models instead project words into a continuous space
in which words with similar contexts have similar representations. We’ll introduce
transformer-based large language models in Chapter 8, along the way introducing
feedforward language models (Bengio et al. 2006, Schwenk 2007) in Chapter 6 and
recurrent language models (Mikolov, 2012) in Chapter 13.
E XERCISES 59
Exercises
3.1 Write out the equation for trigram probability estimation (modifying Eq. 3.11).
Now write out all the non-zero trigram probabilities for the I am Sam corpus
on page 40.
3.2 Calculate the probability of the sentence i want chinese food. Give two
probabilities, one using Fig. 3.2 and the ‘useful probabilities’ just below it on
page 42, and another using the add-1 smoothed table in Fig. 3.7. Assume the
additional add-1 smoothed probabilities P(i|<s>) = 0.19 and P(</s>|food) =
0.40.
3.3 Which of the two probabilities you computed in the previous exercise is higher,
unsmoothed or smoothed? Explain why.
3.4 We are given the following corpus, modified from the one in the chapter:
<s> I am Sam </s>
<s> Sam I am </s>
<s> I am Sam </s>
<s> I do not like green eggs and Sam </s>
Using a bigram language model with add-one smoothing, what is P(Sam |
am)? Include <s> and </s> in your counts just like any other token.
3.5 Suppose we didn’t use the end-symbol </s>. Train an unsmoothed bigram
grammar on the following training corpus without using the end-symbol </s>:
<s> a b
<s> b b
<s> b a
<s> a a
Demonstrate that your bigram model does not assign a single probability dis-
tribution across all sentence lengths by showing that the sum of the probability
of the four possible 2 word sentences over the alphabet {a,b} is 1.0, and the
sum of the probability of all possible 3 word sentences over the alphabet {a,b}
is also 1.0.
3.6 Suppose we train a trigram language model with add-one smoothing on a
given corpus. The corpus contains V word types. Express a formula for esti-
mating P(w3|w1,w2), where w3 is a word which follows the bigram (w1,w2),
in terms of various n-gram counts and V. Use the notation c(w1,w2,w3) to
denote the number of times that trigram (w1,w2,w3) occurs in the corpus, and
so on for bigrams and unigrams.
3.7 We are given the following corpus, modified from the one in the chapter:
<s> I am Sam </s>
<s> Sam I am </s>
<s> I am Sam </s>
<s> I do not like green eggs and Sam </s>
If we use linear interpolation smoothing between a maximum-likelihood bi-
gram model and a maximum-likelihood unigram model with λ1 = 12 and λ2 =
1
2 , what is P(Sam|am)? Include <s> and </s> in your counts just like any
other token.
3.8 Write a program to compute unsmoothed unigrams and bigrams.
60 C HAPTER 3 • N- GRAM L ANGUAGE M ODELS
3.9 Run your n-gram program on two different small corpora of your choice (you
might use email text or newsgroups). Now compare the statistics of the two
corpora. What are the differences in the most common unigrams between the
two? How about interesting differences in bigrams?
3.10 Add an option to your program to generate random sentences.
3.11 Add an option to your program to compute the perplexity of a test set.
3.12 You are given a training set of 100 numbers that consists of 91 zeros and 1
each of the other digits 1-9. Now we see the following test set: 0 0 0 0 0 3 0 0
0 0. What is the unigram perplexity?
CHAPTER
But what makes classification so important is that language modeling can also
be viewed as classification: each word can be thought of as a class, and so predicting
the next word is classifying the context-so-far into a class for each next word. As
we’ll see, this intuition underlies large language models.
The algorithm for classification we introduce in this chapter, logistic regression,
is equally important, in a number of ways. First, logistic regression has a close
relationship with neural networks. As we will see in Chapter 6, a neural network
can be viewed as a series of logistic regression classifiers stacked on top of each
other. Second, logistic regression introduces ideas that are fundamental to neural
sigmoid networks and language models, like the sigmoid and softmax functions, the logit,
softmax and the key gradient descent algorithm for learning. Finally, logistic regression is
logit also one of the most important analytic tools in the social and natural sciences.
addition to the input and the set of output classes, we have a labeled training set
and a learning algorithm. We talked about training sets in Chapter 3 as a locus for
computing n-gram statistics. But in supervised machine learning the training set is
labeled, meaning that it contains a set of input observations, each observation asso-
ciated with the correct output (a ‘supervision signal’). We can generally refer to a
training set of m input/output pairs, where each input x is a text, in the case of text
classification, and each is hand-labeled with an associated class (the correct label):.
dot product In the rest of the book we’ll represent such sums using the dot product notation
from linear algebra. The dot product of two vectors a and b, written as a · b, is the
sum of the products of the corresponding elements of each vector. (Notice that we
represent vectors using the boldface notation b). Thus the following is an equivalent
formation to Eq. 4.2:
z = w·x+b (4.3)
But note that nothing in Eq. 4.3 forces z to be a legal probability, that is, to lie
between 0 and 1. In fact, since weights are real-valued, the output might even be
negative; z ranges from −∞ to ∞.
sigmoid To create a probability, we’ll pass z through the sigmoid function, σ (z). The
sigmoid function (named because it looks like an s) is also called the logistic func-
logistic tion, and gives logistic regression its name. The sigmoid has the following equation,
function
shown graphically in Fig. 4.1:
1 1
σ (z) = = (4.4)
1+e −z 1 + exp (−z)
(For the rest of the book, we’ll use the notation exp(x) to mean ex .) The sigmoid
has a number of advantages; it takes a real-valued number and maps it into the range
4.3 • C LASSIFICATION WITH L OGISTIC R EGRESSION 65
1
Figure 4.1 The sigmoid function σ (z) = 1+e −z takes a real value and maps it to the range
(0, 1). It is nearly linear around 0 but outlier values get squashed toward 0 or 1.
(0, 1), which is just what we want for a probability. Because it is nearly linear around
0 but flattens toward the ends, it tends to squash outlier values toward 0 or 1. And
it’s differentiable, which as we’ll see in Section 4.15 will be handy for learning.
We’re almost there. If we apply the sigmoid to the sum of the weighted features,
we get a number between 0 and 1. To make it a probability, we just need to make
sure that the two cases, P(y = 1) and P(y = 0), sum to 1. We can do this as follows:
P(y = 1) = σ (w · x + b)
1
=
1 + exp (−(w · x + b))
P(y = 0) = 1 − σ (w · x + b)
1
= 1−
1 + exp (−(w · x + b))
exp (−(w · x + b))
= (4.5)
1 + exp (−(w · x + b))
The sigmoid function has the property
1 − σ (x) = σ (−x) (4.6)
Let’s have some examples of applying logistic regression as a classifier for language
tasks.
x2=2
x3=1
It's hokey . There are virtually no surprises , and the writing is second-rate .
So why was it so enjoyable ? For one thing , the cast is
great . Another nice touch is the music . I was overcome with the urge to get off
the couch and start dancing . It sucked me in , and it'll do the same to you .
x4=3
x1=3 x5=0 x6=4.19
Figure 4.2 A sample mini test document showing the extracted features in the vector x.
Let’s assume for the moment that we’ve already learned a real-valued weight
for each of these features, and that the 6 weights corresponding to the 6 features
are [2.5, −5.0, −1.2, 0.5, 2.0, 0.7], while b = 0.1. (We’ll discuss in the next section
how the weights are learned.) The weight w1 , for example indicates how important
a feature the number of positive lexicon words (great, nice, enjoyable, etc.) is to
a positive sentiment decision, while w2 tells us the importance of negative lexicon
words. Note that w1 = 2.5 is positive, while w2 = −5.0, meaning that negative words
are negatively associated with a positive sentiment decision, and are about twice as
important as positive words.
Given these 6 features and the input review x, P(+|x) and P(−|x) can be com-
4.3 • C LASSIFICATION WITH L OGISTIC R EGRESSION 67
standardize standardize input values by centering them to result in a zero mean and a standard
z-score deviation of one (this transformation is sometimes called the z-score). That is, if µi
is the mean of the values of feature xi across the m observations in the input dataset,
and σi is the standard deviation of the values of features xi across the input dataset,
we can replace each feature xi by a new feature xi0 computed as follows:
v
m u X
1 X ( j) u 1 m ( j) 2
µi = xi σi = t xi − µi
m m
j=1 j=1
xi − µi
xi0 = (4.9)
σi
normalize Alternatively, we can normalize the input features values to lie between 0 and 1:
xi − min(xi )
xi0 = (4.10)
max(xi ) − min(xi )
Having input data with comparable range is useful when comparing values across
features. Data scaling is especially important in large neural networks, since it helps
speed up gradient descent.
For the first 3 test examples, then, we would be separately computing the pre-
dicted ŷ(i) as follows:
But it turns out that we can slightly modify our original equation Eq. 4.5 to do
this much more efficiently. We’ll use matrix arithmetic to assign a class to all the
examples with one matrix operation!
First, we’ll pack all the input feature vectors for each input x into a single input
matrix X, where each row i is a row vector consisting of the feature vector for in-
put example x(i) (i.e., the vector x(i) ). Assuming each example has f features and
4.4 • M ULTINOMIAL LOGISTIC REGRESSION 69
ŷ = σ (Xw + b) (4.13)
You should convince yourself that Eq. 4.13 computes the same thing as our for-loop
in Eq. 4.11. For example ŷ(1) , the first entry of the output vector y, will correctly be:
(1) (1) (1)
ŷ(1) = [x1 , x2 , ..., x f ] · [w1 , w2 , ..., w f ] + b (4.14)
Note that we had to reorder X and w from the order they appeared in in Eq. 4.5 to
make the multiplications come out properly. Here is Eq. 4.13 again with the shapes
shown:
ŷ = σ (X w + b)
(m × 1) (m × f ) ( f × 1) (m × 1) (4.15)
Modern compilers and compute hardware can compute this matrix operation very
efficiently, making the computation much faster, which becomes important when
training or testing on very large datasets.
Note by the way that we could have kept X and w in the original order (as
ŷ = σ (wX + b)) if we had chosen to define X differently as a matrix of column
vectors, one vector for each input example, instead of row vectors, and then it would
have shape [ f × m]. But we conventionally represent inputs as rows.
multiple classes). Let’s use the following representation: the output y for each input
x will be a vector of length K. If class c is the correct class, we’ll set yc = 1, and
set all the other elements of y to be 0, i.e., yc = 1 and y j = 0 ∀ j 6= c. A vector like
this y, with one value=1 and the rest 0, is called a one-hot vector. The job of the
classifier is to produce an estimate vector ŷ. For each class k, the value ŷk will be
the classifier’s estimate of the probability P(yk = 1|x).
4.4.1 Softmax
The multinomial logistic classifier uses a generalization of the sigmoid, called the
softmax softmax function, to compute p(yk = 1|x). The softmax function takes a vector
z = [z1 , z2 , ..., zK ] of K arbitrary values and maps them to a probability distribution,
with each value in the range [0,1], and all the values summing to 1. Like the sigmoid,
it is an exponential function.
For a vector z of dimensionality K, the softmax is defined as:
exp (zi )
softmax(zi ) = PK 1≤i≤K (4.16)
j=1 exp (z j )
exp (wk · x + bk )
P(yk = 1|x) = K
(4.18)
X
exp (w j · x + b j )
j=1
The form of Eq. 4.18 makes it seem that we would compute each output sep-
arately. Instead, it’s more common to set up the equation for more efficient com-
putation by modern vector processing hardware. We’ll do this by representing the
4.4 • M ULTINOMIAL LOGISTIC REGRESSION 71
set of K weight vectors as a weight matrix W and a bias vector b. Each row k of
W corresponds to the vector of weights wk . W thus has shape [K × f ], for K the
number of output classes and f the number of input features. The bias vector b has
one value for each of the K output classes. If we represent the weights in this way,
we can compute ŷ, the vector of output probabilities for each of the K classes, by a
single elegant equation:
ŷ = softmax(Wx + b) (4.19)
If you work out the matrix arithmetic, you can see that the estimated score of
the first output class ŷ1 (before we take the softmax) will correctly turn out to be
w1 · x + b1 .
One helpful interpretation of the weight matrix W is to see each row wk as a
prototype prototype of class k. The weight vector wk that is learned represents the class as
a kind of template. Since two vectors that are more similar to each other have a
higher dot product with each other, the dot product acts as a similarity function.
Logistic regression is thus learning an exemplar representation for each class, such
that incoming vectors are assigned the class k they are most similar to from the K
classes (Doumbouya et al., 2025).
Fig. 4.3 shows the difference between binary and multinomial logistic regression
by illustrating the weight vector versus weight matrix in the computation of the
output class probabilities.
Feature Definition
w5,+ w5,− w5,0
1 if “!” ∈ doc
f5 (x) 3.5 3.1 −5.3
0 otherwise
Because these feature weights are dependent both on the input text and the output
class, we sometimes make this dependence explicit and represent the features them-
selves as f (x, y): a function of both the input and the class. Using such a notation
f5 (x) above could be represented as three features f5 (x, +), f5 (x, −), and f5 (x, 0),
each of which has a single weight. We’ll use this kind of notation in our description
of the CRF in Chapter 17.
72 C HAPTER 4 • L OGISTIC R EGRESSION
Output y y^
sigmoid [scalar]
Weight vector w
[1⨉f]
Input feature x x1 x2 x3 … xf
vector [f ⨉1]
wordcount positive lexicon count of
=3 words = 1 “no” = 0
Figure 4.3 Binary versus multinomial logistic regression. Binary logistic regression uses a
single weight vector w, and has a scalar output ŷ. In multinomial logistic regression we have
K separate weight vectors corresponding to the K classes, all packed into a single weight
matrix W, and a vector output ŷ. We omit the biases from both figures for clarity.
We do this via a loss function that prefers the correct class labels of the train-
ing examples to be more likely. This is called conditional maximum likelihood
estimation: we choose the parameters w, b that maximize the log probability of
the true y labels in the training data given the observations x. The resulting loss
cross-entropy function is the negative log likelihood loss, generally called the cross-entropy loss.
loss
Let’s derive this loss function, applied to a single observation x. We’d like to
learn weights that maximize the probability of the correct label p(y|x). Since there
are only two discrete outcomes (1 or 0), this is a Bernoulli distribution, and we can
express the probability p(y|x) that our classifier produces for one observation as the
following (keeping in mind that if y = 1, Eq. 4.21 simplifies to ŷ; if y = 0, Eq. 4.21
simplifies to 1 − ŷ):
Now we take the log of both sides. This will turn out to be handy mathematically,
and doesn’t hurt us; whatever values maximize a probability will also maximize the
log of the probability:
log p(y|x) = log ŷ y (1 − ŷ)1−y
= y log ŷ + (1 − y) log(1 − ŷ) (4.22)
Eq. 4.22 describes a log likelihood that should be maximized. In order to turn this
into a loss function (something that we need to minimize), we’ll just flip the sign on
Eq. 4.22. The result is the cross-entropy loss LCE :
Let’s see if this loss function does the right thing for our example from Fig. 4.2. We
want the loss to be smaller if the model’s estimate is close to correct, and bigger if
the model is confused. So first let’s suppose the correct gold label for the sentiment
example in Fig. 4.2 is positive, i.e., y = 1. In this case our model is doing well, since
74 C HAPTER 4 • L OGISTIC R EGRESSION
from Eq. 4.8 it indeed gave the example a higher probability of being positive (.70)
than negative (.30). If we plug σ (w · x + b) = .70 and y = 1 into Eq. 4.24, the right
side of the equation drops out, leading to the following loss (we’ll use log to mean
natural log when the base is not specified):
LCE (ŷ, y) = −[y log σ (w · x + b) + (1 − y) log (1 − σ (w · x + b))]
= − [log σ (w · x + b)]
= − log(.70)
= .36
By contrast, let’s pretend instead that the example in Fig. 4.2 was actually negative,
i.e., y = 0 (perhaps the reviewer went on to say “But bottom line, the movie is
terrible! I beg you not to see it!”). In this case our model is confused and we’d want
the loss to be higher. Now if we plug y = 0 and 1 − σ (w · x + b) = .30 from Eq. 4.8
into Eq. 4.24, the left side of the equation drops out:
LCE (ŷ, y) = −[y log σ (w · x + b)+(1 − y) log (1 − σ (w · x + b))]
= − [log (1 − σ (w · x + b))]
= − log (.30)
= 1.2
Sure enough, the loss for the first classifier (.36) is less than the loss for the second
classifier (1.2).
Why does minimizing this negative log probability do what we want? A perfect
classifier would assign probability 1 to the correct outcome (y = 1 or y = 0) and
probability 0 to the incorrect outcome. That means if y equals 1, the higher ŷ is (the
closer it is to 1), the better the classifier; the lower ŷ is (the closer it is to 0), the
worse the classifier. If y equals 0, instead, the higher 1 − ŷ is (closer to 1), the better
the classifier. The negative log of ŷ (if the true y equals 1) or 1 − ŷ (if the true y
equals 0) is a convenient loss metric since it goes from 0 (negative log of 1, no loss)
to infinity (negative log of 0, infinite loss). This loss function also ensures that as
the probability of the correct answer is maximized, the probability of the incorrect
answer is minimized; since the two sum to one, any increase in the probability of the
correct answer is coming at the expense of the incorrect answer. It’s called the cross-
entropy loss, because Eq. 4.22 is also the formula for the cross-entropy between the
true probability distribution y and our estimated distribution ŷ.
Now we know what we want to minimize; in the next section, we’ll see how to
find the minimum.
How shall we find the minimum of this (or any) loss function? Gradient descent is
a method that finds a minimum of a function by figuring out in which direction (in
the space of the parameters θ ) the function’s slope is rising the most steeply, and
moving in the opposite direction. The intuition is that if you are hiking in a canyon
and trying to descend most quickly down to the river at the bottom, you might look
around yourself in all directions, find the direction where the ground is sloping the
steepest, and walk downhill in that direction.
convex For logistic regression, this loss function is conveniently convex. A convex func-
tion has at most one minimum; there are no local minima to get stuck in, so gradient
descent starting from any point is guaranteed to find the minimum. (By contrast,
the loss for multi-layer neural networks is non-convex, and gradient descent may
get stuck in local minima for neural network training and never find the global opti-
mum.)
Although the algorithm (and the concept of gradient) are designed for direction
vectors, let’s first consider a visualization of the case where the parameter of our
system is just a single scalar w, shown in Fig. 4.4.
Given a random initialization of w at some value w1 , and assuming the loss
function L happened to have the shape in Fig. 4.4, we need the algorithm to tell us
whether at the next iteration we should move left (making w2 smaller than w1 ) or
right (making w2 bigger than w1 ) to reach the minimum.
Loss
one step
of gradient
slope of loss at w1 descent
is negative
w1 wmin w
0 (goal)
Figure 4.4 The first step in iteratively finding the minimum of this loss function, by moving
w in the reverse direction from the slope of the function. Since the slope is negative, we need
to move w in a positive direction, to the right. Here superscripts are used for learning steps,
so w1 means the initial value of w (which is 0), w2 the value at the second step, and so on.
gradient The gradient descent algorithm answers this question by finding the gradient
of the loss function at the current point and moving in the opposite direction. The
gradient of a function of many variables is a vector pointing in the direction of the
greatest increase in a function. The gradient is a multi-variable generalization of the
slope, so for a function of one variable like the one in Fig. 4.4, we can informally
think of the gradient as the slope. The dotted line in Fig. 4.4 shows the slope of this
hypothetical loss function at point w = w1 . You can see that the slope of this dotted
line is negative. Thus to find the minimum, gradient descent tells us to go in the
opposite direction: moving w in a positive direction.
The magnitude of the amount to move in gradient descent is the value of the
d
learning rate slope dw L( f (x; w), y) weighted by a learning rate η. A higher (faster) learning
76 C HAPTER 4 • L OGISTIC R EGRESSION
rate means that we should move w more on each step. The change we make in our
parameter is the learning rate times the gradient (or the slope, in our single-variable
example):
d
wt+1 = wt − η L( f (x; w), y) (4.26)
dw
Now let’s extend the intuition from a function of one scalar variable w to many
variables, because we don’t just want to move left or right, we want to know where
in the N-dimensional space (of the N parameters that make up θ ) we should move.
The gradient is just such a vector; it expresses the directional components of the
sharpest slope along each of those N dimensions. If we’re just imagining two weight
dimensions (say for one weight w and one bias b), the gradient might be a vector with
two orthogonal components, each of which tells us how much the ground slopes in
the w dimension and in the b dimension. Fig. 4.5 shows a visualization of the value
of a 2-dimensional gradient vector taken at the red point.
In an actual logistic regression, the parameter vector w is much longer than 1 or
2, since the input feature vector x can be quite long, and we need a weight wi for
each xi . For each dimension/variable wi in w (plus the bias b), the gradient will have
a component that tells us the slope with respect to that variable. In each dimension
wi , we express the slope as a partial derivative ∂∂wi of the loss function. Essentially
we’re asking: “How much would a small change in that variable wi influence the
total loss function L?”
Formally, then, the gradient of a multi-variable function f is a vector in which
each component expresses the partial derivative of f with respect to one of the vari-
ables. We’ll use the inverted Greek delta symbol ∇ to refer to the gradient, and
represent ŷ as f (x; θ ) to make the dependence on θ more obvious:
∂
∂ w1 L( f (x; θ ), y)
∂ L( f (x; θ ), y)
∂ w2
..
∇L( f (x; θ ), y) = .
(4.27)
∂
∂ w L( f (x; θ ), y)
n
∂ b L( f (x; θ ), y)
∂
Cost(w,b)
b
w
Figure 4.5 Visualization of the gradient vector at the red point in two dimensions w and
b, showing a red arrow in the x-y plane pointing in the direction we will go to look for the
minimum: the opposite direction of the gradient (recall that the gradient points in the direction
of increase not decrease).
4.7 • G RADIENT D ESCENT 77
It turns out that the derivative of this function for one observation vector x is Eq. 4.30
(the interested reader can see Section 4.15 for the derivation of this equation):
∂ LCE (ŷ, y)
= [σ (w · x + b) − y]x j
∂wj
= (ŷ − y)x j (4.30)
Figure 4.6 The stochastic gradient descent algorithm. Step 1 (computing the loss) is used
mainly to report how well we are doing on the current tuple; we don’t need to compute the
loss in order to compute the gradient. The algorithm can terminate when it converges (when
the gradient norm < ), or when progress halts (for example when the loss starts going up on
a held-out set). Weights are initialized to 0 for logistic regression, but to small random values
for neural networks, as we’ll see in Chapter 6.
Let’s assume the initial weights and bias in θ 0 are all set to 0, and the initial learning
rate η is 0.1:
w1 = w2 = b = 0
η = 0.1
The single update step requires that we compute the gradient, multiplied by the
learning rate
In our mini example there are three parameters, so the gradient vector has 3 dimen-
sions, for w1 , w2 , and b. We can compute the first gradient as follows:
∂ LCE (ŷ,y)
∂ w1 (σ (w · x + b) − y)x1 (σ (0) − 1)x1 −0.5x1 −1.5
(ŷ,y)
∇w,b L = ∂ LCE
∂ w2 = (σ (w · x + b) − y)x2 = (σ (0) − 1)x2 = −0.5x2 = −1.0
∂ LCE (ŷ,y) σ (w · x + b) − y σ (0) − 1 −0.5 −0.5
∂b
Now that we have a gradient, we compute the new parameter vector θ 1 by moving
θ 0 in the opposite direction from the gradient:
w1 −1.5 .15
θ 1 = w2 − η −1.0 = .1
b −0.5 .05
So after one step of gradient descent, the weights have shifted to be: w1 = .15,
w2 = .1, and b = .05.
Note that this observation x happened to be a positive example. We would expect
that after seeing more negative examples with high counts of negative words, that
the weight w2 would shift to have a negative value.
4.7 • G RADIENT D ESCENT 79
Now the cost function for the mini-batch of m examples is the average loss for each
example:
m
1X
Cost(ŷ, y) = LCE (ŷ(i) , y(i) )
m
i=1
Xm
1
= − y(i) log σ (w · x(i) + b) + (1 − y(i) ) log 1 − σ (w · x(i) + b) (4.33)
m
i=1
The mini-batch gradient is the average of the individual gradients from Eq. 4.30:
m
∂Cost(ŷ, y) 1 Xh i
(i)
= σ (w · x(i) + b) − y(i) x j (4.34)
∂wj m
i=1
Instead of using the sum notation, we can more efficiently compute the gradient
in its matrix form, following the vectorization we saw on page 69, where we have
a matrix X of size [m × f ] representing the m inputs in the batch, and a vector y of
size [m × 1] representing the correct outputs:
80 C HAPTER 4 • L OGISTIC R EGRESSION
∂Cost(ŷ, y) 1
= (ŷ − y)| X
∂w m
1
= (σ (Xw + b) − y)| X (4.35)
m
The loss function for multinomial logistic regression generalizes the two terms in
Eq. 4.36 (one that is non-zero when y = 1 and one that is non-zero when y = 0) to
K terms. As we mentioned above, for multinomial regression we’ll represent both y
and ŷ as vectors. The true label y is a vector with K elements, each corresponding
to a class, with yc = 1 if the correct class is c, with all other elements of y being 0.
And our classifier will produce an estimate vector with K elements ŷ, each element
ŷk of which represents the estimated probability p(yk = 1|x).
The loss function for a single example x, generalizing from binary logistic re-
gression, is the sum of the logs of the K output classes, each weighted by the indi-
cator function yk (Eq. 4.37). This turns out to be just the negative log probability of
the correct class c (Eq. 4.38):
K
X
LCE (ŷ, y) = − yk log ŷk (4.37)
k=1
How did we get from Eq. 4.37 to Eq. 4.38? Because only one class (let’s call it c) is
the correct one, the vector y takes the value 1 only for this value of k, i.e., has yc = 1
and y j = 0 ∀ j 6= c. That means the terms in the sum in Eq. 4.37 will all be 0 except
for the term corresponding to the true class c. Hence the cross-entropy loss is simply
the log of the output probability corresponding to the correct class, and we therefore
negative log also call Eq. 4.38 the negative log likelihood loss.
likelihood loss
Of course for gradient descent we don’t need the loss, we need its gradient. The
gradient for a single example turns out to be very similar to the gradient for binary
logistic regression, (ŷ − y)x, that we saw in Eq. 4.30. Let’s consider one piece of the
gradient, the derivative for a single weight. For each class k, the weight of the ith
element of input x is wk,i . What is the partial derivative of the loss with respect to
wk,i ? This derivative turns out to be just the difference between the true value for the
class k (which is either 1 or 0) and the probability the classifier outputs for class k,
weighted by the value of the input xi corresponding to the ith element of the weight
4.9 • E VALUATION : P RECISION , R ECALL , F- MEASURE 81
We’ll return to this case of the gradient for softmax regression when we introduce
neural networks in Chapter 6, and at that time we’ll also discuss the derivation of
this gradient in equations Eq. 6.35–Eq. 6.43.
Figure 4.7 A confusion matrix for visualizing how well a binary classification system per-
forms against gold standard labels.
true positives
Precision =
true positives + false positives
recall Recall measures the percentage of items actually present in the input that were
correctly identified by the system. Recall is defined as
true positives
Recall =
true positives + false negatives
Precision and recall will help solve the problem with the useless “nothing is
pie” classifier. This classifier, despite having a fabulous accuracy of 99.99%, has
a terrible recall of 0 (since there are no true positives, and 100 false negatives, the
recall is 0/100). You should convince yourself that the precision at finding relevant
tweets is equally problematic. Thus precision and recall, unlike accuracy, emphasize
true positives: finding the things that we are supposed to be looking for.
There are many ways to define a single metric that incorporates aspects of both
F-measure precision and recall. The simplest of these combinations is the F-measure (van
Rijsbergen, 1975) , defined as:
(β 2 + 1)PR
Fβ =
β 2P + R
rocals:
n
HarmonicMean(a1 , a2 , a3 , a4 , ..., an ) = 1 1
(4.42)
a1 + a2 + a13 + ... + a1n
Harmonic mean is used because the harmonic mean of two values is closer to the
minimum of the two values than the arithmetic mean is. Thus it weighs the lower of
the two numbers more heavily, which is more conservative in this situation.
gold labels
urgent normal spam
8
urgent 8 10 1 precisionu=
8+10+1
system 60
output normal 5 60 50 precisionn=
5+60+50
200
spam 3 30 200 precisions=
3+30+200
recallu = recalln = recalls =
8 60 200
8+5+3 10+60+30 1+50+200
Figure 4.8 Confusion matrix for a three-class categorization task, showing for each pair of
classes (c1 , c2 ), how many documents from c1 were (in)correctly assigned to c2 .
But we’ll need to slightly modify our definitions of precision and recall. Con-
sider the sample confusion matrix for a hypothetical 3-way one-of email catego-
rization decision (urgent, normal, spam) shown in Fig. 4.8. The matrix shows, for
example, that the system mistakenly labeled one spam document as urgent, and we
have shown how to compute a distinct precision and recall value for each class. In
order to derive a single metric that tells us how well the system is doing, we can com-
macroaveraging bine these values in two ways. In macroaveraging, we compute the performance
microaveraging for each class, and then average over classes. In microaveraging, we collect the de-
cisions for all classes into a single confusion matrix, and then compute precision and
recall from that table. Fig. 4.9 shows the confusion matrix for each class separately,
and shows the computation of microaveraged and macroaveraged precision.
As the figure shows, a microaverage is dominated by the more frequent class (in
this case spam), since the counts are pooled. The macroaverage better reflects the
statistics of the smaller classes, and so is more appropriate when performance on all
the classes is equally important.
84 C HAPTER 4 • L OGISTIC R EGRESSION
macroaverage = .42+.52+.86
= .60
precision 3
Figure 4.9 Separate confusion matrices for the 3 classes from the previous figure, showing the pooled confu-
sion matrix and the microaveraged and macroaveraged precision.
The training and testing procedure for text classification follows what we saw with
language modeling (Section 3.2): we use the training set to train the model, then use
development the development test set (also called a devset) to perhaps tune some parameters,
test set
devset and in general decide what the best model is. Once we come up with what we think
is the best model, we run it on the (hitherto unseen) test set to report its performance.
While the use of a devset avoids overfitting the test set, having a fixed train-
ing set, devset, and test set creates another problem: in order to save lots of data
for training, the test set (or devset) might not be large enough to be representative.
Wouldn’t it be better if we could somehow use all our data for training and still use
cross-validation all our data for test? We can do this by cross-validation.
In cross-validation, we choose a number k, and partition our data into k disjoint
folds subsets called folds. Now we choose one of those k folds as a test set, train our
classifier on the remaining k − 1 folds, and then compute the error rate on the test
set. Then we repeat with another fold as the test set, again training on the other k − 1
folds. We do this sampling process k times and average the test set error rate from
these k runs to get an average error rate. If we choose k = 10, we would train 10
different models (each on 90% of our data), test the model 10 times, and average
10-fold these 10 values. This is called 10-fold cross-validation.
cross-validation
The only problem with cross-validation is that because all the data is used for
testing, we need the whole corpus to be blind; we can’t examine any of the data
to suggest possible features and in general see what’s going on, because we’d be
peeking at the test set, and such cheating would cause us to overestimate the perfor-
mance of our system. However, looking at the corpus to understand what’s going
on is important in designing NLP systems! What to do? For this reason, it is com-
mon to create a fixed training set and test set, then do 10-fold cross-validation inside
the training set, but compute error rate the normal way in the test set, as shown in
Fig. 4.10.
4.11 • S TATISTICAL S IGNIFICANCE T ESTING 85
We would like to know if δ (x) > 0, meaning that our logistic regression classifier
effect size has a higher F1 than our naive Bayes classifier on x. δ (x) is called the effect size; a
bigger δ means that A seems to be way better than B; a small δ means A seems to
be only a little better.
Why don’t we just check if δ (x) is positive? Suppose we do, and we find that
the F1 score of A is higher than B’s by .04. Can we be certain that A is better? We
cannot! That’s because A might just be accidentally better than B on this particular x.
We need something more: we want to know if A’s superiority over B is likely to hold
again if we checked another test set x0 , or under some other set of circumstances.
In the paradigm of statistical hypothesis testing, we test this by formalizing two
hypotheses.
H0 : δ (x) ≤ 0
H1 : δ (x) > 0 (4.45)
null hypothesis The hypothesis H0 , called the null hypothesis, supposes that δ (x) is actually nega-
tive or zero, meaning that A is not better than B. We would like to know if we can
confidently rule out this hypothesis, and instead support H1 , that A is better.
We do this by creating a random variable X ranging over all test sets. Now we
ask how likely is it, if the null hypothesis H0 was correct, that among these test sets
86 C HAPTER 4 • L OGISTIC R EGRESSION
we would encounter the value of δ (x) that we found, if we repeated the experiment
p-value a great many times. We formalize this likelihood as the p-value: the probability,
assuming the null hypothesis H0 is true, of seeing the δ (x) that we saw or one even
greater
P(δ (X) ≥ δ (x)|H0 is true) (4.46)
So in our example, this p-value is the probability that we would see δ (x) assuming
A is not better than B. If δ (x) is huge (let’s say A has a very respectable F1 of .9
and B has a terrible F1 of only .2 on x), we might be surprised, since that would be
extremely unlikely to occur if H0 were in fact true, and so the p-value would be low
(unlikely to have such a large δ if A is in fact not better than B). But if δ (x) is very
small, it might be less surprising to us even if H0 were true and A is not really better
than B, and so the p-value would be higher.
A very small p-value means that the difference we observed is very unlikely
under the null hypothesis, and we can reject the null hypothesis. What counts as very
small? It is common to use values like .05 or .01 as the thresholds. A value of .01
means that if the p-value (the probability of observing the δ we saw assuming H0 is
true) is less than .01, we reject the null hypothesis and assume that A is indeed better
statistically
significant than B. We say that a result (e.g., “A is better than B”) is statistically significant if
the δ we saw has a probability that is below the threshold and we therefore reject
this null hypothesis.
How do we compute this probability we need for the p-value? In NLP we gen-
erally don’t use simple parametric tests like t-tests or ANOVAs that you might be
familiar with. Parametric tests make assumptions about the distributions of the test
statistic (such as normality) that don’t generally hold in our cases. So in NLP we
usually use non-parametric tests based on sampling: we artificially create many ver-
sions of the experimental setup. For example, if we had lots of different test sets x0
we could just measure all the δ (x0 ) for all the x0 . That gives us a distribution. Now
we set a threshold (like .01) and if we see in this distribution that 99% or more of
those deltas are smaller than the delta we observed, i.e., that p-value(x)—the proba-
bility of seeing a δ (x) as big as the one we saw—is less than .01, then we can reject
the null hypothesis and agree that δ (x) was a sufficiently surprising difference and
A is really a better algorithm than B.
There are two common non-parametric tests used in NLP: approximate ran-
approximate domization (Noreen, 1989) and the bootstrap test. We will describe bootstrap
randomization
below, showing the paired version of the test, which again is most common in NLP.
paired Paired tests are those in which we compare two sets of observations that are aligned:
each observation in one set can be paired with an observation in another. This hap-
pens naturally when we are comparing the performance of two systems on the same
test set; we can pair the performance of system A on an individual observation xi
with the performance of system B on the same xi .
Consider a tiny text classification example with a test set x of 10 documents. The
first row of Fig. 4.11 shows the results of two classifiers (A and B) on this test set.
Each document is labeled by one of the four possibilities (A and B both right, both
wrong, A right and B wrong, A wrong and B right). A slash through a letter ( B)
means that that classifier got the answer wrong. On the first document both A and
B get the correct class (AB), while on the second document A got it right but B got
it wrong (A B). If we assume for simplicity that our metric is accuracy, A has an
accuracy of .70 and B of .50, so δ (x) is .20.
Now we create a large number b (perhaps 105 ) of virtual test sets x(i) , each of size
n = 10. Fig. 4.11 shows a couple of examples. To create each virtual test set x(i) , we
repeatedly (n = 10 times) select a cell from row x with replacement. For example, to
create the first cell of the first virtual test set x(1) , if we happened to randomly select
the second cell of the x row, we would copy the value A B into our new cell, and
move on to create the second cell of x(1) , each time sampling (randomly choosing)
from the original x with replacement.
1 2 3 4 5 6 7 8 9 10 A% B% δ ()
x AB AB AB AB AB AB AB AB AB AB
.70 .50 .20
x(1) AB AB AB AB AB AB AB AB AB AB .60 .60 .00
x(2) AB AB AB AB AB AB AB AB AB AB .60 .70 -.10
...
x(b)
Figure 4.11 The paired bootstrap test: Examples of b pseudo test sets x(i) being created
from an initial true test set x. Each pseudo test set is created by sampling n = 10 times with
replacement; thus an individual sample is a single cell, a document with its gold label and
the correct or incorrect performance of classifiers A and B. Of course real test sets don’t have
only 10 examples, and b needs to be large as well.
Now that we have the b test sets, providing a sampling distribution, we can do
statistics on how often A has an accidental advantage. There are various ways to
compute this advantage; here we follow the version laid out in Berg-Kirkpatrick
et al. (2012). Assuming H0 (A isn’t better than B), we would expect that δ (X),
estimated over many test sets, would be zero or negative; a much higher value would
be surprising, since H0 specifically assumes A isn’t better than B. To measure exactly
how surprising our observed δ (x) is, we would in other circumstances compute the
p-value by counting over many test sets how often δ (x(i) ) exceeds the expected zero
value by δ (x) or more:
b
1 X (i)
p-value(x) = 1 δ (x ) − δ (x) ≥ 0
b
i=1
(We use the notation 1(x) to mean “1 if x is true, and 0 otherwise”.) However,
although it’s generally true that the expected value of δ (X) over many test sets,
(again assuming A isn’t better than B) is 0, this isn’t true for the bootstrapped test
sets we created. That’s because we didn’t draw these samples from a distribution
with 0 mean; we happened to create them from the original test set x, which happens
to be biased (by .20) in favor of A. So to measure how surprising is our observed
δ (x), we actually compute the p-value by counting over many test sets how often
88 C HAPTER 4 • L OGISTIC R EGRESSION
b
1 X (i)
p-value(x) = 1 δ (x ) − δ (x) ≥ δ (x)
b
i=1
b
1X
= 1 δ (x(i) ) ≥ 2δ (x) (4.47)
b
i=1
So if for example we have 10,000 test sets x(i) and a threshold of .01, and in only 47
of the test sets do we find that A is accidentally better δ (x(i) ) ≥ 2δ (x), the resulting
p-value of .0047 is smaller than .01, indicating that the delta we found, δ (x) is indeed
sufficiently surprising and unlikely to have happened by accident, and we can reject
the null hypothesis and conclude A is better than B.
Figure 4.12 A version of the paired bootstrap algorithm after Berg-Kirkpatrick et al.
(2012).
The full algorithm for the bootstrap is shown in Fig. 4.12. It is given a test set
x, a number of samples b, and counts the percentage of the b bootstrap test sets in
which δ (x(i) ) > 2δ (x). This percentage then acts as a one-sided empirical p-value.
In other tasks classifiers may lead to both representational harms and other
harms, such as silencing. For example the important text classification task of tox-
toxicity icity detection is the task of detecting hate speech, abuse, harassment, or other
detection
kinds of toxic language. While the goal of such classifiers is to help reduce soci-
etal harm, toxicity classifiers can themselves cause harms. For example, researchers
have shown that some widely used toxicity classifiers incorrectly flag as being toxic
sentences that are non-toxic but simply mention identities like women (Park et al.,
2018), blind people (Hutchinson et al., 2020) or gay people (Dixon et al., 2018;
Dias Oliva et al., 2021), or simply use linguistic features characteristic of varieties
like African-American Vernacular English (Sap et al. 2019, Davidson et al. 2019).
Such false positive errors could lead to the silencing of discourse by or about these
groups.
These model problems can be caused by biases or other problems in the training
data; in general, machine learning systems replicate and even amplify the biases
in their training data. But these problems can also be caused by the labels (for
example due to biases in the human labelers), by the resources used (like lexicons,
or model components like pretrained embeddings), or even by model architecture
(like what the model is trained to optimize). While the mitigation of these biases
(for example by carefully considering the training data sources) is an important area
of research, we currently don’t have general solutions. For this reason it’s important,
when introducing any NLP model, to study these kinds of factors and make them
model card clear. One way to do this is by releasing a model card (Mitchell et al., 2019) for
each version of a model. A model card documents a machine learning model with
information like:
• training algorithms and parameters
• training data sources, motivation, and preprocessing
• evaluation data sources, motivation, and preprocessing
• intended use and users
• model performance across different demographic or other groups and envi-
ronmental situations
ciated with negative sentiment, or if negative reviews of movies are more likely to
discuss the cinematography. However, in doing so it’s necessary to control for po-
tential confounds: other factors that might influence sentiment (the movie genre, the
year it was made, perhaps the length of the review in words). Or we might be study-
ing the relationship between NLP-extracted linguistic features and non-linguistic
outcomes (hospital readmissions, political outcomes, or product sales), but need to
control for confounds (the age of the patient, the county of voting, the brand of the
product). In such cases, logistic regression allows us to test whether some feature is
associated with some outcome above and beyond the effect of other features.
There is a problem with learning weights that make the model perfectly match the
training data. If a feature is perfectly predictive of the outcome because it happens
to only occur in one class, it will be assigned a very high weight. The weights for
features will attempt to perfectly fit details of the training set, in fact too perfectly,
modeling noisy factors that just accidentally correlate with the class. This problem is
overfitting called overfitting. A good model should be able to generalize well from the training
generalize data to the unseen test set, but a model that overfits will have poor generalization.
regularization To avoid overfitting, a new regularization term R(θ ) is added to the loss func-
tion in Eq. 4.25, resulting in the following loss for a batch of m examples (slightly
rewritten from Eq. 4.25 to be maximizing log probability rather than minimizing
loss, and removing the m1 term which doesn’t affect the argmax):
m
X
θ̂ = argmax log P(y(i) |x(i) ) − αR(θ ) (4.48)
θ i=1
The new regularization term R(θ ) is used to penalize large weights. Thus a setting of
the weights that matches the training data perfectly— but uses many weights with
high values to do so—will be penalized more than a setting that matches the data
a little less well, but does so using smaller weights. The higher the regularization
strength parameter α, the lower the model’s weights will be, reducing its reliance on
the training data.
There are two common ways to compute this regularization term R(θ ). L2 reg-
L2
regularization ularization is a quadratic function of the weight values, named because it uses the
(square of the) L2 norm of the weight values. The L2 norm, ||θ ||2 , is the same as
the Euclidean distance of the vector θ from the origin. If θ consists of n weights,
then:
n
X
R(θ ) = ||θ ||22 = θ j2 (4.49)
j=1
4.14 • A DVANCED : R EGULARIZATION 91
L1
regularization L1 regularization is a linear function of the weight values, named after the L1 norm
||W ||1 , the sum of the absolute values of the weights, or Manhattan distance (the
Manhattan distance is the distance you’d have to walk between two points in a city
with a street grid like New York):
n
X
R(θ ) = ||θ ||1 = |θi | (4.51)
i=1
If we multiply each weight by a Gaussian prior on the weight, we are thus maximiz-
ing the following constraint:
m n
!
Y
(i) (i)
Y 1 (θ j − µ j )2
θ̂ = argmax P(y |x ) × q exp − (4.54)
θ i=1 j=1 2πσ 2 2σ 2j
j
d 1
ln(x) = (4.56)
dx x
Second, the (very elegant) derivative of the sigmoid:
dσ (z)
= σ (z)(1 − σ (z)) (4.57)
dz
chain rule Finally, the chain rule of derivatives. Suppose we are computing the derivative
of a composite function f (x) = u(v(x)). The derivative of f (x) is the derivative of
u(x) with respect to v(x) times the derivative of v(x) with respect to x:
df du dv
= · (4.58)
dx dv dx
First, we want to know the derivative of the loss function with respect to a single
weight w j (we’ll need to compute it for each weight, and for the bias):
∂ LCE ∂
= − [y log σ (w · x + b) + (1 − y) log (1 − σ (w · x + b))]
∂wj ∂wj
∂ ∂
= − y log σ (w · x + b) + (1 − y) log [1 − σ (w · x + b)]
∂wj ∂wj
(4.59)
Next, using the chain rule, and relying on the derivative of log:
∂ LCE y ∂ 1−y ∂
= − σ (w · x + b) − [1 − σ (w · x + b)]
∂wj σ (w · x + b) ∂ w j 1 − σ (w · x + b) ∂ w j
(4.60)
Rearranging terms:
∂ LCE y 1−y ∂
= − − σ (w · x + b)
∂wj σ (w · x + b) 1 − σ (w · x + b) ∂ w j
And now plugging in the derivative of the sigmoid, and using the chain rule one
more time, we end up with Eq. 4.61:
∂ LCE y − σ (w · x + b) ∂ (w · x + b)
= − σ (w · x + b)[1 − σ (w · x + b)]
∂wj σ (w · x + b)[1 − σ (w · x + b)] ∂wj
y − σ (w · x + b)
= − σ (w · x + b)[1 − σ (w · x + b)]x j
σ (w · x + b)[1 − σ (w · x + b)]
= −[y − σ (w · x + b)]x j
= [σ (w · x + b) − y]x j (4.61)
4.16 • S UMMARY 93
4.16 Summary
This chapter introduced the logistic regression model of classification.
• Logistic regression is a supervised machine learning classifier that extracts
real-valued features from the input, multiplies each by a weight, sums them,
and passes the sum through a sigmoid function to generate a probability. A
threshold is used to make a decision.
• Logistic regression can be used with two classes (e.g., positive and negative
sentiment) or with multiple classes (multinomial logistic regression, for ex-
ample for n-ary text classification, part-of-speech labeling, etc.).
• Multinomial logistic regression uses the softmax function to compute proba-
bilities.
• The weights (vector w and bias b) are learned from a labeled training set via a
loss function, such as the cross-entropy loss, that must be minimized.
• Minimizing this loss function is a convex optimization problem, and iterative
algorithms like gradient descent are used to find the optimal weights.
• Regularization is used to avoid overfitting.
• Logistic regression is also one of the most useful analytic tools, because of its
ability to transparently study the importance of individual features.
Historical Notes
Logistic regression was developed in the field of statistics, where it was used for
the analysis of binary data by the 1960s, and was particularly common in medicine
(Cox, 1969). Starting in the late 1970s it became widely used in linguistics as one
of the formal foundations of the study of linguistic variation (Sankoff and Labov,
1979).
Nonetheless, logistic regression didn’t become common in natural language pro-
cessing until the 1990s, when it seems to have appeared simultaneously from two
directions. The first source was the neighboring fields of information retrieval and
speech processing, both of which had made use of regression, and both of which
lent many other statistical techniques to NLP. Indeed a very early use of logistic
regression for document routing was one of the first NLP applications to use (LSI)
embeddings as word representations (Schütze et al., 1995).
At the same time in the early 1990s logistic regression was developed and ap-
maximum
entropy plied to NLP at IBM Research under the name maximum entropy modeling or
maxent (Berger et al., 1996), seemingly independent of the statistical literature. Un-
der that name it was applied to language modeling (Rosenfeld, 1996), part-of-speech
tagging (Ratnaparkhi, 1996), parsing (Ratnaparkhi, 1997), coreference resolution
(Kehler, 1997b), and text classification (Nigam et al., 1999).
There are a variety of sources covering the many kinds of text classification
tasks. For sentiment analysis see Pang and Lee (2008), and Liu and Zhang (2012).
Stamatatos (2009) surveys authorship attribute algorithms. On language identifica-
tion see Jauhiainen et al. (2019); Jaech et al. (2016) is an important early neural
system. The task of newswire indexing was often used as a test case for text classi-
fication algorithms, based on the Reuters-21578 collection of newswire articles.
See Manning et al. (2008) and Aggarwal and Zhai (2012) on text classification;
classification in general is covered in machine learning textbooks (Hastie et al. 2001,
94 C HAPTER 4 • L OGISTIC R EGRESSION
Exercises
CHAPTER
5 Embeddings
The asphalt that Los Angeles is famous for occurs mainly on its freeways. But
in the middle of the city is another patch of asphalt, the La Brea tar pits, and this
asphalt preserves millions of fossil bones from the last of the Ice Ages of the Pleis-
tocene Epoch. One of these fossils is the Smilodon, or saber-toothed tiger, instantly
recognizable by its long canines. Five million years ago or so, a completely different
saber-tooth tiger called Thylacosmilus lived
in Argentina and other parts of South Amer-
ica. Thylacosmilus was a marsupial whereas
Smilodon was a placental mammal, but Thy-
lacosmilus had the same long upper canines
and, like Smilodon, had a protective bone
flange on the lower jaw. The similarity of
these two mammals is one of many examples
of parallel or convergent evolution, in which particular contexts or environments
lead to the evolution of very similar structures in different species (Gould, 1980).
The role of context is also important in the similarity of a less biological kind
of organism: the word. Words that occur in similar contexts tend to have similar
meanings. This link between similarity in how words are distributed and similarity
distributional
hypothesis in what they mean is called the distributional hypothesis. The hypothesis was
first formulated in the 1950s by linguists like Joos (1950), Harris (1954), and Firth
(1957), who noticed that words which are synonyms (like oculist and eye-doctor)
tended to occur in the same environment (e.g., near words like eye or examined)
with the amount of meaning difference between two words “corresponding roughly
to the amount of difference in their environments” (Harris, 1954, p. 157).
embeddings In this chapter we introduce embeddings, vector representations of the meaning
of words that are learned directly from word distributions in texts. Embeddings lie
at the heart of large language models and other modern applications. The static em-
beddings we introduce here underlie the more powerful dynamic or contextualized
embeddings like BERT that we will see in Chapter 10 and Chapter 8.
The linguistic field that studies embeddings and their meanings is called vector
vector semantics. Embeddings are also the first example in this book of representation
semantics
representation
learning learning, automatically learning useful representations of the input text. Finding
such self-supervised ways to learn representations of language, instead of creat-
ing representations by hand via feature engineering, is an important principle of
modern NLP (Bengio et al., 2013).
96 C HAPTER 5 • E MBEDDINGS
identical to a sense of another word, or nearly identical, we say the two senses of
synonym those two words are synonyms. Synonyms include such pairs as
couch/sofa vomit/throw up filbert/hazelnut car/automobile
A more formal definition of synonymy (between words rather than senses) is that
two words are synonymous if they are substitutable for one another in any sentence
without changing the truth conditions of the sentence, the situations in which the
sentence would be true.
While substitutions between some pairs of words like car / automobile or wa-
ter / H2 O are truth preserving, the words are still not identical in meaning. Indeed,
probably no two words are absolutely identical in meaning. One of the fundamental
principle of tenets of semantics, called the principle of contrast (Girard 1718, Bréal 1897, Clark
contrast
1987), states that a difference in linguistic form is always associated with some dif-
ference in meaning. For example, the word H2 O is used in scientific contexts and
would be inappropriate in a hiking guide—water would be more appropriate— and
this genre difference is part of the meaning of the word. In practice, the word syn-
onym is therefore used to describe a relationship of approximate or rough synonymy.
Word Similarity While words don’t have many synonyms, most words do have
lots of similar words. Cat is not a synonym of dog, but cats and dogs are certainly
similar words. In moving from synonymy to similarity, it will be useful to shift from
talking about relations between word senses (like synonymy) to relations between
words (like similarity). Dealing with words avoids having to commit to a particular
representation of word senses, which will turn out to simplify our task.
similarity The notion of word similarity is very useful in larger semantic tasks. Knowing
how similar two words are can help in computing how similar the meaning of two
phrases or sentences are, a very important component of tasks like question answer-
ing, paraphrasing, and summarization. One way of getting values for word similarity
is to ask humans to judge how similar one word is to another. A number of datasets
have resulted from such experiments. For example the SimLex-999 dataset (Hill
et al., 2015) gives values on a scale from 0 to 10, like the examples below, which
range from near-synonyms (vanish, disappear) to pairs that scarcely seem to have
anything in common (hole, agreement):
vanish disappear 9.8
belief impression 5.95
muscle bone 3.65
modest flexible 0.98
hole agreement 0.3
Word Relatedness The meaning of two words can be related in ways other than
relatedness similarity. One such class of connections is called word relatedness (Budanitsky
association and Hirst, 2006), also traditionally called word association in psychology.
Consider the meanings of the words coffee and cup. Coffee is not similar to cup;
they share practically no features (coffee is a plant or a beverage, while a cup is a
manufactured object with a particular shape). But coffee and cup are clearly related;
they are associated by co-participating in an everyday event (the event of drinking
coffee out of a cup). Similarly scalpel and surgeon are not similar but are related
eventively (a surgeon tends to make use of a scalpel).
One common kind of relatedness between words is if they belong to the same
semantic field semantic field. A semantic field is a set of words which cover a particular semantic
domain and bear structured relations with each other. For example, words might be
98 C HAPTER 5 • E MBEDDINGS
related by being in the semantic field of hospitals (surgeon, scalpel, nurse, anes-
thetic, hospital), restaurants (waiter, menu, plate, food, chef), or houses (door, roof,
topic models kitchen, family, bed). Semantic fields are also related to topic models, like Latent
Dirichlet Allocation, LDA, which apply unsupervised learning on large sets of texts
to induce sets of associated words from text. Semantic fields and topic models are
very useful tools for discovering topical structure in documents.
In Appendix G we’ll introduce more relations between senses like hypernymy
or IS-A, antonymy (opposites) and meronymy (part-whole relations).
connotations Connotation Finally, words have affective meanings or connotations. The word
connotation has different meanings in different fields, but here we use it to mean the
aspects of a word’s meaning that are related to a writer or reader’s emotions, senti-
ment, opinions, or evaluations. For example some words have positive connotations
(wonderful) while others have negative connotations (dreary). Even words whose
meanings are similar in other ways can vary in connotation; consider the difference
in connotations between fake, knockoff, forgery, on the one hand, and copy, replica,
reproduction on the other, or innocent (positive connotation) and naive (negative
connotation). Some words describe positive evaluation (great, love) and others neg-
ative evaluation (terrible, hate). Positive or negative evaluation language is called
sentiment sentiment, as we saw in Appendix K, and word sentiment plays a role in impor-
tant tasks like sentiment analysis, stance detection, and applications of NLP to the
language of politics and consumer reviews.
Early work on affective meaning (Osgood et al., 1957) found that words varied
along three important dimensions of affective meaning:
valence: the pleasantness of the stimulus
arousal: the intensity of emotion provoked by the stimulus
dominance: the degree of control exerted by the stimulus
Thus words like happy or satisfied are high on valence, while unhappy or an-
noyed are low on valence. Excited is high on arousal, while calm is low on arousal.
Controlling is high on dominance, while awed or influenced are low on dominance.
Each word is thus represented by three numbers, corresponding to its value on each
of the three dimensions:
Valence Arousal Dominance
courageous 8.0 5.5 7.4
music 7.7 5.6 6.5
heartbreak 2.5 5.7 3.6
cub 6.7 4.0 4.2
Osgood et al. (1957) noticed that in using these 3 numbers to represent the
meaning of a word, the model was representing each word as a point in a three-
dimensional space, a vector whose three dimensions corresponded to the word’s
rating on the three scales. This revolutionary idea that word meaning could be rep-
resented as a point in space (e.g., that part of the meaning of heartbreak can be
represented as the point [2.5, 5.7, 3.6]) was the first expression of the vector seman-
tics models that we introduce next.
us model many of the aspects of word meaning we saw in the previous section. The
roots of the model lie in the 1950s when two big ideas converged: Osgood’s 1957
idea mentioned above to use a point in three-dimensional space to represent the
connotation of a word, and the proposal by linguists like Joos (1950), Harris (1954),
and Firth (1957) to define the meaning of a word by its distribution in language
use, meaning its neighboring words or grammatical environments. Their idea was
that two words that occur in very similar distributions (whose neighboring words are
similar) have similar meanings.
For example, suppose you didn’t know the meaning of the word ongchoi (a re-
cent borrowing from Cantonese) but you see it in the following contexts:
(5.1) Ongchoi is delicious sauteed with garlic.
(5.2) Ongchoi is superb over rice.
(5.3) ...ongchoi leaves with salty sauces...
And suppose that you had seen many of these context words in other contexts:
(5.4) ...spinach sauteed with garlic over rice...
(5.5) ...chard stems and leaves are delicious...
(5.6) ...collard greens and other salty leafy greens
The fact that ongchoi occurs with words like rice and garlic and delicious and
salty, as do words like spinach, chard, and collard greens might suggest that ongchoi
is a leafy green similar to these other leafy greens.1 We can implement the same
intuition computationally by just counting words in the context of ongchoi.
200-dimensional space into a 2-dimensional space. Note that the nearest neighbors
of sweet are semantically related words like honey, candy, juice, chocolate. This idea
that similar words are near each other in high-dimensional space is an important
that offers enormous power to language models and other NLP applications. For
example the sentiment classifiers of Chapter 4 depend on the same words appearing
in the training and test sets. But by representing words as embeddings, a classifier
can assign sentiment as long as it sees some words with similar meanings. And as
we’ll see, vector semantic models like the ones showed in Fig. 5.1 can be learned
automatically from text without supervision.
In this chapter we’ll begin with a simple pedagogical model of embeddings in
which the meaning of a word is defined by a vector with the counts of nearby words.
We introduce this model as a helpful way to understand the concept of vectors and
what it means for a vector to be a representation of word meaning, but more sophis-
ticated variants like the tf-idf model we will introduce in Chapter 11 are important
methods you should understand. We will see that this method results in very long
vectors that are sparse, i.e. mostly zeros (since most words simply never occur in the
context of others). We’ll then introduce the word2vec model family for constructing
short, dense vectors that have even more useful semantic properties.
We’ll also introduce the cosine, the standard way to use embeddings to com-
pute semantic similarity, between two words, two sentences, or two documents, an
important tool in practical applications.
context co-occurrence matrix is very large, because for each word in the vocabulary
(since |V |) we have to count how often it occurs with every other word in the vo-
cabulary, hence dimensionality |V | × |V |. Let’s therefore instead sketch the process
on a smaller scale. Imagine that we are going to look at only the 4 words, and only
consider the following 3 context words: a, computer, and pie. Furthermore let’s
assume we only count occurrences in the mini-corpus above.
So before looking at Fig. 5.2, compute by hand the counts for these 3 context
words for the four words cherry, strawberry, digital, and information.
a computer pie
cherry 1 0 1
strawberry 0 0 2
digital 0 1 0
information 1 1 0
Figure 5.2 Co-occurrence vectors for four words with counts from the 4 windows above,
showing just 3 of the potential context word dimensions. The vector for cherry is outlined in
red. Note that a real vector would have vastly more dimensions and thus be even sparser.
Hopefully your count matches what is shown in Fig. 5.2, so that each cell repre-
sents the number of times a particular word (defined by the row) occurs in a partic-
ular context (defined by the word column).
Each row, then, is a vector representing a word. To review some basic linear
vector algebra, a vector is, at heart, just a list or array of numbers. So cherry is represented
as the list [1,0,1] (the first row vector in Fig. 5.2) and information is represented as
the list [1,1,0] (the fourth row vector).
vector space A vector space is a collection of vectors, and is characterized by its dimension.
dimension Vectors in a 3-dimensional vector space have an element for each dimension of the
space. We will loosely refer to a vector in a 3-dimensional space as a 3-dimensional
vector, with one element along each dimension. In the example in Fig. 5.2, we’ve
chosen to make the document vectors of dimension 3, just so they fit on the page; in
real term-document matrices, the document vectors would have dimensionality |V |,
the vocabulary size.
The ordering of the numbers in a vector space indicates the different dimensions
on which documents vary. The third dimension for all these vectors corresponds
to the number of times pie occurs in the context. The second dimension for all of
them corresponds to the number of times the word computer occurs. Notice that
the vectors for information and digital have the same value (1) for this “computer”
dimension.
In reality, we don’t compute word vectors on a single context window. Instead,
we compute them over an entire corpus. Let’s see what some real counts look like.
Let’s look at some vectors computed in this way. Fig. 5.3 shows a subset of the
word-word co-occurrence matrix for these four words, where, again because it’s
impossible to visualize all |V | possible context words on the page of this textbook,
we show a subset of 6 of the dimensions, with counts computed from the Wikipedia
corpus (Davies, 2015).
Note in Fig. 5.3 that the two words cherry and strawberry are more similar to
each other (both pie and sugar tend to occur in their window) than they are to other
words like digital; conversely, digital and information are more similar to each other
than, say, to strawberry.
We can think of the vector for a document as a point in |V |-dimensional space;
thus the documents in Fig. 5.3 are points in 3-dimensional space. Fig. 5.4 shows a
spatial visualization.
102 C HAPTER 5 • E MBEDDINGS
4000
computer information
3000 [3982,3325]
digital
2000 [1683,1670]
1000
Note that |V |, the dimensionality of the vector, is generally the size of the vo-
cabulary, often between 10,000 and 50,000 words (using the most frequent words
in the training corpus; keeping words after about the most frequent 50,000 or so is
generally not helpful). Since most of these numbers are zero these are sparse vector
representations; there are efficient algorithms for storing and computing with sparse
matrices.
It’s also possible to applying various kinds of weighting functions to the counts
in these cells. The most popular such weighting is tf-idf, which we’ll introduce in
Chapter 11, but there have historically been a wide variety of other weightings.
Now that we have some intuitions, let’s move on to examine the details of com-
puting word similarity.
The dot product acts as a similarity metric because it will tend to be high just when
the two vectors have large values in the same dimensions. Alternatively, vectors that
5.4 • C OSINE FOR MEASURING SIMILARITY 103
The dot product is higher if a vector is longer, with higher values in each dimension.
More frequent words have longer vectors, since they tend to co-occur with more
words and have higher co-occurrence values with each of them. The raw dot product
thus will be higher for frequent words. But this is a problem; we’d like a similarity
metric that tells us how similar two words are regardless of their frequency.
We modify the dot product to normalize for the vector length by dividing the
dot product by the lengths of each of the two vectors. This normalized dot product
turns out to be the same as the cosine of the angle between the two vectors, following
from the definition of the dot product between two vectors a and b:
a · b = |a||b| cos θ
a·b
= cos θ (5.9)
|a||b|
cosine The cosine similarity metric between two vectors v and w thus can be computed as:
N
X
vi wi
v·w i=1
cosine(v, w) = =v v (5.10)
|v||w| u
uXN u N
uX
t v2 t w2
i i
i=1 i=1
The model decides that information is way closer to digital than it is to cherry, a
result that seems sensible. Fig. 5.5 shows a visualization.
104 C HAPTER 5 • E MBEDDINGS
Dimension 1: ‘pie’
500
cherry
digital information
Dimension 2: ‘computer’
Figure 5.5 A (rough) graphical demonstration of cosine similarity, showing vectors for
three words (cherry, digital, and information) in the two dimensional space defined by counts
of the words computer and pie nearby. The figure doesn’t show the cosine, but it highlights the
angles; note that the angle between digital and information is smaller than the angle between
cherry and information. When two vectors are more similar, the cosine is larger but the angle
is smaller; the cosine has its maximum (1) when the angle between two vectors is smallest
(0◦ ); the cosine of all other angles is less than 1.
can be used to compute word similarity, for tasks like finding word paraphrases,
tracking changes in word meaning, or automatically discovering meanings of words
in different corpora. For example, we can find the 10 most similar words to any
target word w by computing the cosines between w and each of the |V | − 1 other
words, sorting, and looking at the top 10.
5.5 Word2vec
In the previous sections we saw how to represent a word as a sparse, long vector with
dimensions corresponding to words in the vocabulary. We now introduce a more
powerful word representation: embeddings, short dense vectors. Unlike the vectors
we’ve seen so far, embeddings are short, with number of dimensions d ranging from
50-1000, rather than the much larger vocabulary size |V |.These d dimensions don’t
have a clear interpretation. And the vectors are dense: instead of vector entries
being sparse, mostly-zero counts or functions of counts, the values will be real-
valued numbers that can be negative.
It turns out that dense vectors work better in every NLP task than sparse vectors.
While we don’t completely understand all the reasons for this, we have some intu-
itions. Representing words as 300-dimensional dense vectors requires our classifiers
to learn far fewer weights than if we represented words as 50,000-dimensional vec-
tors, and the smaller parameter space possibly helps with generalization and avoid-
ing overfitting. Dense vectors may also do a better job of capturing synonymy.
For example, in a sparse vector representation, dimensions for synonyms like car
and automobile dimension are distinct and unrelated; sparse vectors may thus fail
to capture the similarity between a word with car as a neighbor and a word with
automobile as a neighbor.
skip-gram In this section we introduce one method for computing embeddings: skip-gram
SGNS with negative sampling, sometimes called SGNS. The skip-gram algorithm is one
word2vec of two algorithms in a software package called word2vec, and so sometimes the
algorithm is loosely referred to as word2vec (Mikolov et al. 2013a, Mikolov et al.
2013b). The word2vec methods are fast, efficient to train, and easily available on-
line with code and pretrained embeddings. Word2vec embeddings are static em-
5.5 • W ORD 2 VEC 105
static
embeddings beddings, meaning that the method learns one fixed embedding for each word in the
vocabulary. In Chapter 10 we’ll introduce methods for learning dynamic contextual
embeddings like the popular family of BERT representations, in which the vector
for each word is different in different contexts.
The intuition of word2vec is that instead of counting how often each word w oc-
curs near, say, apricot, we’ll instead train a classifier on a binary prediction task: “Is
word w likely to show up near apricot?” We don’t actually care about this prediction
task; instead we’ll take the learned classifier weights as the word embeddings.
The revolutionary intuition here is that we can just use running text as implicitly
supervised training data for such a classifier; a word c that occurs near the target
word apricot acts as gold ‘correct answer’ to the question “Is word c likely to show
self-supervision up near apricot?” This method, often called self-supervision, avoids the need for
any sort of hand-labeled supervision signal. This idea was first proposed in the task
of neural language modeling, when Bengio et al. (2003) and Collobert et al. (2011)
showed that a neural language model (a neural network that learned to predict the
next word from prior words) could just use the next word in running text as its
supervision signal, and could be used to learn an embedding representation for each
word as part of doing this prediction task.
We’ll see how to do neural networks in the next chapter, but word2vec is a
much simpler model than the neural network language model, in two ways. First,
word2vec simplifies the task (making it binary classification instead of word pre-
diction). Second, word2vec simplifies the architecture (training a logistic regression
classifier instead of a multi-layer neural network with hidden layers that demand
more sophisticated training algorithms). The intuition of skip-gram is:
1. Treat the target word and a neighboring context word as positive examples.
2. Randomly sample other words in the lexicon to get negative samples.
3. Use logistic regression to train a classifier to distinguish those two cases.
4. Use the learned weights as the embeddings.
P(+|w, c) (5.11)
The probability that word c is not a real context word for w is just 1 minus
Eq. 5.11:
occur near the target if its embedding vector is similar to the target embedding. To
compute similarity between these dense embeddings, we rely on the intuition that
two vectors are similar if they have a high dot product (after all, cosine is just a
normalized dot product). In other words:
Similarity(w, c) ≈ c · w (5.13)
The dot product c · w is not a probability, it’s just a number ranging from −∞ to ∞
(since the elements in word2vec embeddings can be negative, the dot product can be
negative). To turn the dot product into a probability, we’ll use the logistic or sigmoid
function σ (x), the fundamental core of logistic regression:
1
σ (x) = (5.14)
1 + exp (−x)
We model the probability that word c is a real context word for target word w as:
1
P(+|w, c) = σ (c · w) = (5.15)
1 + exp (−c · w)
The sigmoid function returns a number between 0 and 1, but to make it a probability
we’ll also need the total probability of the two possible events (c is a context word,
and c isn’t a context word) to sum to 1. We thus estimate the probability that word c
is not a real context word for w as:
P(−|w, c) = 1 − P(+|w, c)
1
= σ (−c · w) = (5.16)
1 + exp (c · w)
Equation 5.15 gives us the probability for one word, but there are many context
words in the window. Skip-gram makes the simplifying assumption that all context
words are independent, allowing us to just multiply their probabilities:
L
Y
P(+|w, c1:L ) = σ (ci · w) (5.17)
i=1
XL
log P(+|w, c1:L ) = log σ (ci · w) (5.18)
i=1
In summary, skip-gram trains a probabilistic classifier that, given a test target word
w and its context window of L words c1:L , assigns a probability based on how similar
this context window is to the target word. The probability is based on applying the
logistic (sigmoid) function to the dot product of the embeddings of the target word
with each context word. To compute this probability, we just need embeddings for
each target word and context word in the vocabulary.
Fig. 5.6 shows the intuition of the parameters we’ll need. Skip-gram actually
stores two embeddings for each word, one for the word as a target, and one for the
word considered as context. Thus the parameters we need to learn are two matrices
W and C, each containing an embedding for every one of the |V | words in the
vocabulary V .2 Let’s now turn to learning these embeddings (which is the real goal
of training this classifier in the first place).
2 In principle the target matrix and the context matrix could use different vocabularies, but we’ll simplify
by assuming one shared vocabulary V .
5.5 • W ORD 2 VEC 107
1..d
aardvark 1
apricot
𝜽=
… … W target words
zebra |V|
aardvark |V|+1
apricot
Figure 5.6 The embeddings learned by the skipgram model. The algorithm stores two em-
beddings for each word, the target embedding (sometimes called the input embedding) and
the context embedding (sometimes called the output embedding). The parameter θ that the al-
gorithm learns is thus a matrix of 2|V | vectors, each of dimension d, formed by concatenating
two matrices, the target embeddings W and the context+noise embeddings C.
choose aardvark, and so on. But in practice it is common to set α = 0.75, i.e. use
the weighting P3 (w):
4
count(w)α
Pα (w) = P (5.19)
w0 count(w )
0 α
Setting α = .75 gives better performance because it gives rare noise words slightly
higher probability: for rare words, Pα (w) > P(w). To illustrate this intuition, it
might help to work out the probabilities for an example with α = .75 and two events,
P(a) = 0.99 and P(b) = 0.01:
.99.75
Pα (a) = = 0.97
.99.75 + .01.75
.01.75
Pα (b) = = 0.03 (5.20)
.99.75 + .01.75
Thus using α = .75 increases the probability of the rare event b from 0.01 to 0.03.
Given the set of positive and negative training instances, and an initial set of
embeddings, the goal of the learning algorithm is to adjust those embeddings to
• Maximize the similarity of the target word, context word pairs (w, cpos ) drawn
from the positive examples
• Minimize the similarity of the (w, cneg ) pairs from the negative examples.
If we consider one word/context pair (w, cpos ) with its k noise words cneg1 ...cnegk ,
we can express these two goals as the following loss function L to be minimized
(hence the −); here the first term expresses that we want the classifier to assign the
real context word cpos a high probability of being a neighbor, and the second term
expresses that we want to assign each of the noise words cnegi a high probability of
being a non-neighbor, all multiplied because we assume independence:
" k
#
Y
L = − log P(+|w, cpos ) P(−|w, cnegi )
i=1
" k
#
X
= − log P(+|w, cpos ) + log P(−|w, cnegi )
i=1
" k
#
X
= − log P(+|w, cpos ) + log 1 − P(+|w, cnegi )
i=1
" k
#
X
= − log σ (cpos · w) + log σ (−cnegi · w) (5.21)
i=1
That is, we want to maximize the dot product of the word with the actual context
words, and minimize the dot products of the word with the k negative sampled non-
neighbor words.
We minimize this loss function using stochastic gradient descent. Fig. 5.7 shows
the intuition of one step of learning.
To get the gradient, we need to take the derivative of Eq. 5.21 with respect to
the different embeddings. It turns out the derivatives are the following (we leave the
5.5 • W ORD 2 VEC 109
aardvark
move apricot and jam closer,
apricot w increasing cpos w
W
“…apricot jam…”
zebra
! aardvark move apricot and matrix apart
cpos decreasing cneg1 w
jam
C matrix cneg1
k=2
Tolstoy cneg2 move apricot and Tolstoy apart
decreasing cneg2 w
zebra
Figure 5.7 Intuition of one step of gradient descent. The skip-gram model tries to shift em-
beddings so the target embeddings (here for apricot) are closer to (have a higher dot product
with) context embeddings for nearby words (here jam) and further from (lower dot product
with) context embeddings for noise words that don’t occur nearby (here Tolstoy and matrix).
∂L
= [σ (cpos · w) − 1]w (5.22)
∂ cpos
∂L
= [σ (cneg · w)]w (5.23)
∂ cneg
X k
∂L
= [σ (cpos · w) − 1]cpos + [σ (cnegi · w)]cnegi (5.24)
∂w
i=1
The update equations going from time step t to t + 1 in stochastic gradient descent
are thus:
ct+1 t t t
pos = cpos − η[σ (cpos · w ) − 1]w
t
(5.25)
ct+1
neg = ctneg − η[σ (ctneg · wt )]wt (5.26)
" k
#
X
wt+1 = wt − η [σ (ctpos · wt ) − 1]ctpos + [σ (ctnegi · wt )]ctnegi (5.27)
i=1
Just as in logistic regression, then, the learning algorithm starts with randomly ini-
tialized W and C matrices, and then walks through the training corpus using gradient
descent to move W and C so as to minimize the loss in Eq. 5.21 by making the up-
dates in (Eq. 5.25)-(Eq. 5.27).
Recall that the skip-gram model learns two separate embeddings for each word i:
target
embedding the target embedding wi and the context embedding ci , stored in two matrices, the
context
embedding target matrix W and the context matrix C. It’s common to just add them together,
representing word i with the vector wi + ci . Alternatively we can throw away the C
matrix and just represent each word i by the vector wi .
As with the simple count-based methods like tf-idf, the context window size L
affects the performance of skip-gram embeddings, and experiments often tune the
parameter L on a devset.
110 C HAPTER 5 • E MBEDDINGS
2014), short for Global Vectors, because the model is based on capturing global
FRANCE
CHINA
WRIST
corpus statistics. GloVe is based on ratios of probabilities from the word-word co-
EUROPE
ASIA
occurrence matrix.
ANKLE AFRICA
AMERICA
ARM
BRAZIL
SHOULDER
It turns out that dense embeddings like word2vec actually have an elegant math-
FINGER
EYE
FACE
EARHAND MOSCOW
TOE LEG
FOOT ematical relationship with count-based embeddings, in which word2vec can be seen
as implicitly optimizing a function of a count matrix with a particular (PPMI) weight-
HAWAII
TOOTH
NOSE
HEAD TOKYO
5.6
DOG
CAT
Visualizing Embeddings
TURTLE
LION NASHVILLE
PUPPY
KITTEN COW
OYSTER
“I see well in many dimensions as long as the dimensions are around two.”
BULL The late economist Martin Shubik
Figure 8: Multidimensional scaling for three noun classes.
Visualizing embeddings is an important goal in helping understand, apply, and
improve these models of word meaning. But how can we visualize a (for example)
100-dimensional vector?
WRIST The simplest way to visualize the meaning of a word
ANKLE
SHOULDER
ARM
w embedded in a space is to list the most similar words to
LEG
HAND w by sorting the vectors for all words in the vocabulary by
FOOT
HEAD
NOSE
their cosine with the vector for w. For example the 7 closest
FINGER
TOE words to frog using a particular embeddings computed with
FACE
EAR
EYE
the GloVe algorithm are: frogs, toad, litoria, leptodactyli-
DOG
TOOTH
dae, rana, lizard, and eleutherodactylus (Pennington et al.,
CAT
PUPPY
KITTEN
2014).
MOUSE
COW
Yet another visualization method is to use a clustering
TURTLE
LION
OYSTER algorithm to show a hierarchical representation of which
BULL
CHICAGO words are similar to others in the embedding space. The
uncaptioned figure on the left uses hierarchical clustering
ATLANTA
MONTREAL
NASHVILLE
CHINA
TOKYO
of some embedding vectors for nouns as a visualization
method (Rohde et al., 2006).
RUSSIA
AFRICA
ASIA
EUROPE
AMERICA
BRAZIL
MOSCOW
FRANCE
HAWAII
Figure 9: Hierarchical clustering for three noun classes using distances based on vector correlations.
5.7 • S EMANTIC PROPERTIES OF EMBEDDINGS 111
tree
apple
vine
grape
Figure 5.8 The parallelogram model for analogy problems (Rumelhart and Abrahamson,
# » # » # » # »
1973): the location of vine can be found by subtracting apple from tree and adding grape.
ing representations of relations like MALE - FEMALE, or CAPITAL - CITY- OF, or even
COMPARATIVE / SUPERLATIVE , as shown in Fig. 5.9 from GloVe.
(a) (b)
Figure 5.9 Relational properties of the GloVe vector space, shown by projecting vectors onto two dimensions.
# » # » # » is close to queen.
# » (b) offsets seem to capture comparative and superlative morphology
(a) king − man + woman
(Pennington et al., 2014).
Figure 5.10 A t-SNE visualization of the semantic change of 3 words in English using
Figure
word2vec5.1: Two-dimensional
vectors. The modern sensevisualization of semantic
of each word, and the change in English
grey context words,using SGNS
are com-
vectors
puted from(seetheSection 5.8 for
most recent the visualization
(modern) algorithm).
time-point embedding A,Earlier
space. The word
pointsgay shifted
are com-
from
puted meaning “cheerful”
from earlier historicalorembedding
“frolicsome” to referring
spaces. to homosexuality.
The visualizations A, In the
show the changes early
in the
20th century broadcast referred to “casting out seeds”; with the rise of television
word gay from meanings related to “cheerful” or “frolicsome” to referring to homosexuality, and
radio
the development of the modern “transmission” sense of broadcast from its original sense of of
its meaning shifted to “transmitting signals”. C, Awful underwent a process
pejoration,
sowing seeds,asandit shifted from meaning
the pejoration “full
of the word of awe”
awful to meaning
as it shifted “terrible“full
from meaning or appalling”
of awe”
to meaning “terrible or appalling” (Hamilton et al., 2016b).
[212].
5.8 Biaswhere
andtheyEmbeddings
shift from objective statements about the world (e.g., “Sorry, the car is
actually broken”) to subjective statements (e.g., “I can’t believe he actually did that”,
indicating surprise/disbelief).
In addition to their ability to learn word meaning from text, embeddings, alas,
also reproduce the implicit biases and stereotypes that were latent in the text. As
5.2.2
the prior Computational
section just showed, linguistic
embeddings can studies
roughly model relational similar-
ity: ‘queen’ as the closest word to ‘king’ - ‘man’ + ‘woman’ implies the analogy
There are also a number of recent works analyzing semantic change using computational
man:woman::king:queen. But these same embedding analogies also exhibit gender
stereotypes.
methods. [200] Foruse
example Bolukbasianalysis
latent semantic et al. (2016) find that
to analyze how the closest
word occupation
meanings broaden
to ‘computer
and narrow over programmer’
time. [113] - ‘man’ + ‘woman’
use raw in word2vec
co-occurrence vectorsembeddings
to perform trained
a numberon of
news text is ‘homemaker’, and that the embeddings similarly suggest the
historical case-studies on semantic change, and [252] perform a similar set of small- analogy
‘father’ is to ‘doctor’ as ‘mother’ is to ‘nurse’. This could result in what Crawford
allocational scale case-studies using temporal topic models. [87] construct point-wise mutual
harm (2017) and Blodgett et al. (2020) call an allocational harm, when a system allo-
cates resources (jobs
information-based or credit) unfairly
embeddings to different
and found [Link]
that semantic For example algorithms
uncovered by their
that use had
method embeddings
reasonable as agreement
part of a search for hiring
with human potential [129]
judgments. programmers
and [119]oruse
doctors
“neural”
might thus incorrectly downweight documents with women’s names.
word-embedding methods to detect linguistic change points. Finally, [257] analyze
It turns out that embeddings don’t just reflect the statistics of their input, but also
bias historical co-occurrences to test whether synonyms tend to change in similar ways.
amplification amplify bias; gendered terms become more gendered in embedding space than they
were in the input text statistics (Zhao et al. 2017, Ethayarajh et al. 2019b, Jia et al.
2020), and biases are more exaggerated than in actual labor employment statistics
(Garg et al., 2018).
Embeddings also encode the implicit associations that are a property of human
reasoning. The Implicit Association Test (Greenwald et al., 1998) measures peo-
114 C HAPTER 5 • E MBEDDINGS
ple’s associations between concepts (like ‘flowers’ or ‘insects’) and attributes (like
‘pleasantness’ and ‘unpleasantness’) by measuring differences in the latency with
which they label words in the various categories.3 Using such methods, people
in the United States have been shown to associate African-American names with
unpleasant words (more than European-American names), male names more with
mathematics and female names with the arts, and old people’s names with unpleas-
ant words (Greenwald et al. 1998, Nosek et al. 2002a, Nosek et al. 2002b). Caliskan
et al. (2017) replicated all these findings of implicit associations using GloVe vectors
and cosine similarity instead of human latencies. For example African-American
names like ‘Leroy’ and ‘Shaniqua’ had a higher GloVe cosine with unpleasant words
while European-American names (‘Brad’, ‘Greg’, ‘Courtney’) had a higher cosine
with pleasant words. These problems with embeddings are an example of a repre-
representational
harm
sentational harm (Crawford 2017, Blodgett et al. 2020), which is a harm caused by
a system demeaning or even ignoring some social groups. Any embedding-aware al-
gorithm that made use of word sentiment could thus exacerbate bias against African
Americans.
Recent research focuses on ways to try to remove these kinds of biases, for
example by developing a transformation of the embedding space that removes gen-
der stereotypes but preserves definitional gender (Bolukbasi et al. 2016, Zhao et al.
2017) or changing the training procedure (Zhao et al., 2018b). However, although
debiasing these sorts of debiasing may reduce bias in embeddings, they do not eliminate it
(Gonen and Goldberg, 2019), and this remains an open problem.
Historical embeddings are also being used to measure biases in the past. Garg
et al. (2018) used embeddings from historical texts to measure the association be-
tween embeddings for occupations and embeddings for names of various ethnici-
ties or genders (for example the relative cosine similarity of women’s names versus
men’s to occupation words like ‘librarian’ or ‘carpenter’) across the 20th century.
They found that the cosines correlate with the empirical historical percentages of
women or ethnic groups in those occupations. Historical embeddings also repli-
cated old surveys of ethnic stereotypes; the tendency of experimental participants in
1933 to associate adjectives like ‘industrious’ or ‘superstitious’ with, e.g., Chinese
ethnicity, correlates with the cosine between Chinese last names and those adjectives
using embeddings trained on 1930s text. They also were able to document historical
gender biases, such as the fact that embeddings for adjectives related to competence
(‘smart’, ‘wise’, ‘thoughtful’, ‘resourceful’) had a higher cosine with male than fe-
male words, and showed that this bias has been slowly decreasing since 1960. We
return in later chapters to this question about the role of bias in natural language
processing.
5.10 Summary
• In vector semantics, a word is modeled as a vector—a point in high-dimensional
space, also called an embedding. In this chapter we focus on static embed-
dings, where each word is mapped to a fixed embedding.
• Vector semantic models fall into two classes: sparse and dense. In sparse
models each dimension corresponds to a word in the vocabulary V and cells
are functions of co-occurrence counts. The word-context or term-term ma-
trix has a row for each (target) word in the vocabulary and a column for each
context term in the vocabulary.
116 C HAPTER 5 • E MBEDDINGS
Historical Notes
The idea of vector semantics arose out of research in the 1950s in three distinct
fields: linguistics, psychology, and computer science, each of which contributed a
fundamental aspect of the model.
The idea that meaning is related to the distribution of words in context was
widespread in linguistic theory of the 1950s, among distributionalists like Zellig
Harris, Martin Joos, and J. R. Firth, and semioticians like Thomas Sebeok. As Joos
(1950) put it,
the linguist’s “meaning” of a morpheme. . . is by definition the set of conditional
probabilities of its occurrence in context with all other morphemes.
The idea that the meaning of a word might be modeled as a point in a multi-
dimensional semantic space came from psychologists like Charles E. Osgood, who
had been studying how people responded to the meaning of words by assigning val-
ues along scales like happy/sad or hard/soft. Osgood et al. (1957) proposed that the
meaning of a word in general could be modeled as a point in a multidimensional
Euclidean space, and that the similarity of meaning between two words could be
modeled as the distance between these points in the space.
A final intellectual source in the 1950s and early 1960s was the field then called
mechanical
indexing mechanical indexing, now known as information retrieval. In what became known
as the vector space model for information retrieval (Salton 1971, Sparck Jones
1986), researchers demonstrated new ways to define the meaning of words in terms
of vectors (Switzer, 1965), and refined methods for word similarity based on mea-
sures of statistical association between words like mutual information (Giuliano,
1965) and idf (Sparck Jones, 1972), and showed that the meaning of documents
could be represented in the same vector spaces used for words. Around the same
time, (Cordier, 1965) showed that factor analysis of word association probabilities
could be used to form dense vector representations of words.
Some of the philosophical underpinning of the distributional way of thinking
came from the late writings of the philosopher Wittgenstein, who was skeptical of
the possibility of building a completely formal theory of meaning definitions for
each word. Wittgenstein suggested instead that “the meaning of a word is its use in
the language” (Wittgenstein, 1953, PI 43). That is, instead of using some logical lan-
guage to define each word, or drawing on denotations or truth values, Wittgenstein’s
H ISTORICAL N OTES 117
idea is that we should define a word by how it is used by people in speaking and un-
derstanding in their day-to-day interactions, thus prefiguring the movement toward
embodied and experiential models in linguistics and NLP (Glenberg and Robertson
2000, Lake and Murphy 2021, Bisk et al. 2020, Bender and Koller 2020).
More distantly related is the idea of defining words by a vector of discrete fea-
tures, which has roots at least as far back as Descartes and Leibniz (Wierzbicka 1992,
Wierzbicka 1996). By the middle of the 20th century, beginning with the work of
Hjelmslev (Hjelmslev, 1969) (originally 1943) and fleshed out in early models of
generative grammar (Katz and Fodor, 1963), the idea arose of representing mean-
semantic ing with semantic features, symbols that represent some sort of primitive meaning.
feature
For example words like hen, rooster, or chick, have something in common (they all
describe chickens) and something different (their age and sex), representable as:
hen +female, +chicken, +adult
rooster -female, +chicken, +adult
chick +chicken, -adult
The dimensions used by vector models of meaning to define words, however, are
only abstractly related to this idea of a small fixed number of hand-built dimensions.
Nonetheless, there has been some attempt to show that certain dimensions of em-
bedding models do contribute some specific compositional aspect of meaning like
these early semantic features.
The use of dense vectors to model word meaning, and indeed the term embed-
ding, grew out of the latent semantic indexing (LSI) model (Deerwester et al.,
1988) recast as LSA (latent semantic analysis) (Deerwester et al., 1990). In LSA
SVD singular value decomposition—SVD— is applied to a term-document matrix (each
cell weighted by log frequency and normalized by entropy), and then the first 300
dimensions are used as the LSA embedding. Singular Value Decomposition (SVD)
is a method for finding the most important dimensions of a data set, those dimen-
sions along which the data varies the most. LSA was then quickly widely applied:
as a cognitive model (Landauer and Dumais, 1997), and for tasks like spell checking
(Jones and Martin, 1997), language modeling (Bellegarda 1997, Coccaro and Ju-
rafsky 1998, Bellegarda 2000), morphology induction (Schone and Jurafsky 2000,
Schone and Jurafsky 2001b), multiword expressions (MWEs) (Schone and Juraf-
sky, 2001a), and essay grading (Rehder et al., 1998). Related models were simul-
taneously developed and applied to word sense disambiguation by Schütze (1992b).
LSA also led to the earliest use of embeddings to represent words in a probabilis-
tic classifier, in the logistic regression document router of Schütze et al. (1995).
The idea of SVD on the term-term matrix (rather than the term-document matrix)
as a model of meaning for NLP was proposed soon after LSA by Schütze (1992b).
Schütze applied the low-rank (97-dimensional) embeddings produced by SVD to the
task of word sense disambiguation, analyzed the resulting semantic space, and also
suggested possible techniques like dropping high-order dimensions. See Schütze
(1997).
A number of alternative matrix models followed on from the early SVD work,
including Probabilistic Latent Semantic Indexing (PLSI) (Hofmann, 1999), Latent
Dirichlet Allocation (LDA) (Blei et al., 2003), and Non-negative Matrix Factoriza-
tion (NMF) (Lee and Seung, 1999).
The LSA community seems to have first used the word “embedding” in Landauer
et al. (1997), in a variant of its mathematical meaning as a mapping from one space
or mathematical structure to another. In LSA, the word embedding seems to have
described the mapping from the space of sparse count vectors to the latent space of
118 C HAPTER 5 • E MBEDDINGS
SVD dense vectors. Although the word thus originally meant the mapping from one
space to another, it has metonymically shifted to mean the resulting dense vector in
the latent space, and it is in this sense that we currently use the word.
By the next decade, Bengio et al. (2003) and Bengio et al. (2006) showed that
neural language models could also be used to develop embeddings as part of the task
of word prediction. Collobert and Weston (2007), Collobert and Weston (2008), and
Collobert et al. (2011) then demonstrated that embeddings could be used to represent
word meanings for a number of NLP tasks. Turian et al. (2010) compared the value
of different kinds of embeddings for different NLP tasks. Mikolov et al. (2011)
showed that recurrent neural nets could be used as language models. The idea of
simplifying the hidden layer of these neural net language models to create the skip-
gram (and also CBOW) algorithms was proposed by Mikolov et al. (2013a). The
negative sampling training algorithm was proposed in Mikolov et al. (2013b). There
are numerous surveys of static embeddings and their parameterizations (Bullinaria
and Levy 2007, Bullinaria and Levy 2012, Lapesa and Evert 2014, Kiela and Clark
2014, Levy et al. 2015).
See Manning et al. (2008) and Chapter 11 for a deeper understanding of the role
of vectors in information retrieval, including how to compare queries with docu-
ments, more details on tf-idf, and issues of scaling to very large datasets. See Kim
(2019) for a clear and comprehensive tutorial on word2vec. Cruse (2004) is a useful
introductory linguistic text on lexical semantics.
Exercises
CHAPTER
6 Neural Networks
6.1 Units
The building block of a neural network is a single computational unit. A unit takes
a set of real valued numbers as input, performs some computation on them, and
produces an output.
At its heart, a neural unit is taking a weighted sum of its inputs, with one addi-
bias term tional term in the sum called a bias term. Given a set of inputs x1 ...xn , a unit has
a set of corresponding weights w1 ...wn and a bias b, so the weighted sum z can be
represented as: X
z = b+ wi xi (6.1)
i
Often it’s more convenient to express this weighted sum using vector notation; recall
vector from linear algebra that a vector is, at heart, just a list or array of numbers. Thus
we’ll talk about z in terms of a weight vector w, a scalar bias b, and an input vector
x, and we’ll replace the sum with the convenient dot product:
z = w·x+b (6.2)
As defined in Eq. 6.2, z is just a real valued number.
Finally, instead of using z, a linear function of x, as the output, neural units
apply a non-linear function f to z. We will refer to the output of this function as
activation the activation value for the unit, a. Since we are just modeling a single unit, the
activation for the node is in fact the final output of the network, which we’ll generally
call y. So the value y is defined as:
y = a = f (z)
We’ll discuss three popular non-linear functions f below (the sigmoid, the tanh, and
the rectified linear unit or ReLU) but it’s pedagogically convenient to start with the
sigmoid sigmoid function since we saw it in Chapter 4:
1
y = σ (z) = (6.3)
1 + e−z
The sigmoid (shown in Fig. 6.1) has a number of advantages; it maps the output
into the range (0, 1), which is useful in squashing outliers toward 0 or 1. And it’s
differentiable, which as we saw in Section 4.15 will be handy for learning.
Figure 6.1 The sigmoid function takes a real value and maps it to the range (0, 1). It is
nearly linear around 0 but outlier values get squashed toward 0 or 1.
Substituting Eq. 6.2 into Eq. 6.3 gives us the output of a neural unit:
1
y = σ (w · x + b) = (6.4)
1 + exp(−(w · x + b))
6.1 • U NITS 121
Fig. 6.2 shows a final schematic of a basic neural unit. In this example the unit
takes 3 input values x1 , x2 , and x3 , and computes a weighted sum, multiplying each
value by a weight (w1 , w2 , and w3 , respectively), adds them to a bias term b, and then
passes the resulting sum through a sigmoid function to result in a number between 0
and 1.
x1 w1
w2 z a
x2 ∑ σ y
w3
x3 b
+1
Figure 6.2 A neural unit, taking 3 inputs x1 , x2 , and x3 (and a bias b that we represent as a
weight for an input clamped at +1) and producing an output y. We include some convenient
intermediate variables: the output of the summation, z, and the output of the sigmoid, a. In
this case the output of the unit y is the same as a, but in deeper networks we’ll reserve y to
mean the final output of the entire network, leaving a as the activation of an individual node.
Let’s walk through an example just to get an intuition. Let’s suppose we have a
unit with the following weight vector and bias:
ez − e−z
y = tanh(z) = (6.5)
ez + e−z
The simplest activation function, and perhaps the most commonly used, is the rec-
ReLU tified linear unit, also called the ReLU, shown in Fig. 6.3b. It’s just the same as z
when z is positive, and 0 otherwise:
These activation functions have different properties that make them useful for differ-
ent language applications or network architectures. For example, the tanh function
has the nice properties of being smoothly differentiable and mapping outlier values
toward the mean. The rectifier function, on the other hand, has nice properties that
122 C HAPTER 6 • N EURAL N ETWORKS
(a) (b)
Figure 6.3 The tanh and ReLU activation functions.
result from it being very close to linear. In the sigmoid or tanh functions, very high
saturated values of z result in values of y that are saturated, i.e., extremely close to 1, and have
derivatives very close to 0. Zero derivatives cause problems for learning, because as
we’ll see in Section 6.6, we’ll train networks by propagating an error signal back-
wards, multiplying gradients (partial derivatives) from each layer of the network;
gradients that are almost 0 cause the error signal to get smaller and smaller until it is
vanishing
gradient too small to be used for training, a problem called the vanishing gradient problem.
Rectifiers don’t have this problem, since the derivative of ReLU for high values of z
is 1 rather than very close to 0.
AND OR XOR
x1 x2 y x1 x2 y x1 x2 y
0 0 0 0 0 0 0 0 0
0 1 0 0 1 1 0 1 1
1 0 0 1 0 1 1 0 1
1 1 1 1 1 1 1 1 0
perceptron This example was first shown for the perceptron, which is a very simple neural
unit that has a binary output and has a very simple step function as its non-linear
activation function. The output y of a perceptron is 0 or 1, and is computed as
follows (using the same weight w, input x, and bias b as in Eq. 6.2):
0, if w · x + b ≤ 0
y= (6.7)
1, if w · x + b > 0
6.2 • T HE XOR PROBLEM 123
It’s very easy to build a perceptron that can compute the logical AND and OR
functions of its binary inputs; Fig. 6.4 shows the necessary weights.
x1 x1
1 1
x2 1 x2 1
-1 0
+1 +1
(a) (b)
Figure 6.4 The weights w and bias b for perceptrons for computing logical functions. The
inputs are shown as x1 and x2 and the bias as a special node with value +1 which is multiplied
with the bias weight b. (a) logical AND, with weights w1 = 1 and w2 = 1 and bias weight
b = −1. (b) logical OR, with weights w1 = 1 and w2 = 1 and bias weight b = 0. These
weights/biases are just one from an infinite number of possible sets of weights and biases that
would implement the functions.
It turns out, however, that it’s not possible to build a perceptron to compute
logical XOR! (It’s worth spending a moment to give it a try!)
The intuition behind this important result relies on understanding that a percep-
tron is a linear classifier. For a two-dimensional input x1 and x2 , the perceptron
equation, w1 x1 + w2 x2 + b = 0 is the equation of a line. (We can see this by putting
it in the standard linear format: x2 = (−w1 /w2 )x1 + (−b/w2 ).) This line acts as a
decision
boundary decision boundary in two-dimensional space in which the output 0 is assigned to all
inputs lying on one side of the line, and the output 1 to all input points lying on the
other side of the line. If we had more than 2 inputs, the decision boundary becomes
a hyperplane instead of a line, but the idea is the same, separating the space into two
categories.
Fig. 6.5 shows the possible logical inputs (00, 01, 10, and 11) and the line drawn
by one possible set of parameters for an AND and an OR classifier. Notice that there
is simply no way to draw a line that separates the positive cases of XOR (01 and 10)
linearly
separable from the negative cases (00 and 11). We say that XOR is not a linearly separable
function. Of course we could draw a boundary with a curve, or some other function,
but not a single line.
x2 x2 x2
1 1 1
?
0 x1 0 x1 0 x1
0 1 0 1 0 1
a) x1 AND x2 b) x1 OR x2 c) x1 XOR x2
Figure 6.5 The functions AND, OR, and XOR, represented with input x1 on the x-axis and input x2 on the
y-axis. Filled circles represent perceptron outputs of 1, and white circles perceptron outputs of 0. There is no
way to draw a line that correctly separates the two categories for XOR. Figure styled after Russell and Norvig
(2002).
x1 1 h1
1
1
y1
1
-2
x2 1 h2 0
0
-1
+1 +1
Figure 6.6 XOR solution after Goodfellow et al. (2016). There are three ReLU units, in
two layers; we’ve called them h1 , h2 (h for “hidden layer”) and y1 . As before, the numbers
on the arrows represent the weights w for each unit, and we represent the bias b as a weight
on a unit clamped to +1, with the bias weights/units in gray.
It’s also instructive to look at the intermediate results, the outputs of the two
hidden nodes h1 and h2 . We showed in the previous paragraph that the h vector for
the inputs x = [0, 0] was [0, 0]. Fig. 6.7b shows the values of the h layer for all
4 inputs. Notice that hidden representations of the two input points x = [0, 1] and
x = [1, 0] (the two cases with XOR output = 1) are merged to the single point h =
[1, 0]. The merger makes it easy to linearly separate the positive and negative cases
of XOR. In other words, we can view the hidden layer of the network as forming a
representation of the input.
In this example we just stipulated the weights in Fig. 6.6. But for real examples
the weights for neural networks are learned automatically using the error backprop-
agation algorithm to be introduced in Section 6.6. That means the hidden layers will
learn to form useful representations. This intuition, that neural networks can auto-
matically learn useful representations of the input, is one of their key advantages,
and one that we will return to again and again in later chapters.
6.3 • F EEDFORWARD N EURAL N ETWORKS 125
x2 h2
1 1
0 x1 0
h1
0 1 0 1 2
x1 W U
y1
h1
x2 h2 y2
h3
…
…
xn
0 hn
1
b yn
2
+1
input layer hidden layer output layer
Figure 6.8 A simple 2-layer feedforward network, with one hidden layer, one output layer,
and one input layer (the input layer is usually not counted when enumerating layers).
efficiently with simple matrix operations. In fact, the computation only has three
steps: multiplying the weight matrix by the input vector x, adding the bias vector b,
and applying the activation function g (such as the sigmoid, tanh, or ReLU activation
function defined above).
The output of the hidden layer, the vector h, is thus the following (for this exam-
ple we’ll use the sigmoid function σ as our activation function):
h = σ (Wx + b) (6.8)
Notice that we’re applying the σ function here to a vector, while in Eq. 6.3 it was
applied to a scalar. We’re thus allowing σ (·), and indeed any activation function
g(·), to apply to a vector element-wise, so g[z1 , z2 , z3 ] = [g(z1 ), g(z2 ), g(z3 )].
Let’s introduce some constants to represent the dimensionalities of these vectors
and matrices. We’ll refer to the input layer as layer 0 of the network, and have
n0 represent the number of inputs, so x is a vector of real numbers of dimension
n0 , or more formally x ∈ Rn0 , a column vector of dimensionality [n0 × 1]. Let’s
call the hidden layer layer 1 and the output layer layer 2. The hidden layer has
dimensionality n1 , so h ∈ Rn1 and also b ∈ Rn1 (since each hidden unit can take a
different bias value). And the weight matrix W has dimensionality W ∈ Rn1 ×n0 , i.e.
[n1 × n0 ].
Take a moment to convince yourself
Pn0 that the matrix multiplication in Eq. 6.8 will
compute the value of each h j as σ i=1 W ji xi + b j .
As we saw in Section 6.2, the resulting value h (for hidden but also for hypoth-
esis) forms a representation of the input. The role of the output layer is to take
this new representation h and compute a final output. This output could be a real-
valued number, but in many cases the goal of the network is to make some sort of
classification decision, and so we will focus on the case of classification.
If we are doing a binary task like sentiment classification, we might have a sin-
gle output node, and its scalar value y is the probability of positive versus negative
sentiment. If we are doing multinomial classification, such as assigning a part-of-
speech tag, we might have one output node for each potential part-of-speech, whose
output value is the probability of that part-of-speech, and the values of all the output
nodes must sum to one. The output layer is thus a vector y that gives a probability
distribution across the output nodes.
6.3 • F EEDFORWARD N EURAL N ETWORKS 127
Let’s see how this happens. Like the hidden layer, the output layer has a weight
matrix (let’s call it U), but some models don’t include a bias vector b in the output
layer, so we’ll simplify by eliminating the bias vector in this example. The weight
matrix is multiplied by its input vector (h) to produce the intermediate output z:
z = Uh
There are n2 output nodes, so z ∈ Rn2 , weight matrix U has dimensionality U ∈
Rn2 ×n1 , and element Ui j is the weight from unit j in the hidden layer to unit i in the
output layer.
However, z can’t be the output of the classifier, since it’s a vector of real-valued
numbers, while what we need for classification is a vector of probabilities. There is
normalizing a convenient function for normalizing a vector of real values, by which we mean
converting it to a vector that encodes a probability distribution (all the numbers lie
softmax between 0 and 1 and sum to 1): the softmax function that we saw on page 70 of
Chapter 4. More generally for any vector z of dimensionality d, the softmax is
defined as:
exp(zi )
softmax(zi ) = Pd 1≤i≤d (6.9)
j=1 exp(z j )
Thus for example given a vector
z = [0.6, 1.1, −1.5, 1.2, 3.2, −1.1], (6.10)
the softmax function will normalize it to a probability distribution (shown rounded):
softmax(z) = [0.055, 0.090, 0.0067, 0.10, 0.74, 0.010] (6.11)
You may recall that we used softmax to create a probability distribution from a
vector of real-valued numbers (computed from summing weights times features) in
the multinomial version of logistic regression in Chapter 4.
That means we can think of a neural network classifier with one hidden layer
as building a vector h which is a hidden layer representation of the input, and then
running standard multinomial logistic regression on the features that the network
develops in h. By contrast, in Chapter 4 the features were mainly designed by hand
via feature templates. So a neural network is like multinomial logistic regression,
but (a) with many layers, since a deep neural network is like layer after layer of lo-
gistic regression classifiers; (b) with those intermediate layers having many possible
activation functions (tanh, ReLU, sigmoid) instead of just sigmoid (although we’ll
continue to use σ for convenience to mean any activation function); (c) rather than
forming the features by feature templates, the prior layers of the network induce the
feature representations themselves.
Here are the final equations for a feedforward network with a single hidden layer,
which takes an input vector x, outputs a probability distribution y, and is parameter-
ized by weight matrices W and U and a bias vector b:
h = σ (Wx + b)
z = Uh
y = softmax(z) (6.12)
And just to remember the shapes of all our variables, x ∈ Rn0 , h ∈ Rn1 , b ∈ Rn1 ,
W ∈ Rn1 ×n0 , U ∈ Rn2 ×n1 , and the output vector y ∈ Rn2 . We’ll call this network a 2-
layer network (we traditionally don’t count the input layer when numbering layers,
but do count the output layer). So by this terminology logistic regression is a 1-layer
network.
128 C HAPTER 6 • N EURAL N ETWORKS
Note that with this notation, the equations for the computation done at each layer are
the same. The algorithm for computing the forward step in an n-layer feedforward
network, given the input vector a[0] is thus simply:
for i in 1,...,n
z[i] = W[i] a[i−1] + b[i]
a[i] = g[i] (z[i] )
ŷ = a[n]
It’s often useful to have a name for the final set of activations right before the final
softmax. So however many layers we have, we’ll generally call the unnormalized
values in the final vector z[n] , the vector of scores right before the final softmax, the
logits logits (see Eq. 4.7).
The need for non-linear activation functions One of the reasons we use non-
linear activation functions for each layer in a neural network is that if we did not, the
resulting network is exactly equivalent to a single-layer network. Let’s see why this
is true. Imagine the first two layers of such a network of purely linear layers:
Replacing the bias unit In describing networks, we will sometimes use a slightly
simplified notation that represents exactly the same function without referring to an
explicit bias node b. Instead, we add a dummy node a0 to each layer whose value
[0]
will always be 1. Thus layer 0, the input layer, will have a dummy node a0 = 1,
[1]
layer 1 will have a0 = 1, and so on. This dummy node still has an associated weight,
and that weight represents the bias value b. For example instead of an equation like
h = σ (Wx + b) (6.15)
we’ll use:
h = σ (Wx) (6.16)
But now instead of our vector x having n0 values: x = x1 , . . . , xn0 , it will have n0 +
1 values, with a new 0th dummy value x0 = 1: x = x0 , . . . , xn0 . And instead of
computing each h j as follows:
n0
!
X
hj = σ Wji xi + b j , (6.17)
i=1
where the value Wj0 replaces what had been b j . Fig. 6.9 shows a visualization.
W U W U
x1 h1 y1 x0=1
h1 y1
h2
x2 y2 x1 h2 y2
h3
…
x2 h3
…
…
…
…
xn
hn
…
0
1
yn hn
b 2 xn 1 yn
+1 0 2
(a) (b)
Figure 6.9 Replacing the bias node (shown in a) with x0 (b).
We’ll continue showing the bias as b when we go over the learning algorithm
in Section 6.6, but going forward in the book, for most figures and some equations
we’ll use this simplified notation without explicit bias terms.
of Chapter 10. Nonetheless seeing a feedforward network text classifier will let us
introduce key ideas that will play a role throughout the rest of the book, includ-
ing the ideas of the embedding matrix, representation pooling, and representation
learning.
But before introducing any of these ideas, let’s start with a classifier by making
only minimal change from the sentiment classifiers we saw in Chapter 4. Like them,
we’ll take hand-built features, pass them through a classifier, and produce a class
probability. The only difference is that we’ll use a neural network instead of logistic
regression as the classifier.
Fig. 6.10 shows a sketch of this architecture. As we mentioned earlier, adding this
hidden layer to our logistic regression classifier allows the network to represent the
non-linear interactions between features. This alone might give us a better sentiment
classifier.
h1
dessert wordcount x1
=3
h2
y^1 p(+)
positive lexicon x2
was words = 1 y^ 2 p(-)
h3
y^3
…
Figure 6.10 Feedforward network sentiment analysis using traditional hand-built features
of the input text.
6.5 • E MBEDDINGS AS THE INPUT TO NEURAL NET CLASSIFIERS 131
H = σ (XW| + b)
Z = HU|
Ŷ = softmax(Z) (6.20)
in which we have already learned embeddings for all the input tokens. An embed-
ding is a vector of dimension d that represents the input token. The dictionary of
embedding static embeddings in which we store these embeddings is the embedding matrix
matrix
E. Each row of the embedding matrix represents each token of the vocabulary V
as a (row) vector of dimensionality d. Since E has a row for each of the |V | to-
kens in the vocabulary, E has shape [|V | × d]. This embedding matrix E plays a role
whenever we are using embeddings as input to neural NLP systems, including in the
transformer-based large language models we will introduce over the next chapters.
Given an input token string like dessert was great we first convert the tokens
into vocabulary indices (these were created when we first tokenized the input using
BPE or SentencePiece). So the representation of dessert was great might be
w = [3, 9824, 226]. Next we use indexing to select the corresponding rows from E
(row 3, row 4000, row 10532).
Another way to think about selecting token embeddings from the embedding
matrix is to represent input tokens as one-hot vectors of shape [1 × |V |], i.e., with
one-hot vector one dimension for each word in the vocabulary. Recall that in a one-hot vector all
the elements are 0 except one, the element whose dimension is the word’s index
in the vocabulary, which has value 1. So if the word “dessert” has index 3 in the
vocabulary, x3 = 1, and xi = 0 ∀i 6= 3, as shown here:
[0 0 1 0 0 0 0 ... 0 0 0 0]
1 2 3 4 5 6 7 ... ... |V|
Multiplying by a one-hot vector that has only one non-zero element xi = 1 simply
selects out the relevant row vector for word i, resulting in the embedding for word i,
as depicted in Fig. 6.11.
3 |V| 3 d
1 0010000…0000 ✕ E = 1
|V|
Figure 6.11 Selecting the embedding vector for word V3 by multiplying the embedding
matrix E with a one-hot vector with a 1 in index 3.
We can extend this idea to represent the entire input token sequence as a matrix
of one-hot vectors, one for each of the N input positions as shown in Fig. 6.12.
d
|V| d
0010000…0000
0000000…0010
1000000…0000 ✕ E =
… N
N 0000100…0000
| V|
Figure 6.12 Selecting the embedding matrix for the input sequence of token ids W by mul-
tiplying a one-hot matrix corresponding to W by the embedding matrix E.
by concatenating all the input vectors into one very long vector of shape [1 × dN].
Then we pass this input to our classifier and let it make its decision. This gives
us lots of information, at the cost of using a pretty large network. Second, we can
pool pool the N embeddings into a single embedding and then pass that single pooled
embedding to the classifier. Pooling gives us less information than would have been
present in all the original embeddings, but has the advantage of being small and ef-
ficient and is especially useful in tasks for which we don’t care as much about the
original word order. Let’s give an example of each: pooling for the sentiment task,
and concatenation for the language modeling task.
Pooling input embeddings for sentiment So let’s begin with seeing how pooling
can work for the sentiment classification task. The intuition of pooling is that for
sentiment, the exact position of the input (is some word like great the first word?
the second word?) is less important than the identity of the word itself.
A pooling function is a way to turn a set of embeddings into a single embedding.
For example, for a text with N input words/tokens w1 , ..., wN , we want to turn
the N row embeddings e(w1 ), ..., e(wN ) (each of dimensionality d) into a single
embedding also of dimensionality d.
mean-pooling There are various ways to pool. The simplest is mean-pooling: taking the mean
by summing the embeddings and then dividing by N:
N
1X
xmean = e(wi ) (6.21)
N
i=1
Here are the equations for this classifier assuming mean pooling:
The architecture is sketched in Fig. 6.13, where we also give the shapes for all the
relevant matrices.
max-pooling There are many other options for pooling, like max-pooling, in which case for
each dimension we take the element-wise max over all the inputs. The element-wise
max of a set of N vectors is a new vector whose kth element is the max of the kth
elements of all the N vectors.
Concatenating input embeddings for language modeling For sentiment analy-
sis we saw how to generate an output vector with probabilities over three classes:
positive, negative, or neutral, given as input a window of N input tokens, by first
pooling those token embeddings into a single embedding vector.
Now let’s consider language modeling: predicting upcoming words from prior
words. In this task we are given the same window of N input tokens, but our task
now is to predict the next token that should follow the window. We’ll sketch a
simple feedforward neural language model, drawing on an algorithm first introduced
by Bengio et al. (2003). The feedforward language model introduces many of the
important concepts of large language modeling that we will return to in Chapter 7
and Chapter 8.
Neural language models have many advantages over the n-gram language mod-
els of Chapter 3. Neural language models can handle much longer histories, can
134 C HAPTER 6 • N EURAL N ETWORKS
[dh⨉3] weights
U
h1 h2 h3 … h
dh
h [1⨉dh] Hidden layer
W [d⨉dh] weights
E E E |V|⨉d E matrix
shared across words
1 3 |V|
1 524 |V|
00 1 00 1 902 |V| N⨉|V| one-hot vectors
00 0 1 0 0
00 0 1 0 0
“dessert” = V3 “was” = V524 “great” = V902
generalize better over contexts of similar words, and are far more accurate at word-
prediction. On the other hand, neural net language models are slower, more com-
plex, need vast amounts of energy to train, and are less interpretable than n-gram
models, so for some smaller tasks an n-gram language model is still the right tool.
A feedforward neural language model is a feedforward network that takes as
input at time t a representation of some number of previous words (wt−1 , wt−2 , etc.)
and outputs a probability distribution over possible next words. Thus—like the n-
gram LM—the feedforward neural LM approximates the probability of a word given
the entire prior context P(wt |w1:t−1 ) by approximating based on the N − 1 previous
words:
In the following examples we’ll use a 4-gram example, so we’ll show a neural net to
estimate the probability P(wt = i|wt−3 , wt−2 , wt−1 ).
Neural language models represent words in this prior context by their embed-
dings, rather than just by their word identity as used in n-gram language models.
Using embeddings allows neural language models to generalize better to unseen
data. For example, suppose we’ve seen this sentence in training:
I have to make sure that the cat gets fed.
6.5 • E MBEDDINGS AS THE INPUT TO NEURAL NET CLASSIFIERS 135
but have never seen the words “gets fed” after the word “dog”. Our test set has the
prefix “I forgot to make sure that the dog gets”. What’s the next word? An n-gram
language model will predict “fed” after “that the cat gets”, but not after “that the dog
gets”. But a neural LM, knowing that “cat” and “dog” have similar embeddings, will
be able to generalize from the “cat” context to assign a high enough probability to
“fed” even after seeing “dog”.
W Nd⨉dh
...
Figure 6.14 Forward inference in a feedforward neural language model. At each timestep
t the network computes a d-dimensional embedding for each of the N = 3 context tokens (by
multiplying a one-hot vector by the embedding matrix E), and concatenates the three to get
the embedding e. This embedding e is multiplied by weight matrix W and then an activation
function is applied element-wise to produce the hidden layer h, which is then multiplied by
another weight matrix U. A softmax layer predicts at each output node i the probability that
the next word wt will be vocabulary word Vi . We show the context window size N as 3 just to
fit on the page, but in practice language modeling requires a much longer context.
size of 3, given one-hot input vectors for each input context word, are:
If we are using the network to classify into 3 or more classes, the loss function is
exactly the same as the loss for multinomial regression that we saw in Chapter 4 on
6.6 • T RAINING N EURAL N ETS 137
page 80. Let’s briefly summarize the explanation here for convenience. First, when
we have more than 2 classes we’ll need to represent both y and ŷ as vectors. Let’s
assume we’re doing hard classification, where only one class is the correct one.
The true label y is then a vector with K elements, each corresponding to a class,
with yc = 1 if the correct class is c, with all other elements of y being 0. Recall that
a vector like this, with one value equal to 1 and the rest 0, is called a one-hot vector.
And our classifier will produce an estimate vector with K elements ŷ, each element
ŷk of which represents the estimated probability p(yk = 1|x).
The loss function for a single example x is the negative sum of the logs of the K
output classes, each weighted by their probability yk :
K
X
LCE (ŷ, y) = − yk log ŷk (6.26)
k=1
We can simplify this equation further; let’s first rewrite the equation using the func-
tion 1{} which evaluates to 1 if the condition in the brackets is true and to 0 oth-
erwise. This makes it more obvious that the terms in the sum in Eq. 6.26 will be 0
except for the term corresponding to the true class for which yk = 1:
K
X
LCE (ŷ, y) = − 1{yk = 1} log ŷk
k=1
In other words, the cross-entropy loss is simply the negative log of the output proba-
bility corresponding to the correct class, and we therefore also call this the negative
negative log log likelihood loss:
likelihood loss
Plugging in the softmax formula from Eq. 6.9, and with K the number of classes:
exp(zc )
LCE (ŷ, y) = − log PK (where c is the correct class) (6.28)
j=1 exp(z j )
Let’s think about the negative log probability as a loss function. A perfect clas-
sifier would assign the correct class i probability 1 and all the incorrect classes prob-
ability 0. That means the higher p(ŷi ) (the closer it is to 1), the better the classifier;
p(ŷi ) is (the closer it is to 0), the worse the classifier. The negative log of this prob-
ability is a beautiful loss metric since it goes from 0 (negative log of 1, no loss)
to infinity (negative log of 0, infinite loss). This loss function also insures that as
probability of the correct answer is maximized, the probability of all the incorrect
answers is minimized; since they all sum to one, any increase in the probability of
the correct answer is coming at the expense of the incorrect answers.
The number K of classes of the output vector ŷ can be small or large. Perhaps
our task is 3-way sentiment, and then the classes might be positive, negative, and
neutral. Or if our task is deciding the part of speech of a word (i.e., whether it is a
noun or verb or adjective, etc.), then K is set of possible parts of speech in our tagset
(of which there are 17 in the tagset we will define in Chapter 17). And if our task
is language modeling, and our classifier is trying to predict which word is next, then
our set of classes is the set of words, which might be 50,000 or 100,000.
138 C HAPTER 6 • N EURAL N ETWORKS
Or for a network with one weight layer and softmax output (=multinomial logistic
regression), we could use the derivative of the softmax loss from Eq. 4.40, shown
for a particular weight wk and input xi
∂ LCE (ŷ, y)
= −(yk − ŷk )xi
∂ wk,i
= −(yk − p(yk = 1|x))xi
!
exp (wk · x + bk )
= − yk − PK xi (6.30)
j=1 exp (w j · x + b j )
But these derivatives only give correct updates for one weight layer: the last one!
For deep networks, computing the gradients for each weight is much more complex,
since we are computing the derivative with respect to weight parameters that appear
all the way back in the very early layers of the network, even though the loss is
computed only at the very end of the network.
The solution to computing this gradient is an algorithm called error backprop-
error back-
propagation agation or backprop (Rumelhart et al., 1986). While backprop was invented spe-
cially for neural networks, it turns out to be the same as a more general procedure
called backward differentiation, which depends on the notion of computation
graphs. Let’s see how that works in the next subsection.
operation left to right, passing the outputs of each computation as the input to the
next node.
forward pass
a a=3
e=a+d e=5
d=2
b=1
b d = 2b L=ce L=-10
c=-2
c
Figure 6.15 Computation graph for the function L(a, b, c) = c(a+2b), with values for input
nodes a = 3, b = 1, c = −2, showing the forward pass computation of L.
∂L
=e (6.33)
∂c
For the other two, we’ll need to use the chain rule:
∂L ∂L ∂e
=
∂a ∂e ∂a
∂L ∂L ∂e ∂d
= (6.34)
∂b ∂e ∂d ∂b
140 C HAPTER 6 • N EURAL N ETWORKS
d e
d e L
∂L = ∂L ∂e ∂e ∂L
∂d ∂e ∂d ∂d ∂e
downstream local upstream
gradient gradient gradient
Figure 6.16 Each node (like e here) takes an upstream gradient, multiplies it by the local
gradient (the gradient of its output with respect to its input), and uses the chain rule to compute
a downstream gradient to be passed on to a prior node. A node may have multiple local
gradients if it has multiple inputs.
Eq. 6.34 and Eq. 6.33 thus require five intermediate derivatives: ∂∂ Le , ∂∂ Lc , ∂∂ ae , ∂∂ de , and
∂d
∂ b , which are as follows (making use of the fact that the derivative of a sum is the
sum of the derivatives):
∂L ∂L
L = ce : = c, =e
∂e ∂c
∂e ∂e
e = a+d : = 1, =1
∂a ∂d
∂d
d = 2b : =2
∂b
In the backward pass, we compute each of these partials along each edge of the
graph from right to left, using the chain rule just as we did above. Thus we begin by
computing the downstream gradients from node L, which are ∂∂ Le and ∂∂ Lc . For node e,
we then multiply this upstream gradient ∂∂ Le by the local gradient (the gradient of the
output with respect to the input), ∂∂ de to get the output we send back to node d: ∂∂ Ld .
And so on, until we have annotated the graph all the way to all the input variables.
The forward pass conveniently already will have computed the values of the forward
intermediate variables we need (like d and e) to compute these derivatives. Fig. 6.17
shows the backward pass.
a=3
a
∂L = ∂L ∂e =-2
∂a ∂e ∂a e=5
e=d+a
d=2
b=1 ∂e ∂e
=1 =1 ∂L
b d = 2b ∂L = ∂L ∂e =-2 ∂a ∂d =-2 L=-10
∂e L=ce
∂L = ∂L ∂d =-4 ∂d ∂d ∂e ∂d
=2 ∂L
∂b ∂d ∂b ∂b =-2
∂e
c=-2 ∂L
=5
∂c
∂L =5 backward pass
c ∂c
Figure 6.17 Computation graph for the function L(a, b, c) = c(a + 2b), showing the backward pass computa-
tion of ∂∂ La , ∂∂ Lb , and ∂∂ Lc .
6.6 • T RAINING N EURAL N ETS 141
For the backward pass we’ll also need to compute the loss L. The loss function
for binary sigmoid output from Eq. 6.25 is
[1]
w11
*
w[1]
12 z[1] = a1[1] =
* 1
+ ReLU
x1
*
b[1]
1 z[2] =
w[2] a[2] = σ L (a[2],y)
x2 11 +
*
*
w[1] z[1] a[1] w[2]
21 * 2 = 2 = 12
+ ReLU
w[1] b[2]
22 1
b[1]
2
Figure 6.18 Sample computation graph for a simple 2-layer neural net (= 1 hidden layer) with two input units
and 2 hidden units. We’ve adjusted the notation a bit to avoid long equations in the nodes by just mentioning
[1]
the function that is being computed, and the resulting variable name. Thus the * to the right of node w11 means
[1]
that w11 is to be multiplied by x1 , and the node z[1] = + means that the value of z[1] is computed by summing
[1]
the three nodes that feed into it (the two products, and the bias term bi ).
The weights that need updating (those for which we need to know the partial
derivative of the loss function) are shown in teal. In order to do the backward pass,
we’ll need to know the derivatives of all the functions in the graph. We already saw
in Section 4.15 the derivative of the sigmoid σ :
dσ (z)
= σ (z)(1 − σ (z)) (6.38)
dz
142 C HAPTER 6 • N EURAL N ETWORKS
We’ll also need the derivatives of each of the other activation functions. The
derivative of tanh is:
d tanh(z)
= 1 − tanh2 (z) (6.39)
dz
The derivative of the ReLU is2
d ReLU(z) 0 f or z < 0
= (6.40)
dz 1 f or z ≥ 0
We’ll give the start of the computation, computing the derivative of the loss function
L with respect to z, or ∂∂Lz (and leaving the rest of the computation as an exercise for
the reader). By the chain rule:
∂L ∂ L ∂ a[2]
= [2] (6.41)
∂z ∂a ∂z
∂L
So let’s first compute ∂ a[2]
, taking the derivative of Eq. 6.37, repeated here:
h i
LCE (a[2] , y) = − y log a[2] + (1 − y) log(1 − a[2] )
! !
∂L ∂ log(a[2] ) ∂ log(1 − a[2] )
= − y + (1 − y)
∂ a[2] ∂ a[2] ∂ a[2]
1 1
= − y [2] + (1 − y) (−1)
a 1 − a[2]
y y−1
= − [2] + (6.42)
a 1 − a[2]
Next, by the derivative of the sigmoid:
∂ a[2]
= a[2] (1 − a[2] )
∂z
Finally, we can use the chain rule:
∂L ∂ L ∂ a[2]
=
∂z ∂ a[2] ∂ z
y y−1
= − [2] + a[2] (1 − a[2] )
a 1 − a[2]
= a[2] − y (6.43)
Continuing the backward computation of the gradients (next by passing the gra-
[2]
dients over b1 and the two product nodes, and so on, back to all the teal nodes), is
left as an exercise for the reader.
For logistic regression we can initialize gradient descent with all the weights and
biases having the value 0. In neural networks, by contrast, we need to initialize the
weights with small random numbers. It’s also helpful to normalize the input values
to have 0 mean and unit variance.
Various forms of regularization are used to prevent overfitting. One of the most
dropout important is dropout: randomly dropping some units and their connections from
the network during training (Hinton et al. 2012, Srivastava et al. 2014). At each
iteration of training (whenever we update parameters, i.e. each mini-batch if we are
using mini-batch gradient descent), we repeatedly choose a probability p and for
each unit we replace its output with zero with probability p (and renormalize the
rest of the outputs from that layer).
hyperparameter Tuning of hyperparameters is also important. The parameters of a neural net-
work are the weights W and biases b; those are learned by gradient descent. The
hyperparameters are things that are chosen by the algorithm designer; optimal val-
ues are tuned on a devset rather than by gradient descent learning on the training
set. Hyperparameters include the learning rate η, the mini-batch size, the model
architecture (the number of layers, the number of hidden nodes per layer, the choice
of activation functions), how to regularize, and so on. Gradient descent itself also
has many architectural variants such as Adam (Kingma and Ba, 2015).
Finally, most modern neural networks are built using computation graph for-
malisms that make it easy and natural to do gradient computation and parallelization
on vector-based GPUs (Graphic Processing Units). PyTorch (Paszke et al., 2017)
and TensorFlow (Abadi et al., 2015) are two of the most popular. The interested
reader should consult a neural network textbook for further details; some sugges-
tions are at the end of the chapter.
6.7 Summary
• Neural networks are built out of neural units, originally inspired by biological
neurons but now simply an abstract computational device.
• Each neural unit multiplies input values by a weight vector, adds a bias, and
then applies a non-linear activation function like sigmoid, tanh, or rectified
linear unit.
• In a fully-connected, feedforward network, each unit in layer i is connected
to each unit in layer i + 1, and there are no cycles.
• The power of neural networks comes from the ability of early layers to learn
representations that can be utilized by later layers in the network.
• Neural networks are trained by optimization algorithms like gradient de-
scent.
• Error backpropagation, backward differentiation on a computation graph,
is used to compute the gradients of the loss function for a network.
• Neural language models use a neural network as a probabilistic classifier, to
compute the probability of the next word given the previous n words.
• Neural language models can use pretrained embeddings, or can learn embed-
dings from scratch in the process of language modeling.
144 C HAPTER 6 • N EURAL N ETWORKS
Historical Notes
The origins of neural networks lie in the 1940s McCulloch-Pitts neuron (McCul-
loch and Pitts, 1943), a simplified model of the biological neuron as a kind of com-
puting element that could be described in terms of propositional logic. By the late
1950s and early 1960s, a number of labs (including Frank Rosenblatt at Cornell and
Bernard Widrow at Stanford) developed research into neural networks; this phase
saw the development of the perceptron (Rosenblatt, 1958), and the transformation
of the threshold into a bias, a notation we still use (Widrow and Hoff, 1960).
The field of neural networks declined after it was shown that a single perceptron
unit was unable to model functions as simple as XOR (Minsky and Papert, 1969).
While some small amount of work continued during the next two decades, a major
revival for the field didn’t come until the 1980s, when practical tools for building
deeper networks like error backpropagation became widespread (Rumelhart et al.,
1986). During the 1980s a wide variety of neural network and related architec-
tures were developed, particularly for applications in psychology and cognitive sci-
ence (Rumelhart and McClelland 1986b, McClelland and Elman 1986, Rumelhart
connectionist and McClelland 1986a, Elman 1990), for which the term connectionist or paral-
lel distributed processing was often used (Feldman and Ballard 1982, Smolensky
1988). Many of the principles and techniques developed in this period are foun-
dational to modern work, including the ideas of distributed representations (Hinton,
1986), recurrent networks (Elman, 1990), and the use of tensors for compositionality
(Smolensky, 1990).
By the 1990s larger neural networks began to be applied to many practical lan-
guage processing tasks as well, like handwriting recognition (LeCun et al. 1989) and
speech recognition (Morgan and Bourlard 1990). By the early 2000s, improvements
in computer hardware and advances in optimization and training techniques made it
possible to train even larger and deeper networks, leading to the modern term deep
learning (Hinton et al. 2006, Bengio et al. 2007). We cover more related history in
Chapter 13 and Chapter 15.
There are a number of excellent books on neural networks, including Goodfellow
et al. (2016) and Nielsen (2015).
CHAPTER
The literature of the fantastic abounds in inanimate objects magically endowed with
the gift of speech. From Ovid’s statue of Pygmalion to Mary Shelley’s story about
Frankenstein, we continually reinvent stories about
creating something and then having a chat with it.
Legend has it that after finishing his sculpture Moses,
Michelangelo thought it so lifelike that he tapped it
on the knee and commanded it to speak. Perhaps
this shouldn’t be surprising. Language is the mark
of humanity and sentience. conversation is the most
fundamental arena of language, the first kind of lan-
guage we learn as children, and the kind we engage in
constantly, whether we are teaching or learning, or-
dering lunch, or talking with our families or friends.
This chapter introduces the Large Language
Model, or LLM, a computational agent that can in-
teract conversationally with people. The fact that LLMs are designed for interaction
with people has strong implications for their design and use.
Many of these implications already became clear in a computational system from
60 years ago, ELIZA (Weizenbaum, 1966). ELIZA, designed to simulate a Rogerian
psychologist, illustrates a number of important issues with chatbots. For example
people became deeply emotionally involved and conducted very personal conversa-
tions, even to the extent of asking Weizenbaum to leave the room while they were
typing. These issues of emotional engagement and privacy mean we need to think
carefully about how we deploy language models and consider their effect on the
people who are interacting with them.
In this chapter we begin by introducing the computational principles of LLMs;
we’ll discuss their implementation in the transformer architecture in the following
chapter. The central new idea that makes LLMs possible is the idea of pretraining,
so let’s begin by thinking about the idea of learning from text, the basic way that
LLMs are trained.
We know that fluent speakers of a language bring an enormous amount of knowl-
edge to bear during comprehension and production. This knowledge is embodied in
many forms, perhaps most obviously in the vocabulary, the rich representations we
have of words and their meanings and usage. This makes the vocabulary a useful
lens to explore the acquisition of knowledge from text, by both people and machines.
Estimates of the size of adult vocabularies vary widely both within and across
languages. For example, estimates of the vocabulary size of young adult speakers of
American English range from 30,000 to 100,000 depending on the resources used
146 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
to make the estimate and the definition of what it means to know a word. A sim-
ple consequence of these facts is that children have to learn about 7 to 10 words a
day, every single day, to arrive at observed vocabulary levels by the time they are 20
years of age. And indeed empirical estimates of vocabulary growth in late elemen-
tary through high school are consistent with this rate. How do children achieve this
rate of vocabulary growth? Research suggests that the bulk of this knowledge acqui-
sition happens as a by-product of reading. Reading is a process of rich contextual
processing; we don’t learn words one at a time in isolation. In fact, at some points
during learning the rate of vocabulary growth exceeds the rate at which new words
are appearing to the learner! That suggests that every time we read a word, we are
also strengthening our understanding of other words that are associated with it.
Such facts are consistent with the distributional hypothesis of Chapter 5, which
proposes that some aspects of meaning can be learned solely from the texts we en-
counter over our lives, based on the complex association of words with the words
they co-occur with (and with the words that those words occur with). The distribu-
tional hypothesis suggests both that we can acquire remarkable amounts of knowl-
edge from text, and that this knowledge can be brought to bear long after its initial
acquisition. Of course, grounding from real-world interaction or other modalities
can help build even more powerful models, but even text alone is remarkably useful.
What made the modern NLP revolution possible is that large language models
can learn all this knowledge of language, context, and the world simply by being
taught to predict the next word, again and again, based on context, in a (very) large
corpus of text. In this chapter and the next we formalize this idea that we’ll call
pretraining pretraining—learning knowledge about language and the world from iteratively
predicting tokens in vast amounts of text—and call the resulting pretrained models
large language models. Large language models exhibit remarkable performance on
natural language tasks because of the knowledge they learn in pretraining.
What can language models learn from word prediction? Consider the examples
below. What kinds of knowledge do you think the model might pick up from learn-
ing to predict what word fills the underbar (the correct answer is shown in blue)?
Think about this for each example before you read ahead to the next paragraph:.
With roses, dahlias, and peonies, I was surrounded by flowers
The room wasn’t just big it was enormous
The square root of 4 is 2
The author of “A Room of One’s Own” is Virginia Woolf
The professor said that he
From the first sentence a model can learn ontological facts like that roses and
dahlias and peonies are all kinds of flowers. From the second, a model could learn
that “enormous” means something on the same scale as big but further along on
the scale. From the third sentence, the system could learn math, while from the
4th sentence facts about the world and historical authors. Finally, the last sentence,
if a model was exposed to such sentences repeatedly, it might learn to associate
professors only with male pronouns, or other kinds of associations that might cause
models to act unfairly to different people.
What is a large language model? As we saw back in Chapter 3, a language
model is simply a computational system that can predict the next word from previous
words. That is, given a context or prefix of words, a language model assigns a
probability distribution over the possible next words. Fig. 7.1 sketches this idea.
Of course we’ve already seen language models! We saw n-gram language mod-
els in Chapter 3 and briefly touched on the feedforward network applied to language
147
p(w|context)
output
all .44
the .33
your .15
input
context So long and thanks for ?
Figure 7.1 A large language model is a neural network that takes as input a context or
prefix, and outputs a distribution over possible next words.
p(w|context)
output
all .44
the .33
your .15
Fig. 7.2 shows the same example from Fig. 7.1, in which a language model
is given a text prefix and generates a possible completion. The model selects the
word all, adds that to the context, uses the updated context to get a new predictive
distribution, and then selects the from that distribution and generates it, and so
on. Notice that the model is conditioning on both the priming context and its own
subsequently generated outputs.
This kind of setting in which we iteratively predict and generate words left-to-
148 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
right from earlier words is often called causal or autoregressive language mod-
els. (We will introduce alternative non-autoregressive models, like BERT and other
masked language models that predict words using information from both the left and
the right, in Chapter 10.)
This idea of using computational models to generate text, as well as code, speech,
generative AI and images, constitutes the important new area called generative AI. Applying
LLMs to generate text has vastly broadened the scope of NLP, which historically
was focused more on algorithms for parsing or understanding text rather than gen-
erating it.
In the rest of the chapter, we’ll see that almost any NLP task can be modeled
as word prediction in a large language model, if we think about it in the right way,
and we’ll motivate and introduce the idea of prompting language models. We’ll
introduce specific algorithms for generating text from a language model, like greedy
decoding and sampling. We’ll introduce the details of pretraining, the way that
language models are self-trained by iteratively being taught to guess the next word
in the text from the prior words. We’ll sketch out the other two stages of language
model training: instruction tuning (also called supervised finetuning or SFT), and
alignment, concepts that we’ll return to in Chapter 9. And we’ll see how to evaluate
these models. Let’s begin, though, by talking about different kinds of language
models.
w w w
w w w w w
w w w w w w w w w w w w w
Figure 7.3 Three architectures for language models: decoders, encoders, and encoder-decoders. The arrows
sketch out the information flow in the three architectures. Decoders take tokens as input and generate tokens
as output. Encoders take tokens as input and produce an encoding (a vector representation of each token) as
output. Encoder-decoders take tokens as input and generate a series of tokens as output.
decoder The decoder is the architecture we’ve introduced above. It takes as input a series
of tokens, and iteratively generates an output token one at a time. The decoder is
the architecture used to create large language models like GPT, Claude, Llama, and
Mistral. The information flow in decoders goes left-to-right, meaning that the model
7.2 • C ONDITIONAL G ENERATION OF T EXT: T HE I NTUITION 149
predicts the next word only from the prior words. Decoders are generative models,
meaning that, given input tokens, they generate novel output tokens. We’ll discuss
decoders in the rest of this chapter and in Chapter 8.
encoder The encoder takes as input a sequence of tokens and outputs a vector repre-
sentation for each tokens. Encoders are usually masked language models, meaning
they are trained by masking out a word, and learning to predict it by looking at sur-
rounding words on both sides. Masked language models like BERT, RoBERTA, and
others in the BERT family are encoder models. Encoder models are not generative
models; they aren’t used to generate text. Instead encoder models are often used to
create classifiers, for example where the input is text and the output is a label, for
example for sentiment or topic or other classes. This is done by finetuning them
(training them on supervised data). We’ll introduce encoder models in Chapter 10.
encoder- The encoder-decoder takes as input a sequence of tokens and outputs a series
decoder
of tokens. What makes it different than the decoder-only models, is that an encoder-
decoder has a much looser relationship between the input tokens and the output
tokens, and they are used to map between different kinds of tokens. That is, in an
encoder-decoder, the output tokens might be very different token-set or be much
longer or shorter than the input tokens. For example encoder-decoder architectures
are used for machine translation, where the input tokens are in one language and and
the output tokens are in another language, and probably a different length than the
input. Encoder-decoder architectures are also used for speech recognition, where the
input is tokens representing speech, and the output is tokens representing text. We’ll
introduce the encoder-decoder architecture for machine translation in Chapter 12,
and for speech recognition in Chapter 15.
These three architectures can be built out of many kinds of neural networks.
The most widely used network type today is the transformer that we’ll introduce
in Chapter 8. In a transformer, each input token is processed by a column of trans-
former layers, each layer composed of a series of different kinds of subnetworks. In
Chapter 13 we’ll introduce an earlier architecture that is still relevant, the LSTM,
a kind of recurrent neural network . And there are many more recent architectures
such as the state space models.
We’ll focus on transformers for much of this book, but for the purposes of this
chapter, we’ll be architecture-agnostic: we’ll treat network that implements the de-
coder as a black box. The input to this black box is a sequence of tokens, and the
output to the box is a distribution over tokens that we can sample from. We’ll de-
scribe the mechanisms for learning and decoding in a network-agnostic manner.
We’ll talk in future sections about all the details, but in this section our goal is
just to establish the intuition. How can simply computing the probability of the next
token help an LLM do all sorts of different language-related tasks?
Imagine we want to do a classification tasks like sentiment analysis. We can treat
this as conditional generation by giving a language model a context like:
The sentiment of the sentence ‘‘I like Jackie Chan" is:
and comparing the conditional probability of the following token “positive” and the
following token “negative” to see which is higher. That is, as sketched in Fig. 7.4,
we compare these two probabilities:
P(“positive”|“The sentiment of the sentence ‘I like Jackie Chan’ is:”)
P(“negative”|“The sentiment of the sentence ‘I like Jackie Chan’ is:”)
If the token “positive” is more probable, we could say the sentiment of the sen-
prob
“positive” ?
“negative” ?
tence is positive, otherwise if the token “negative” is more probable we say the
sentiment is negative.
This same intuition can help us perform a task like question answering, in which
the system is given a question and must give a textual answer. We can cast the task
of question answering as token prediction by giving a language model a question
and a token like A: suggesting that an answer should come next, like this:
Q: Who wrote the book ‘‘The Origin of Species"? A:
Again, we can ask a language model to compute the probability distribution over
possible next tokens given this prefix, computing the following probability
P(w|Q: Who wrote the book ‘‘The Origin of Species"? A:)
and look at which tokens w have high probabilities. As Fig. 7.5 suggests, we might
expect to see that Charles is very likely, and then if we choose Charles and add
that to our prefix and compute the probability over tokens with this prefix:
P(w|Q: Who wrote the book ‘‘The Origin of Species"? A: Charles)
we might now see that Darwin is the most probable token, and select it.
7.3 Prompting
This simple idea of contextual generation is already very powerful, but becomes
more powerful when language models are specially trained to answer questions and
7.3 • P ROMPTING 151
prob
Charles ?
token ?
token ?
Transformer (or other decoder) token ?
Figure 7.6 Sample 2-shot prompt from MMLU testing high-school computer science. (The
correct answer is (B)).
Demonstrations are generally drawn from a labeled training set. They can be
selected by hand, or the choice of demonstrations can be optimized by using an op-
timizer like DSPy (Khattab et al., 2024) to automatically chose the set of demonstra-
tions that most increases task performance of the prompt on a dev set. The number
of demonstrations doesn’t need to be large; more examples seem to give diminish-
ing returns, and too many examples seems to cause the model to overfit to the exact
examples. The primary benefit of demonstrations seems more to demonstrate the
task and the format of the output rather than demonstrating the right answers for
any particular question. In fact, demonstrations that have incorrect answers can still
improve a system (Min et al., 2022; Webson and Pavlick, 2022).
Prompts are a way to get language models to generate text, but prompts can
also can be viewed as a learning signal. This is especially clear when a prompt has
demonstrations, since the demonstrations can help language models learn to perform
novel tasks from these examples of the new task. This kind of learning is different
than pretraining methods for setting language model weights via gradient descent
methods that we will describe below. The weights of the model are not updated by
prompting; what changes is just the context and the activations in the network.
We therefore call the kind of learning that takes place during prompting in-
in-context
learning context learning—learning that improves model performance or reduces some loss
but does not involve gradient-based updates to the model’s underlying parameters.
system prompt Large language models generally have a system prompt, a single text prompt
that is the first instruction to the language model, and which defines the task or
role for the LM, and sets overall tone and context. The system prompt is silently
prepended to any user text. So for example a minimal system prompt that creates
a multi-turn assistant conversation might be the following including some special
metatokens:
7.4 • G ENERATION AND S AMPLING 153
It’s also possible to create system prompts for other tasks, like the following
prompt for creating a general grammar-checker (Anthropic, 2025):
Your task is to take the text provided and rewrite it into
a clear, grammatically correct version while preserving
the original meaning as closely as possible. Correct any
spelling mistakes, punctuation errors, verb tense issues,
word choice problems, and other grammatical mistakes.
Each user can then make a prompt to have the system fix the grammar of a particular
piece of text.
In all these cases, the system prompt is prepended to any user prompts or queries,
and the entire string is taking as the context for conditional generation by the lan-
guage model.
The generation depends on the probability of each token, so let’s remind our-
selves where this probability distribution comes from. The internal networks for
language models (whether transformers or alternatives like LSTMs or state space
models) generate scores called logits (real valued numbers) for each token in the vo-
cabulary. This score vector u is then normalized by softmax to be a legal probability
distribution, just as we saw for logistic regression in Chapter 4. So if we have a logit
vector u of shape [1 × |V |] that gives a score for each possible next token, we can
pass it through a softmax to get a vector y, also of shape [1 × |V |], which assigns a
probability to each token in the vocabulary, as shown in the following equation:
y = softmax(u) (7.1)
Fig. 7.7 shows an example in which the softmax is computed for pedagogical pur-
poses on a simplified vocabulary of only 4 words.
u y
logits softmax probabilities
all 1.2 all .44
the 0.9 the .33
your 0.1 your .15
that -0.5 that .08
Transformer (or other decoder)
Now given this probability distribution over tokens, we need to select one token
to generate. The task of choosing a token to generate based on the model’s probabil-
decoding ities is often called decoding. As we mentioned above, decoding from a language
model in a left-to-right manner (or right-to-left for languages like Arabic in which
we read from right to left), and thus repeatedly choosing the next token conditioned
autoregressive
generation on our previous choices is called autoregressive generation.1
Fig. 7.8 shows that in our example, the model chooses to generate all.
1 Technically an autoregressive model predicts a value at time t based on a linear function of the values
at times t − 1, t − 2, and so on. Although language models are not linear (since, as we will see, they have
many layers of non-linearities), we loosely refer to this generation technique as autoregressive since the
token generated at each time step is conditioned on the token selected by the network from the previous
step. As we’ll see, alternatives like the masked language models of Chapter 10 are non-causal because
they can predict tokens based on both past and future tokens).
7.4 • G ENERATION AND S AMPLING 155
u y
logits softmax probabilities
In practice, however, we don’t use greedy decoding with large language models.
A major problem with greedy decoding is that because the tokens it chooses are
(by definition) extremely predictable, the resulting text is generic and often quite
repetitive. Indeed, greedy decoding is so predictable that it is deterministic; if the
context is identical, and the probabilistic model is the same, greedy decoding will
always result in generating exactly the same string.
We’ll see in Chapter 12 that an extension to greedy decoding called beam search
works well in tasks like machine translation, which are very constrained in that we
are always generating a text in one language conditioned on a very specific text in
another language.
In most other tasks, however, people prefer text which has been generated by
sampling methods that introduce a bit more diversity into the generations.
random
sampling The algorithm is called random sampling, or random multinomial sampling
(because we are sampling from a multinomial distribution across words). We can
formalize random sampling as follows: we are generating a sequence of tokens
{w1 , w2 , . . . , wN } until we hit the end-of-sequence token, using x ∼ p(x) to mean
‘choose x by sampling from the distribution p(x)’:
i←1
wi ∼ p(w)
while wi != EOS
i←i + 1
wi ∼ p(wi | w<i )
u y
sample
logits softmax probabilities
a word
all 1.2 all .44
the 0.9 the .33
your 0.1 your .15 the
Transformer (or other decoder) that -0.5 that .08
… …
Figure 7.9 Random multinomial sampling: we randomly chose a word according to its
probability.
Alas, it turns out random sampling doesn’t work well either. The problem is that
even though random sampling is mostly going to generate sensible, high-probable
tokens, there are many odd, low-probability tokens in the tail of the distribution, and
even though each one is low-probability, if you add up all the rare tokens, they con-
stitute a large enough portion of the distribution that they get chosen often enough
to result in generating weird sentences.
In other words, greedy decoding is too boring, and random sampling is too ran-
dom. We need something that doesn’t greedily choose the top choice every time, but
doesn’t stray down too far into the very low-probability events.
There are three standard sampling methods that modify random sampling to ad-
dress these issues. We’ll describe the most common, temperature sampling here,
and talk about two others (top-k and top-p) in the next chapter.
y = softmax(u) (7.3)
y = softmax(u/τ) (7.4)
7.5 • T RAINING L ARGE L ANGUAGE M ODELS 157
That is, normally we convert from logits to softmax as shown in Fig. 7.10(a).
But when we use a temperature parameter we first scale the logit as in Fig. 7.10(b).
u y u softmax y
logits softmax probabilities logits with probabilities
temperature
a a
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exp(a)
Z Z
b exp(b)
where
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b exp(b/⌧ )
where
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c Z
+exp(b)
c Z
+exp(b/⌧ )
d exp(c)
+exp(c)
d exp(c/⌧ )
+exp(c/⌧ )
… Z … Z
exp(d) +exp(d) exp(d/⌧ ) +exp(d/⌧ )
Z +... Z +...
… …
(a) (b)
Figure 7.10 (a): Normal softmax without temperature scaling (b) Adding temperature scaling to the softmax
by first dividing by the temperature parameter τ.
Why does dividing by τ increase the high probability elements and decrease the
low probability elements in the vector over vocabulary items? When τ is 1, we are
doing normal softmax, and so when τ is close to 1 the distribution doesn’t change
much. But the lower τ is, the larger the scores being passed to the softmax (because
dividing by a smaller fraction τ ≤ 1 results in making each score larger).
Recall that one of the useful properties of a softmax is that it tends to push high
values toward 1 and low values toward 0. Thus when larger numbers are passed to
a softmax the result is a distribution with increased probabilities of the most high-
probability tokens and decreased probabilities of the low probability tokens, making
the distribution more greedy. By contrast, as as τ approaches 0 the probability of the
most likely word approaches 1, resulting in greedy decoding..
The intuition for temperature sampling comes from thermodynamics, where a
system at a high temperature is very flexible and can explore many possible states,
while a system at a lower temperature is likely to explore a subset of lower energy
(better) states. In low-temperature sampling, we smoothly increase the probability
of the most probable tokens and decrease the probability of the rare tokens.
Fig. 7.11 shows a schematic example again simplified to have a vocabulary with
only 4 tokens (all, the, your, that), and showing how different temperature values
influence the probabilities computed from the initial logits. i τ = 1 is the normal
softmax, and we can see how setting τ = 0.5 increases the probability of the top
candidate from .55 to .59. Setting τ = 0.1 increases the probability of the top candi-
date from .05, getting us close to greedy decoding.
We can also see in Fig. 7.11 some other options for situations where we may want
to flatten the word probability distribution instead of making it greedy. Temperature
sampling can help with this situation too, in this case high-temperature sampling,
in which case we use τ > 1.
dy ax rm
ee ftm ifo
gr so un
to al to
se r m se
c l o no clo
logits 𝜏=0.1 𝜏=0.5 𝜏=1 𝜏=10 𝜏=100
Instruction Preference
1. Pretraining 2. Tuning 3. Alignment
Pretrained Instruction
Aligned LLM
LLM Tuned LLM
Figure 7.12 Three stages of training large language models: pretraining, instruction tuning,
and preference alignment.
the next word in the corpus!) over time it well get better and better at predicting
the correct next word. We call such a model self-supervised because we don’t have
to add any special gold labels to the data; the natural sequence of words is its own
supervision! We simply train the model to minimize the error in predicting the true
next word in the training sequence.
In practice, training the language model means setting the parameters of the
underlying architecture. The transformer that we will introduce in the next chapter
has various weight matrices for its feedforward and attention components. Like any
other neural architecture, they will be trained by error backpropagation with gradient
descent. So all we need is a loss function to minimize and pass back through the
network. The loss function we use for language modeling is the cross-entropy loss
function we’ve now seen twice, in Chapter 4 and Chapter 6.
Recall that the cross-entropy loss measures the difference between a predicted
probability distribution and the correct distribution. The probability distribution is
over the token vocabulary, making the loss be:
X
LCE = − yt [w] log ŷt [w] (7.5)
w∈V
In the case of language modeling, the correct distribution yt comes from knowing the
next word. This is represented as a one-hot vector corresponding to the vocabulary
where the entry for the actual next word is 1, and all the other entries are 0. Thus,
the cross-entropy loss for language modeling is determined by the probability the
model assigns to the correct next token (all other tokens get multiplied by zero by
the first term in Eq. 7.5).
So without loss of generality we can say that at time t the cross-entropy loss in
Eq. 7.5 can be simplified as the negative log probability the model assigns to the next
word in the training sequence, − log p(wt+1 ), or more formally, using ŷ to mean the
the vector of estimated token probabilities from the language model:
LCE (ŷt , yt ) = − log ŷt [wt+1 ] (7.6)
Thus at each word position t of the input, the model takes as input the correct se-
quence of tokens w1:t , and uses them to compute a probability distribution over
160 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
possible next tokens so as to compute the model’s loss for the next token wt+1 . Then
we move to the next word, we ignore what the model predicted for the next word
and instead use the correct sequence of tokens w1:t+1 to get the model to estimate the
probability of token wt+2 . This idea that we always give the model the correct his-
tory sequence to predict the next word (rather than feeding the model its best guess
teacher forcing from the previous time step) is called teacher forcing.
Fig. 7.13 illustrates the general training approach. At each step, given all the
preceding tokens, the language model produces an output distribution over the en-
tire vocabulary. During training, the probability assigned to the correct word is used
to calculate the cross-entropy loss for each item in the sequence. The loss for each
batch is the average cross-entropy loss over the entire sequence of negative log prob-
abilities, or more formally:
T
1X
LCE (batch of length T) = − log ŷt [wt ] (7.7)
T
t=1
The weights in the network are then adjusted to minimize this average cross-entropy
loss over the batch via gradient descent (Fig. 4.6), using error backpropagation on
the computation graph to compute the gradient. Training adjusts all the weights
of the network. For the transformer model we will introduce in the next chapter,
these weights include the embedding matrix E that contains the embeddings for
each word. Thus embeddings will be learned that are most successful at predicting
upcoming words.
LLM …
Web text is usually taken from corpora of automatically-crawled web pages like
common crawl the common crawl, a series of snapshots of the entire web produced by the non-
profit Common Crawl ([Link] that each have billions of
webpages. Various versions of common crawl data exist, such as the Colossal Clean
Crawled Corpus (C4; Raffel et al. 2020), a corpus of 156 billion tokens of English
that is filtered in various ways (deduplicated, removing non-natural language like
code, sentences with offensive words from a blocklist). This C4 corpus seems to
consist in large part of patent text documents, Wikipedia, and news sites (Dodge
et al., 2021).
Wikipedia plays a role in lots of language model training, as do corpora of books.
The Pile The Pile (Gao et al., 2020) is an 825 GB English text corpus that is constructed by
publicly released code, containing again a large amount of text scraped from the web
as well as books and Wikipedia; Fig. 7.14 shows its composition. Dolma is a larger
open corpus of English, created with public tools, containing three trillion tokens,
which similarly consists of web text, academic papers, code, books, encyclopedic
materials, and social media (Soldaini et al., 2024).
Figure 7.14 The Pile corpus, showing the size of different components, color coded as
academic (articles from PubMed and ArXiv, patents from the USPTA; internet (webtext in-
cluding a subset of the common crawl as well as Wikipedia), prose (a large corpus of books),
dialogue (including movie subtitles and chat data), and misc.. Figure from Gao et al. (2020).
Filtering for quality and safety Pretraining data drawn from the web is filtered
for both quality and safety. Quality filters are classifiers that assign a score to each
document. Quality is of course subjective, so different quality filters are trained
in different ways, but often to value high-quality reference corpora like Wikipedia,
PII books, and particular websites and to avoid websites with lots of PII (Personal Iden-
tifiable Information) or adult content. Filters also remove boilerplate text which is
very frequent on the web. Another kind of quality filtering is deduplication, which
can be done at various levels, so as to remove duplicate documents, duplicate web
pages, or duplicate text. Quality filtering generally improves language model per-
formance (Longpre et al., 2024b; Llama Team, 2024).
Safety filtering is again a subjective decision, and often includes toxicity detec-
tion based on running off-the-shelf toxicity classifiers. This can have mixed results.
One problem is that current toxicity classifiers mistakenly flag non-toxic data if it
162 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
7.5.3 Finetuning
Although the vast pretraining data for large language models includes text from
many domains, we might want to apply it in a new domain or task that didn’t appear
sufficiently in the pretraining data. For example, we might want a language model
that’s specialized to legal or medical text. Or we might have a multilingual language
model that knows many languages but might benefit from some more data in our
particular language of interest.
In such cases, we can simply continue training the model on relevant data from
the new domain or language (Gururangan et al., 2020). This process of taking a fully
pretrained model and running additional training passes using the cross-entropy loss
finetuning on some new data is called finetuning. The word “finetuning” means the process
of taking a pretrained model and further adapting some or all of its parameters to
some new data. Over the next few chapters we’ll see a number of different ways
that the word ‘finetuning’ is used, based on exactly which parameters get updated.
The method we describe here, in which we just continue to train, as if the new data
continued
pretraining was at the end of our pretraining data, can also be called continued pretraining.
Fig. 7.15 sketches the paradigm.
7.6 • E VALUATING L ARGE L ANGUAGE M ODELS 163
Fine-
Pretraining Data tuning
Pretrained LM Data Fine-tuned LM
… … … … … …
Pretraining Fine-tuning
Figure 7.15 Pretraining and finetuning. A pre-trained model can be finetuned to a particular
domain or dataset. There are many different ways to finetune, depending on exactly which
parameters are updated from the finetuning data: all the parameters, some of the parameters,
or only the parameters of specific extra circuitry, as we’ll see in future chapters.
7.6.1 Perplexity
As we first saw in Chapter 3, one way to evaluate language models is to measure
how well they predict unseen text. A better language model is better at predicting
upcoming words, and so it will be less surprised by (i.e., assign a higher probability
to) each word when it occurs in the test set.
If we want to know which of two language models is a better model of some text,
we can just see which assigns it a higher probability, or in practice, since we mostly
deal with probabilities in log space, we see which assigns a higher log likelihood.
We’ve been talking about predicting one word at a time, computing the probabil-
ity of the next token wi from the prior context: P(wi |w<i ). But of course as we saw
in Chapter 3 the chain rule allows us to move between computing the probability of
the next token and computing the probability of a whole text:
We can compute the probability of text just by multiplying the conditional proba-
bilities for each token in the text. The resulting (log) likelihood of a text is a useful
metric for comparing how good two language models are on that text:
n
Y
log likelihood(w1:n ) = log P(wi |w<i ) (7.9)
i=1
However, we often use another metric other than log likelihood to evaluate language
models. The reason is that the probability of a test set (or any sequence) depends
on the number of words or tokens in it. In fact, the probability of a test set gets
164 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
smaller the longer the text is; this is clear from the chain rule, since if we are mul-
tiplying more probabilities, and each probability by definition is less than zero, the
product will get smaller and smaller. So it’s useful to have a metric that is per-token,
normalized by length, so we could compare across texts of different lengths.
perplexity A function of probability called perplexity is such a length-normalized metric.
Recall from page 45 that the perplexity of a model θ on an unseen test set is the
inverse probability that θ assigns to the test set (one over the probability of the test
set), normalized by the test set length in tokens. For a test set of n tokens w1:n , the
perplexity is
1
Perplexityθ (w1:n ) = Pθ (w1:n )− n
s
1
= n (7.10)
Pθ (w1:n )
Note that because of the inverse in Eq. 7.10, the higher the probability of the word
sequence, the lower the perplexity. Thus the the lower the perplexity of a model on
the data, the better the model. Minimizing perplexity is equivalent to maximizing
the test set probability according to the language model. Why does perplexity use
the inverse probability? The inverse arises from the original definition of perplexity
from cross-entropy rate in information theory; for those interested, the explanation is
in Section 3.7. Meanwhile, we just have to remember that perplexity has an inverse
relationship with probability.
One caveat: because perplexity depends on the number of tokens n in a text, it
is very sensitive to differences in the tokenization algorithm. That means that it’s
hard to exactly compare perplexities produced by two language models if they have
very different tokenizers. For this reason perplexity is best used when comparing
language models that use the same tokenizer.
One of the reasons that the government discourages and regulates monopo-
lies is that
(A) producer surplus is lost and consumer surplus is gained.
(B) monopoly prices ensure productive efficiency but cost society allocative
efficiency.
(C) monopoly firms do not engage in significant research and development.
(D) consumer surplus is lost with higher prices and lower levels of output.
Fig. 7.16 shows the way MMLU turns these questions into prompted tests of a
language model, in this case showing an example prompt with 2 demonstrations.
The following are multiple choice questions about high school mathematics.
How many numbers are in the list 25, 26, ..., 100?
(A) 75 (B) 76 (C) 22 (D) 23
Answer: B
If 4 daps = 7 yaps, and 5 yaps = 3 baps, how many daps equal 42 baps?
(A) 28 (B) 21 (C) 40 (D) 30
Answer:
Figure 7.16 Sample 2-shot prompt from MMLU testing high-school mathematics. (The
correct answer is (C)).
sizes. Big models also use more energy, and we prefer models that use less energy,
both to reduce the environmental impact of the model and to reduce the financial
cost of building or deploying it. We can target our evaluation to these factors by
measuring performance normalized to a given compute or memory budget. We can
also directly measure the energy usage of our model in kWh or in kilograms of CO2
emitted (Strubell et al., 2019; Henderson et al., 2020; Liang et al., 2023).
Another feature that a language model evaluation can measure is fairness. We
know that language models are biased, exhibiting gendered and racial stereotypes,
or decreased performance for language from or about certain demographics groups.
There are language model evaluation benchmarks that measure the strength of these
biases, such as StereoSet (Nadeem et al., 2021), RealToxicityPrompts (Gehman
et al., 2020), and BBQ (Parrish et al., 2022) among many others. We also want
language models whose performance is equally fair to different groups. For exam-
ple, we could choose an evaluation that is fair in a Rawlsian sense by maximizing
the welfare of the worst-off group (Rawls, 2001; Hashimoto et al., 2018; Sagawa
et al., 2020).
Finally, there are many kinds of leaderboards like Dynabench (Kiela et al., 2021)
and general evaluation protocols like HELM (Liang et al., 2023); we will return to
these in later chapters when we introduce evaluation metrics for specific tasks like
question answering and information retrieval.
led to harm or death. We’ll return to the issue of hallucination and factuality in Chap-
ter 11 where we introduce proposed mitigation methods like retrieval augmented
generation, and Chapter 9 where we discussed safety tuning and alignment.
A system can also harm users by verbally attacking them, or creating represen-
tational harms (Blodgett et al., 2020) for example by generating abusive or harmful
stereotypes (Cheng et al., 2023) and negative attitudes (Brown et al., 2020; Sheng
et al., 2019) that demean particular groups of people; both abuse and stereotypes
can cause psychological harm to users. Gehman et al. (2020) show that even com-
pletely non-toxic prompts can lead large language models to output hate speech and
abuse their users. Liu et al. (2020) testing how systems responded to pairs of simu-
lated user turns that were identical except for mentioning different genders or race.
They found, for example, that simple changes like using the word ‘she’ instead of
‘he’ in a sentence caused systems to respond more offensively and with more nega-
tive sentiment. Hofmann et al. (2024) found that LLMs were likely to discriminate
against people just because they used particular dialects like African-American En-
Tay glish. Again, these problems predate large language models. Microsoft’s 2016 Tay
chatbot, for example, was taken offline 16 hours after it went live, when it began
posting messages with racial slurs, conspiracy theories, and personal attacks on its
users. Tay had learned these biases and actions from its training data, including
from users who seemed to be purposely teaching the system to repeat this kind of
language (Neff and Nagy 2016).
Another important ethical and safety issue is privacy. Privacy has been a con-
cern from the very beginning of computing when Weizenbaum designed the chatbot
ELIZA as an experiment in computational therapy (Weizenbaum, 1966). First, peo-
ple became deeply emotionally involved and conducted very personal conversations
with the ELIZA chatbot, even to the extent of asking Weizenbaum to leave the room
while they were typing. When Weizenbaum suggested that he might want to store
the ELIZA conversations, people immediately pointed out that this would violate
people’s privacy.
Users are likely to give quite personal information to large language models as
well, and indeed the most common current LLM use case is for personal advice and
support (Zao-Sanders, 2025). And the more human-like a system, the more users
are likely to disclose private information, and yet less likely to worry about the harm
of this disclosure (Ischen et al., 2019). We discussed above that pretraining data
also is likely to have private information like phone numbers and addresses. This is
problematic because large language models can leak information from their training
data. That is, an adversary can extract training-data text from a language model
such as a person’s name, phone number, and address (Henderson et al. 2017, Carlini
et al. 2021). This becomes even more problematic when large language models are
trained on extremely sensitive private datasets such as electronic health records.
A related safety issue is emotional dependence. Reeves and Nass (1996) show
that people tend to assign human characteristics to computers and interact with them
in ways that are typical of human-human interactions. They interpret an utterance in
the way they would if it had spoken by a human, (even though they are aware they
are talking to a computer). Thus LLMs have had significant influences on people’s
cognitive and emotional state, leading to problems like emotional dependence on
LLMs. These issues (emotional engagement and privacy) mean we need to think
carefully about the impact of LLMs on the people who are interacting with them.
In addition to their ability to harm their users in these ways, LLMs may carry out
additional harmful activities themselves, especially as agent-based paradigms makes
168 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
7.8 Summary
This chapter has introduced the large language model. Here’s a summary of the
main points that we covered:
• A large language model is a system that can predict the next word for pre-
vious words given a context or prefix of words, and use this prediction to
conditionally generate text.
• There are three major architectures for language models: the encoder, the
decoder, and the encoder-decoder. The well-known large language models
used for generating text are all decoder models; we’ll describe encoders in
Chapter 10 and encoder-decoders in Chapter 12.
• Many NLP tasks—such as question answering and sentiment analysis— can
be cast as tasks of word prediction and addressed with large language models.
• We instruct language models via a prompt, a text string that a user issues
to a language model to get the model to do something useful by iteratively
generating tokens conditioned on the prompt.
• The process of finding effective prompts for a task is known as prompt engi-
neering.
• The choice of which word to generate in large language models is done by
sampling from the distribution of possible next words.
H ISTORICAL N OTES 169
Historical Notes
As we discussed in Chapter 3, the earliest language models were the n-gram lan-
guage models developed (roughly simultaneously and independently) by Fred Je-
linek and colleagues at the IBM Thomas J. Watson Research Center, and James
Baker at CMU. It was the Jelinek and the IBM team who first coined the term lan-
guage model to mean a model of the way any kind of linguistic property (grammar,
semantics, discourse, speaker characteristics), influenced word sequence probabil-
ities (Jelinek et al., 1975). They contrasted the language model with the acoustic
model which captured acoustic/phonetic characteristics of phone sequences.
N-gram language models were very widely used over the next 40 years, across
a wide variety of NLP tasks like speech recognition and machine translations, often
as one of multiple components of the model. The contexts for these n-gram models
grew longer, with 5-gram models used quite commonly by very efficient LM toolkits
(Stolcke, 2002; Heafield, 2011).
The roots of the neural large language model lie in multiple places. One was
the application in the 1990s, again in Jelinek’s group at IBM Research, of discrim-
inative classifiers to language models. Roni Rosenfeld in his dissertation (Rosen-
feld, 1992) first applied logistic regression (under the name maximum entropy or
maxent models) to language modeling in that IBM lab, and published a more fully
formed version in Rosenfeld (1996). His model integrated various sorts of infor-
mation in a logistic regression predictor, including n-gram information along with
other features from the context, including distant n-grams and pairs of associated
words called trigger pairs. Rosenfeld’s model prefigured modern language models
by being a statistical word predictor trained in a self-supervised manner simply by
learning to predict upcoming words in a corpus.
Another was the first use of pretrained embeddings to model word meaning in
the LSA/LSI models (Deerwester et al., 1988). Recall from the history section of
170 C HAPTER 7 • L ARGE L ANGUAGE M ODELS
Chapter 5 that in LSA (latent semantic analysis) a term-document matrix was trained
on a corpus and then singular value decomposition was applied and the first 300
dimensions were used as a vector embedding to represent words. It was Landauer
et al. (1997) who first used the word “embedding”. In addition to their development
of the idea of pretraining and of embeddings, the LSA community also developed
ways to combine LSA embeddings with n-grams in an integrated language model
(Bellegarda, 1997; Coccaro and Jurafsky, 1998).
In a very influential series of papers developing the idea of neural language
models, (Bengio et al. 2000; Bengio et al. 2003; Bengio et al. 2006), Yoshua Ben-
gio and colleagues drew on the central ideas of both these lines of self-supervised
language modeling work (the discriminatively trained word predictor, and the pre-
trained embeddings). Like the maxent models of Rosenfeld, Bengio’s model used
the next word in running text as its supervision signal. Like the LSA models, Ben-
gio’s model learned an embedding, but unlike the LSA models did it as part of the
process of language modeling. The Bengio et al. (2003) model was a neural lan-
guage model: a neural network that learned to predict the next word from prior
words, and did so via learning embeddings as part of the prediction process.
The neural language model was extended in various ways over the years, perhaps
most importantly in the form of the RNN language model of Mikolov et al. (2010)
and Mikolov et al. (2011). The RNN language model was perhaps the first neural
model that was accurate enough to surpass the performance of a traditional 5-gram
language model.
Soon afterwards, Mikolov et al. (2013a) and Mikolov et al. (2013b) proposed to
simplify the hidden layer of these neural net language models to create pretrained
word2vec word embeddings.
The static embedding models like LSA and word2vec instantiated a particular
model of pretraining: a representation was trained on a pretraining dataset, and then
the representations could be used in further tasks. Dai and Le (2015) and Peters
et al. (2018) reframed this idea by proposing models that were pretrained using a
language model objective, and then the identical model could be either frozen and
directly applied for language modeling or further finetuned still using a language
model objective. For example ELMo used a biLSTM self-supervised on a large
pretrained dataset using a language model objective, then finetuned on a domain-
specific dataset, and then froze the weights and added task-specific heads. The
ELMo work was particularly influential and its appearance was perhaps the mo-
ment when it became clear to the community that language models could be used as
a general solution for NLP problems.
Transformers were first applied as encoder-decoders (Vaswani et al., 2017) and
then to masked language modeling (Devlin et al., 2019) (as we’ll see in Chapter 12
and Chapter 10). Radford et al. (2019) then showed that the transformer-based au-
toregressive language model GPT2 could perform zero-shot on many NLP tasks like
summarization and question answering.
The technology used for language models can also be applied to other domains
foundation and tasks, like vision, speech, and genetics. The term foundation model is some-
model
times used as a more general term for this use of large language model technology
across domains and areas, when the elements we are computing over are not nec-
essarily words. Bommasani et al. (2021) is a broad survey that sketches the op-
portunities and risks of foundation models, with special attention to large language
models.
CHAPTER
8 Transformers
In this chapter we introduce the transformer, the standard architecture for build-
ing large language models. As we discussed in the prior chapter, transformer-based
large language models have completely changed the field of speech and language
processing. Indeed, every subsequent chapter in this textbook will make use of them.
As with the previous chapter, we’ll focus for this chapter on the use of transformers
to model left-to-right (sometimes called causal or autoregressive) language model-
ing, in which we are given a sequence of input tokens and predict output tokens one
by one by conditioning on the prior context.
The transformer is a neural network with a specific structure that includes a
mechanism called self-attention or multi-head attention.1 Attention can be thought
of as a way to build contextual representations of a token’s meaning by attending to
and integrating information from surrounding tokens, helping the model learn how
tokens relate to each other over large spans.
Language
Modeling
logits logits logits logits logits …
Head U U U U U
Stacked
… … … … …
Transformer …
Blocks
x1 x2 x3 x4 x5 …
+ 1 + 2 + 3 + 4 + 5
Input
Encoding E E E E E
…
Fig. 8.1 sketches the transformer architecture. A transformer has three major
components. At the center are columns of transformer blocks. Each block is a
multilayer network (a multi-head attention layer, feedforward networks and layer
1 Although multi-head attention developed historically from the RNN attention mechanism (Chap-
ter 13), we’ll define attention from scratch here.
172 C HAPTER 8 • T RANSFORMERS
8.1 Attention
Recall from Chapter 5 that for word2vec and other static embeddings, the repre-
sentation of a word’s meaning is always the same vector irrespective of the context:
the word chicken, for example, is always represented by the same fixed vector. So
a static vector for the word it might somehow encode that this is a pronoun used
for animals and inanimate entities. But in context it has a much richer meaning.
Consider it in one of these two sentences:
(8.1) The chicken didn’t cross the road because it was too tired.
(8.2) The chicken didn’t cross the road because it was too wide.
In (8.1) it is the chicken (i.e., the reader knows that the chicken was tired), while
in (8.2) it is the road (and the reader knows that the road was wide).2 That is, if
we are to compute the meaning of this sentence, we’ll need the meaning of it to be
associated with the chicken in the first sentence and associated with the road in
the second one, sensitive to the context.
Furthermore, consider reading left to right like a causal language model, pro-
cessing the sentence up to the word it:
(8.3) The chicken didn’t cross the road because it
At this point we don’t yet know which thing it is going to end up referring to! So a
representation of it at this point might have aspects of both chicken and road as
the reader is trying to guess what happens next.
This fact that words have rich linguistic relationships with other words that may
be far away pervades language. Consider two more examples:
(8.4) The keys to the cabinet are on the table.
2 We say that in the first example it corefers with the chicken, and in the second it corefers with the
road; we’ll return to this in Chapter 23.
8.1 • ATTENTION 173
(8.5) I walked along the pond, and noticed one of the trees along the bank.
In (8.4), the phrase The keys is the subject of the sentence, and in English and many
languages, must agree in grammatical number with the verb are; in this case both are
plural. In English we can’t use a singular verb like is with a plural subject like keys
(we’ll discuss agreement more in Chapter 18). In (8.5), we know that bank refers
to the side of a pond or river and not a financial institution because of the context,
including words like pond. (We’ll discuss word senses more in Chapter 10.)
The point of all these examples is that these contextual words that help us com-
pute the meaning of words in context can be quite far away in the sentence or para-
graph. Transformers can build contextual representations of word meaning, contex-
contextual
embeddings tual embeddings, by integrating the meaning of these helpful contextual words. In a
transformer, layer by layer, we build up richer and richer contextualized representa-
tions of the meanings of input tokens. At each layer, we compute the representation
of a token i by combining information about i from the previous layer with infor-
mation about the neighboring tokens to produce a contextualized representation for
each word at each position.
Attention is the mechanism in the transformer that weighs and combines the
representations from appropriate other tokens in the context from layer k to build
the representation for tokens in layer k + 1.
because
didn’t
cross
tired
Layer k+1
road
The
the
was
too
it
self-attention distribution
chicken
because
didn’t
cross
tired
Layer k
road
The
the
was
too
it
Figure 8.2 The self-attention weight distribution α that is part of the computation of the
representation for the word it at layer k + 1. In computing the representation for it, we attend
differently to the various words at layer k, with darker shades indicating higher self-attention
values. Note that the transformer is attending highly to the columns corresponding to the
tokens chicken and road , a sensible result, since at the point where it occurs, it could plausibly
corefer with the chicken or the road, and hence we’d like the representation for it to draw on
the representation for these earlier words. Figure adapted from Uszkoreit (2017).
a1 a2 a3 a4 a5
x1 x2 x3 x4 x5
Figure 8.3 Information flow in causal self-attention. When processing each input xi , the
model attends to all the inputs up to, and including xi .
Each αi j is a scalar used for weighing the value of input x j when summing up
the inputs to compute ai . How shall we compute this α weighting? In attention we
weight each prior embedding proportionally to how similar it is to the current token
i. So the output of attention is a sum of the embeddings of prior tokens weighted
by their similarity with the current token embedding. We compute similarity scores
via dot product, which maps two vectors into a scalar value ranging from −∞ to
∞. The larger the score, the more similar the vectors that are being compared. We’ll
normalize these scores with a softmax to create the vector of weights αi j , j ≤ i.
the softmax weight will likely be highest for xi , since xi is very similar to itself,
resulting in a high dot product. But other context words may also be similar to i, and
the softmax will also assign some weight to those words. Then we use these weights
as the α values in Eq. 8.6 to compute the weighted sum that is our a3 .
The simplified attention in equations 8.6 – 8.8 demonstrates the attention-based
approach to computing ai : compare the xi to prior vectors, normalize those scores
into a probability distribution used to weight the sum of the prior vector. But now
we’re ready to remove the simplifications.
A single attention head using query, key, and value matrices Now that we’ve
attention head seen a simple intuition of attention, let’s introduce the actual attention head, the
head version of attention that’s used in transformers. (The word head is often used in
transformers to refer to specific structured layers). The attention head allows us to
distinctly represent three different roles that each input embedding plays during the
course of the attention process:
• As the current element being compared to the preceding inputs. We’ll refer to
query this role as a query.
• In its role as a preceding input that is being compared to the current element
key to determine a similarity weight. We’ll refer to this role as a key.
value • And finally, as a value of a preceding element that gets weighted and summed
up to compute the output for the current element.
To capture these three different roles, transformers introduce weight matrices
WQ , WK , and WV . These weights will project each input vector xi into a represen-
tation of its role as a query, key, or value:
qi = xi WQ ; ki = xi WK ; vi = xi WV (8.9)
Given these projections, when we are computing the similarity of the current ele-
ment xi with some prior element x j , we’ll use the dot product between the current
element’s query vector qi and the preceding element’s key vector k j . Furthermore,
the result of a dot product can be an arbitrarily large (positive or negative) value, and
exponentiating large values can lead to numerical issues and loss of gradients during
training. To avoid this, we scale the dot product by a factor related to the size of the
embeddings, via dividing by the square root of the dimensionality of the query and
key vectors (dk ). We thus replace the simplified Eq. 8.7 with Eq. 8.11. The ensuing
softmax calculation resulting in αi j remains the same, but the output calculation for
headi is now based on a weighted sum over the value vectors v (Eq. 8.13).
Here’s a final set of equations for computing self-attention for a single self-
attention output vector ai from a single input vector xi . This version of attention
computes ai by summing the values of the prior elements, each weighted by the
similarity of its key to the query from the current element:
qi = xi WQ ; k j = x j WK ; v j = x j WV (8.10)
qi · k j
score(xi , x j ) = √ (8.11)
dk
αi j = softmax(score(xi , x j )) ∀ j ≤ i (8.12)
X
headi = αi j v j (8.13)
j≤i
ai = headi WO (8.14)
176 C HAPTER 8 • T RANSFORMERS
8. Output of self-attention a3 [1 × d]
7. Reshape to [1 x d] WO [dv × d]
[1 × dv]
6. Sum the weighted
value vectors
×
×
4. Turn into 𝛼i,j weights via softmax
1. Generate k q v k q v k q v
key, query, value WK WQ WV WK WQ WV WK WQ WV
vectors
x1 x2 x3
[1 × d] [1 × d] [1 × d]
Figure 8.4 Calculating the value of a3 , the third element of a sequence using causal (left-
to-right) self-attention.
We illustrate this in Fig. 8.4 for the case of calculating the value of the third output
a3 in a sequence.
Note that we’ve also introduced one more matrix, WO , which is right-multiplied
by the attention head. This is necessary to reshape the output of the head. The input
to attention xi and the output from attention ai both have the same dimensionality
[1 × d]. We often call d the model dimensionality, and indeed as we’ll discuss in
Section 8.2 the output hi of each transformer block, as well as the intermediate vec-
tors inside the transformer block also have the same dimensionality [1 × d]. Having
everything be the same dimensionality makes the transformer very modular.
So let’s talk shapes. How do we get from [1 × d] at the input to [1 × d] at the
output? Let’s look at all the internal shapes. We’ll have a dimension dk for the
query and key vectors. The query vector and the key vector are both dimensionality
[1 × dk ], so we can take their dot product qi · k j to produce a scalar. We’ll have a
separate dimension dv for the value vectors. The transform matrix WQ has shape
[d × dk ], WK is [d × dk ], and WV is [d × dv ]. So the output of headi in equation
Eq. 8.13 is of shape [1 × dv ]. To get the desired output shape [1 × d] we’ll need to
reshape the head output, and so WO is of shape [dv × d]. In the original transformer
work (Vaswani et al., 2017), d was 512, dk and dv were both 64.
Multi-head Attention Equations 8.11-8.13 describe a single attention head. But
actually, transformers use multiple attention heads. The intuition is that each head
might be attending to the context for different purposes: heads might be special-
ized to represent different linguistic relationships between context elements and the
current token, or to look for particular kinds of patterns in the context.
multi-head So in multi-head attention we have A separate attention heads that reside in
attention
parallel layers at the same depth in a model, each with its own set of parameters that
allows the head to model different aspects of the relationships among inputs. Thus
8.2 • T RANSFORMER B LOCKS 177
each head i in a self-attention layer has its own set of query, key, and value matrices:
WQi , WKi , and WVi . These are used to project the inputs into separate query, key,
and value embeddings for each head.
When using multiple heads the model dimension d is still used for the input
and output, the query and key embeddings have dimensionality dk , and the value
embeddings are of dimensionality dv (again, in the original transformer paper dk =
dv = 64, A = 8, and d = 512). Thus for each head i, we have weight layers WQi of
shape [d × dk ], WKi of shape [d × dk ], and WVi of shape [d × dv ].
Below are the equations for attention augmented with multiple heads; Fig. 8.5
shows an intuition.
qci = xi WQc ; kcj = x j WKc ; vcj = x j WVc ; ∀ c 1 ≤ c ≤ A (8.15)
qci · kcj
scorec (xi , x j ) = √ (8.16)
dk
αicj = softmax(scorec (xi , x j )) ∀ j ≤ i (8.17)
X
headci = αicj vcj (8.18)
j≤i
ai
[1 x d]
Project down to d WO [Adv x d]
… [1 x Adv ]
Concatenate Outputs
[1 x dv ] [1 x dv ]
Each head Head 1 Head 2 Head 8
attends differently …
WK1 WV1 WQ1 WK2 WV2 WQ2 WK8 WV8 WQ8
to context
hi-1 hi hi+1
Residual
Stream
Feedforward
Layer Norm
… …
+
MultiHead
Attention
Layer Norm
xi-1 xi xi+1
Figure 8.6 The architecture of a transformer block showing the residual stream. This
figure shows the prenorm version of the architecture, in which the layer norms happen before
the attention and feedforward layers rather than after.
tention layer that we have seen, and the feedforward layer that we will introduce.
Before the attention and feedforward layer is a computation called the layer norm.
Thus the initial vector is passed through a layer norm and attention layer, and
the result is added back into the stream, in this case to the original input vector
xi . And then this summed vector is again passed through another layer norm and a
feedforward layer, and the output of those is added back into the residual, and we’ll
use hi to refer to the resulting output of the transformer block for token i. (In earlier
descriptions the residual stream was often described using a different metaphor as
residual connections that add the input of a component to its output, but the residual
stream is a more perspicuous way of visualizing the transformer.)
8.2 • T RANSFORMER B LOCKS 179
We’ve already seen the attention layer, so let’s now introduce the feedforward
and layer norm computations in the context of processing a single input xi at token
position i.
Layer Norm At two stages in the transformer block we normalize the vector (Ba
layer norm et al., 2016). This process, called layer norm (short for layer normalization), is one
of many forms of normalization that can be used to improve training performance
in deep neural networks by keeping the values of a hidden layer in a range that
facilitates gradient-based training.
Layer norm is a variation of the z-score from statistics, applied to a single vec-
tor in a hidden layer. That is, the term layer norm is a bit confusing; layer norm
is not applied to an entire transformer layer, but just to the embedding vector of a
single token. Thus the input to layer norm is a single vector of dimensionality d
and the output is that vector normalized, again of dimensionality d. The first step in
layer normalization is to calculate the mean, µ, and standard deviation, σ , over the
elements of the vector to be normalized. Given an embedding vector x of dimen-
sionality d, these values are calculated as follows.
d
1X
µ = xi (8.22)
d
i=1
v
u d
u1 X
σ = t (xi − µ)2 (8.23)
d
i=1
Given these values, the vector components are normalized by subtracting the mean
from each and dividing by the standard deviation. The result of this computation is
a new vector with zero mean and a standard deviation of one.
(x − µ)
x̂ = (8.24)
σ
(x − µ)
LayerNorm(x) = γ +β (8.25)
σ
Putting it all together The function computed by a transformer block can be ex-
pressed by breaking it down with one equation for each component computation,
using t (of shape [1 × d]) to stand for transformer and superscripts to demarcate
180 C HAPTER 8 • T RANSFORMERS
Notice that the only component that takes as input information from other tokens
(other residual streams) is multi-head attention, which (as we see from Eq. 8.27)
looks at all the neighboring tokens in the context. The output from attention, how-
ever, is then added into this token’s embedding stream. In fact, Elhage et al. (2021)
show that we can view attention heads as literally moving information from the
residual stream of a neighboring token into the current stream. The high-dimensional
embedding space at each position thus contains information about the current to-
ken and about neighboring tokens, albeit in different subspaces of the vector space.
Fig. 8.7 shows a visualization of this movement.
Token A Token B
residual residual
stream stream
Figure 8.7 An attention head can move information from token A’s residual stream into
token B’s residual stream.
Crucially, the input and output dimensions of transformer blocks are matched so
they can be stacked. Each token vector xi at the input to the block has dimensionality
d, and the output hi also has dimensionality d. Transformers for large language
models stack many of these blocks, from 12 layers (used for the T5 or GPT-3-small
language models) to 96 layers (used for GPT-3 large), to even more for more recent
models. We’ll come back to this issue of stacking in a bit.
Equation 8.26 and following are just the equation for a single transformer block,
but the residual stream metaphor goes through all the transformer layers, from the
first transformer blocks to the 12th, in a 12-layer transformer. At the earlier trans-
former blocks, the residual stream is representing the current token. At the highest
transformer blocks, the residual stream is usually representing the following token,
since at the very end it’s being trained to predict the next token.
Once we stack many blocks, there is one more requirement: at the very end of
the last (highest) transformer block, there is a single extra layer norm that is run on
the last hi of each token stream (just below the language model head layer that we
will define soon). 3
3 Note that we are using the most common current transformer architecture, which is called the prenorm
8.3 • PARALLELIZING COMPUTATION USING A SINGLE MATRIX X 181
Given these matrices we can compute all the requisite query-key comparisons simul-
taneously by multiplying Q and K| in a single matrix multiplication. The product is
of shape N × N, visualized in Fig. 8.8.
Figure 8.8 The N × N QK| matrix showing how it computes all qi · k j comparisons in a
single matrix multiple.
Once we have this QK| matrix, we can very efficiently scale these scores, take
the softmax, and then multiply the result by V resulting in a matrix of shape N × d:
a vector embedding representation for each token in the input. We’ve reduced the
entire self-attention step for an entire sequence of N tokens for one head to the
architecture. The original definition of the transformer in Vaswani et al. (2017) used an alternative archi-
tecture called the postnorm transformer in which the layer norm happens after the attention and FFN
layers; it turns out moving the layer norm beforehand works better, but does require this one extra layer
at the end.
182 C HAPTER 8 • T RANSFORMERS
following computation:
QK|
head = softmax mask √ V (8.33)
dk
A = head WO (8.34)
Masking out the future You may have noticed that we introduced a mask function
in Eq. 8.34 above. This is because the self-attention computation as we’ve described
it has a problem: the calculation of QK| results in a score for each query value to
every key value, including those that follow the query. This is inappropriate in the
setting of language modeling: guessing the next word is pretty simple if you already
know it! To fix this, the elements in the upper-triangular portion of the matrix are set
to −∞, which the softmax will turn to zero, thus eliminating any knowledge of words
that follow in the sequence. This is done in practice by adding a mask matrix M in
which Mi j = −∞ ∀ j > i (i.e. for the upper-triangular portion) and Mi j = 0 otherwise.
Fig. 8.9 shows the resulting masked QK| matrix. (we’ll see in Chapter 10 how to
make use of words in the future for tasks that need it).
q1•k1 −∞ −∞ −∞
q2•k1 q2•k2 −∞ −∞
N
q3•k1 q3•k2 q3•k3 −∞
Figure 8.9 The N × N QK| matrix showing the qi · k j values, with the upper-triangle por-
tion of the comparisons matrix zeroed out (set to −∞, which the softmax will turn to zero).
Fig. 8.10 shows a schematic of all the computations for a single attention head
parallelized in matrix form.
Fig. 8.8 and Fig. 8.9 also make it clear that attention is quadratic in the length
of the input, since at each layer we need to compute dot products between each pair
of tokens in the input. This makes it expensive to compute attention over very long
documents (like entire novels). Nonetheless modern large language models manage
to use quite long contexts of thousands or tens of thousands of tokens.
X Q X K X V
Input
WQ Query Input WK Key Input WV Value
Token 1 Token 1 Token 1 Token 1 Token 1
Token 1
Input Input Key Input Value
Query
Token 2 Token 2 Token 2 Token 2
Input x =
Token 2
x = Key
x =
Token 2
Query Input Input Value
Token 3 Token 3 Token 3 Token 3 Token 3
Token 3
Input Input Key Input Value
Query
Token 4 Token 4 Token 4 Token 4
Token 4 d x dk d x dv Token 4
d x dk
Nxd N x dk Nxd N x dk N x dv
Nxd
q1
x = −∞ −∞ −∞ v1 a1
k1
k2
k3
k4
N x dk NxN NxN N x dv N x dv
Figure 8.10 Schematic of the attention computation for a single attention head in parallel. The first row shows
the computation of the Q, K, and V matrices. The second row shows the computation of QKT , the masking
(the softmax computation and the normalizing by dimensionality are not shown) and then the weighted sum of
the value vectors to get the final attention vectors.
Putting it all together with the parallel input matrix X The function computed
in parallel by an entire layer of N transformer blocks—each block over one of the N
input tokens—can be expressed as:
O = X + MultiHeadAttention(LayerNorm(X)) (8.38)
H = O + FFN(LayerNorm(O)) (8.39)
Note that in Eq. 8.38 we are using X to mean the input to the layer, wherever it
comes from. For the first layer, as we will see in the next section, that input is the
initial word + positional embedding vectors that we have been describing by X. But
for subsequent layers k, the input is the output from the previous layer Hk−1 . We
can also break down the computation performed in a transformer layer, showing one
equation for each component computation. We’ll use T (of shape [N × d]) to stand
for transformer and superscripts to demarcate each computation inside the block,
and again use X to mean the input to the block from the previous layer or the initial
184 C HAPTER 8 • T RANSFORMERS
embedding:
T1 = LayerNorm(X) (8.40)
T 2
= MultiHeadAttention(T )
1
(8.41)
T3 = T2 + X (8.42)
T4 = LayerNorm(T3 ) (8.43)
T 5
= FFN(T )4
(8.44)
5 3
H = T +T (8.45)
Here when we use a notation like FFN(T3 ) we mean that the same FFN is applied
in parallel to each of the N embedding vectors in the window. Similarly, each of the
N tokens is normed in parallel in the LayerNorm. Crucially, the input and output
dimensions of transformer blocks are matched so they can be stacked. Since each
token xi at the input to the block is represented by an embedding of dimensionality
[1 × d], that means the input X and output H are both of shape [N × d].
5 |V| 5 d
1 0000100…0000 ✕ E = 1
|V|
Figure 8.11 Selecting the embedding vector for word V5 by multiplying the embedding
matrix E with a one-hot vector with a 1 in index 5.
We can extend this idea to represent the entire token sequence as a matrix of one-
hot vectors, one for each of the N positions in the transformer’s context window, as
shown in Fig. 8.12.
d
|V| d
0000100…0000
0000000…0010
1000000…0000 ✕ E =
…
N 0000100…0000
N
| V|
Figure 8.12 Selecting the embedding matrix for the input sequence of token ids W by mul-
tiplying a one-hot matrix corresponding to W by the embedding matrix E.
Transformer Block
X = Composite
Embeddings
(word + position)
+
+
Word
Janet
back
will
the
bill
Embeddings
Position
1
Embeddings
Janet will back the bill
Figure 8.13 A simple way to model position: add an embedding of the absolute position to
the token embedding to produce a new embedding of the same dimensionality.
186 C HAPTER 8 • T RANSFORMERS
Language models give us the ability to assign such a conditional probability to every
possible next word, giving us a distribution over the entire vocabulary. The n-gram
language models of Chapter 3 compute the probability of a word given counts of
its occurrence with the n − 1 prior words. The context is thus of size n − 1. For
transformer language models, the context is the size of the transformer’s context
window, which can be quite large, like 32K tokens for large models (and much larger
contexts of millions of words are possible with special long-context architectures).
The job of the language modeling head is to take the output of the final trans-
former layer from the last token N and use it to predict the upcoming word at posi-
tion N + 1. Fig. 8.14 shows how to accomplish this task, taking the output of the last
token at the last layer (the d-dimensional output embedding of shape [1 × d]) and
producing a probability distribution over words (from which we will choose one to
generate).
The first module in Fig. 8.14 is a linear layer, whose job is to project from the
output hLN , which represents the output token embedding at position N from the final
8.5 • T HE L ANGUAGE M ODELING H EAD 187
Figure 8.14 The language modeling head: the circuit at the top of a transformer that maps from the output
embedding for token N from the last transformer layer (hLN ) to a probability distribution over words in the
vocabulary V .
logit block L, (hence of shape [1 × d]) to the logit vector, or score vector, that will have a
single score for each of the |V | possible words in the vocabulary V . The logit vector
u is thus of dimensionality [1 × |V |].
This linear layer can be learned, but more commonly we tie this matrix to (the
weight tying transpose of) the embedding matrix E. Recall that in weight tying, we use the
same weights for two different matrices in the model. Thus at the input stage of the
transformer the embedding matrix (of shape [|V | × d]) is used to map from a one-hot
vector over the vocabulary (of shape [1 × |V |]) to an embedding (of shape [1 × d]).
And then in the language model head, ET , the transpose of the embedding matrix (of
shape [d × |V |]) is used to map back from an embedding (shape [1 × d]) to a vector
over the vocabulary (shape [1×|V |]). In the learning process, E will be optimized to
be good at doing both of these mappings. We therefore sometimes call the transpose
unembedding ET the unembedding layer because it is performing this reverse mapping.
A softmax layer turns the logits u into the probabilities y over the vocabulary.
u = hLN ET (8.46)
y = softmax(u) (8.47)
hLi
feedforward
layer norm
Layer L
attention
layer norm
hL-1i = xLi
…
h2i = x3i
feedforward
layer norm
Layer 2
attention
layer norm
h1i = x2i
feedforward
layer norm
Layer 1
attention
layer norm
x1i
+ i
Input
Encoding E
Input token wi
causal language model was defined by using only the decoder part of this original
architecture).
8.7 Training
We described the training process for language models in the prior chapter. Re-
call that large language models are trained with cross-entropy loss, also called the
negative log likelihood loss. At time t the cross-entropy loss is the negative log prob-
ability the model assigns to the next word in the training sequence, − log p(wt+1 ).
Fig. 8.16 illustrates the general training approach. At each step, given all the
preceding words, the final transformer layer produces an output distribution over the
entire vocabulary. During training, the probability assigned to the correct word by
the model is used to calculate the cross-entropy loss for each item in the sequence.
The loss for a training sequence is the average cross-entropy loss over the entire
sequence. The weights in the network are adjusted to minimize the average CE loss
over the training sequence via gradient descent.
190 C HAPTER 8 • T RANSFORMERS
log ythanks
<latexit sha1_base64="q3ZgXDyG7qtkT7t8hT47RdlwYG4=">AAAB+XicbVDLSsNAFJ3UV62vWHe6GVsEN5bERXUlBUVcVrAPaEqYTCft0MlMmJkIIQT8AT/CTRE3Cv6Ev+DfmLTdtPXAwOGcM9x7jxcyqrRl/RqFtfWNza3idmlnd2//wDwst5WIJCYtLJiQXQ8pwignLU01I91QEhR4jHS88W3ud56JVFTwJx2HpB+gIac+xUhnkmseXzhMDGHsJk6A9EgGiR4hPlZpWnLNqlWzpoCrxJ6TauP0tXw3qdw0XfPHGQgcBYRrzJBSPdsKdT9BUlPMSFpyIkVChMdoSJLp5ik8y6QB9IXMHtdwqi7kUKBUHHhZMl9PLXu5+J/Xi7R/3U8oDyNNOJ4N8iMGtYB5DXBAJcGaxRlBWNJsQ4hHSCKss7Ly0+3lQ1dJ+7Jm12v1x6yDezBDEZyACjgHNrgCDfAAmqAFMHgBE/AJvozEeDPejY9ZtGDM/xyBBRjff79pldo=</latexit>
Loss
Language
Modeling
logits logits logits logits logits …
Head U U U U U
Stacked
Transformer
… … … … … …
Blocks
x1 x2 x3 x4 x5 …
+ 1 + 2 + 3 + 4 + 5
Input
Encoding E E E E E
…
With transformers, each training item can be processed in parallel since the out-
put for each element in the sequence is computed separately.
Large models are generally trained by filling the full context window (for exam-
ple 4096 tokens for GPT4 or 8192 for Llama 3) with text. If documents are shorter
than this, multiple documents are packed into the window with a special end-of-text
token between them. The batch size for gradient descent is usually quite large (the
largest GPT-3 model uses a batch size of 3.2 million tokens).
8.8.2 KV Cache
We saw in Fig. 8.10 and in Eq. 8.34 (repeated below) how the attention vector can
be very efficiently computed in parallel for training, via two matrix multiplications:
QK|
A = softmax √ V (8.53)
dk
Q QKT V A
KT v1
x = x v2
k1
k2
k3
k4
=
v3
dk x N v4
q4 q4•k1 q4•k2 q4•k3 q4•k4 a4
1 x dk 1xN N x dv 1 x dv
Figure 8.17 Parts of the attention computation (extracted from Fig. 8.10) showing, in black,
the vectors that can be stored in the cache rather than recomputed when computing the atten-
tion score for the 4th token.
the computation that takes place for a single new token, showing which values we
can take from the cache rather than recompute.
h = xW + xAB (8.54)
d
h 1
d
× r B
Pretrained
Weights
N N A
W
d r
x 1
d
Figure 8.18 The intuition of LoRA. We freeze W to its pretrained values, and instead fine-
tune by training a pair of matrices A and B, updating those instead of W, and just sum W and
the updated AB.
That means it doesn’t add any time during inference. And it also means it’s possible
to build LoRA modules for different domains and just swap them in and out by
adding them in or subtracting them from W.
In its original version LoRA was applied just to the matrices in the attention
computation (the WQ , WK , WV , and WO layers). Many variants of LoRA exist.
guage models do from their prompts. In-context learning means language models
learning to do new tasks, better predict tokens, or generally reduce their loss dur-
ing the forward-pass at inference-time, without any gradient-based updates to the
model’s parameters.
How does in-context learning work? While we don’t know for sure, there are
induction heads some intriguing ideas. One hypothesis is based on the idea of induction heads
(Elhage et al., 2021; Olsson et al., 2022). Induction heads are the name for a circuit,
which is a kind of abstract component of a network. The induction head circuit
is part of the attention computation in transformers, discovered by looking at mini
language models with only 1-2 attention heads.
The function of the induction head is to predict repeated sequences. For example
if it sees the pattern AB...A in an input sequence, it predicts that B will follow,
instantiating the pattern completion rule AB...A→ B. It does this by having a prefix
matching component of the attention computation that, when looking at the current
token A, searches back over the context to find a prior instance of A. If it finds one,
the induction head has a copying mechanism that “copies” the token B that followed
the earlier A, by increasing the probability the B will occur next. Fig. 8.19 shows an
example.
Figure
Figure 1:8.19 An induction
In the sequence head
“...vintage looking
cars ... vintage”,atanvintage uses
induction head the prefix
identifies matching
the initial mechanism
occurrence of “vintage”,to
find a prior
attends to theinstance
subsequentofword “cars” forand
vintage, prefixthe copying
matching, and mechanism
predicts “cars” to predict
as the next wordthatthrough will
cars the occur
copying
mechanism.
again. Figure from Crosbie and Shutova (2022).
determines each head’s independent output for the 4.2 Identifying Induction Heads
Olsson et al. (2022) propose that a generalized fuzzy version of this pattern com-
current token. To identify
pletion rule, implementing a rule like A*B*...A→
Leveraging this decomposition, Elhage et al. sure the ability B,induction
where heads
A* ≈within
A and models,
B* ≈ weB mea-
(by
of all attention heads to perform
we mean
≈(2021) theyathey
discovered arebehaviour
distinct semantically
in certainsimilar in some way), might be responsible
prefix matching on random input sequences. We 4
forattention
in-context learning.
heads, which Suggestive
they named induction evidence for the
heads. follow their hypothesis
task-agnostic comes
approach from Cros-
to computing pre-
ablating This
bie andbehaviour
Shutova emerges when who
(2022), these heads
showprocess
that ablating induction
fix matching headsbycauses
scores outlined Bansal etin-context
al. (2023).
sequences of the form "[A] [B] ... [A] → ". In Weis argue that focusing solely on term
prefix matching
learning performance to decrease. Ablation originally a medical
these heads, the QK circuit directs attention to- scores is sufficient for our analysis, as high pre-
meaning
the removal
wards [B], whichofappears
something. Wetheuse
directly after it in NLP
previous interpretability studies as a tool for
fix matching cores specifically indicate induction
testing
occurrencecausal
of theeffects; if we
current token [A].knock out a hypothesized
This behaviour heads, while less cause,
relevantwe would
heads tend toexpect
show high the
is termed
effect prefix matching.
to disappear. The OV
Crosbie andcircuit subse- (2022)
Shutova copyingablate induction
capabilities (Bansalheads by first
et al., 2023). We find-
gen-
quently increases the output logit of the [B] token, erate a sequence of 50 random tokens, excluding
ing attention heads that perform as induction heads on random input sequences, and
termed copying. An overview of this mechanism is the 4% most common and least common tokens.
then
shown zeroing
in Figureout
1. the output of these heads by setting certain terms of the output ma-
This sequence is repeated four times to form the
trix WO to zero. Indeed they find that ablated
inputmodels are The
to the model. much worse
prefix at in-context
matching score is cal-
4 Methods
learning: they have much worse performance at learning
culated from
by averaging the demonstrations
attention values fromin the
each
prompts. token to the tokens that directly followed the same
4.1 Models
token in earlier repeats. The final prefix matching
We utilise two recently developed open-source scores are averaged over five random sequences.
8.9.2 Logit
models, namely Lens 2 and InternLM2-20B
Llama-3-8B The prefix matching scores for Llama-3-8B are
(Cai et al., 2024), both of which are based on the shown in Figure 2. For IntermLM2-20B, we refer
logit lens original Llama
Another useful(Touvron et al., 2023a)
interpretability tool, the logit
architec- lens 8(Nostalgebraist,
to Figure in Appendix A.1. Both 2020),
modelsoffers
exhibit a
ture. These models feature grouped-query atten- heads with notably high prefix matching scores,
way to visualize what the internal layers of the transformer might be representing.
tion mechanisms (Ainslie et al., 2023) to enhance distributed across various layers. In the Llama-3-
The idea
efficiency. is that we
Llama-3-8B, take any
comprises vector
32 layers, eachfrom8Bany layer
model, ~3% of theheads
of the transformer
have a prefixand, pre-
matching
tending that it is
with 32 attention theand
heads prefinal
it uses aembedding,
query group simply
score of multiply by the unembedding
it indicating
0.3 or higher, a degree of spe-
layer
size ofto4 get
attention heads.
logits, andIt compute
has shown a superior
softmaxcialisation
to see the distribution
in prefix matching, andover
somewords that
heads have
performance compared to its predecessors, even high scores of up to 0.98.
that vector might be
the larger Llama-2 models.
representing. This can be a useful window into the internal
representations
InternLM2-20B,offeaturing
the model. Since
48 layers with the
48 at-network wasn’t
4.3 Head trained to make the internal
Ablations
tention heads each, uses a query group size of 6 To investigate the significance of induction heads
attention heads. We selected InternLM2-20B for for a specific ICL task, we conduct zero-ablations
its exemplary performance on the Needle-in-the- of 1% and 3% of the heads with the highest prefix
Haystack3 task, which assesses LLMs’ ability to matching scores. This ablation process involves
retrieve a single critical piece of information em- masking the corresponding partition of the output
bedded within a lengthy text. This mirrors the matrix, denoted as Woh in Eq. 1, by setting it to
functionality of induction heads, which scan the zero. This effectively renders the heads inactive
8.10 • S UMMARY 195
representations function in this way, the logit lens doesn’t always work perfectly, but
this can still be a useful trick to help us visualize the internal layers of a transformer.
8.10 Summary
This chapter has introduced the transformer and its components for the language
modeling task introduced in the previous chapter. Here’s a summary of the main
points that we covered:
• Transformers are non-recurrent networks based on multi-head attention, a
kind of self-attention. A multi-head attention computation takes an input
vector xi and maps it to an output ai by adding in vectors from prior tokens,
weighted by how relevant they are for the processing of the current word.
• A transformer block consists of a residual stream in which the input from
the prior layer is passed up to the next layer, with the output of different com-
ponents added to it. These components include a multi-head attention layer
followed by a feedforward layer, each preceded by layer normalizations.
Transformer blocks are stacked to make deeper and more powerful networks.
• The input to a transformer is computed by adding an embedding (computed
with an embedding matrix) to a positional encoding that represents the se-
quential position of the token in the window.
• Language models can be built out of stacks of transformer blocks, with a
language model head at the top, which applies an unembedding matrix to
the output H of the top layer to generate the logits, which are then passed
through a softmax to generate word probabilities.
• Transformer-based language models have a wide context window (200K to-
kens or even more for very large models with special mechanisms) allowing
them to draw on enormous amounts of context to predict upcoming words.
• There are various computational tricks for making large language models
more efficient, such as the KV cache and parameter-efficient finetuning.
Historical Notes
The transformer (Vaswani et al., 2017) was developed drawing on two lines of prior
research: self-attention and memory networks.
Encoder-decoder attention, the idea of using a soft weighting over the encodings
of input words to inform a generative decoder (see Chapter 12) was developed by
Graves (2013) in the context of handwriting generation, and Bahdanau et al. (2015)
for MT. This idea was extended to self-attention by dropping the need for separate
encoding and decoding sequences and instead seeing attention as a way of weighting
the tokens in collecting information passed from lower layers to higher layers (Ling
et al., 2015; Cheng et al., 2016; Liu et al., 2016).
Other aspects of the transformer, including the terminology of key, query, and
value, came from memory networks, a mechanism for adding an external read-
write memory to networks, by using an embedding of a query to match keys rep-
196 C HAPTER 8 • T RANSFORMERS
In the previous two chapters we introduced the transformer and saw how to pre-
train a transformer language model as a causal or left-to-right language model. In
this chapter we’ll introduce a second paradigm for pretrained language models, the
BERT
masked
bidirectional transformer encoder, and the most widely-used version, the BERT
language model (Devlin et al., 2019). This model is trained via masked language modeling,
modeling
where instead of predicting the following word, we mask a word in the middle and
ask the model to guess the word given the words on both sides. This method thus
allows the model to see both the right and left context.
finetuning We also introduced finetuning in the prior chapter. Here we describe a new
kind of finetuning, in which we take the transformer network learned by these pre-
trained models, add a neural net classifier after the top layer of the network, and train
it on some additional labeled data to perform some downstream task like named
entity tagging or natural language inference. As before, the intuition is that the
pretraining phase learns a language model that instantiates rich representations of
word meaning, that thus enables the model to more easily learn (‘be finetuned to’)
the requirements of a downstream language understanding task. This aspect of the
transfer
learning pretrain-finetune paradigm is an instance of what is called transfer learning in ma-
chine learning: the method of acquiring knowledge from one task or domain, and
then applying it (transferring it) to solve a new task.
The second idea that we introduce in this chapter is the idea of contextual em-
beddings: representations for words in context. The methods of Chapter 5 like
word2vec or GloVe learned a single vector embedding for each unique word w in
the vocabulary. By contrast, with contextual embeddings, such as those learned by
masked language models like BERT, each word w will be represented by a different
vector each time it appears in a different context. While the causal language models
of Chapter 8 also use contextual embeddings, the embeddings created by masked
language models seem to function particularly well as representations.
cially true for sequence labeling tasks in which we want to tag each token with a
label, such as the named entity tagging task we’ll introduce in Section 9.5, or tasks
like part-of-speech tagging or parsing that come up in later chapters.
The bidirectional encoders that we introduce here are a different kind of beast
than causal models. The causal models of Chapter 8 are generative models, de-
signed to easily generate the next token in a sequence. But the focus of bidirec-
tional encoders is instead on computing contextualized representations of the input
tokens. Bidirectional encoders use self-attention to map sequences of input embed-
dings (x1 , ..., xn ) to sequences of output embeddings of the same length (h1 , ..., hn ),
where the output vectors have been contextualized using information from the en-
tire input sequence. These output embeddings are contextualized representations of
each input token that are useful across a range of applications where we need to do
a classification or a decision based on the token in context.
Remember that we said the models of Chapter 8 are sometimes called decoder-
only, because they correspond to the decoder part of the encoder-decoder model we
will introduce in Chapter 12. By contrast, the masked language models of this chap-
ter are sometimes called encoder-only, because they produce an encoding for each
input token but generally aren’t used to produce running text by decoding/sampling.
That’s an important point: masked language models are not used for generation.
They are generally instead used for interpretative tasks.
a1 a2 a3 a4 a5 a1 a2 a3 a4 a5
attention attention attention attention attention attention attention attention attention attention
x1 x2 x3 x4 x5 x1 x2 x3 x4 x5
Figure 9.1 (a) The causal transformer from Chapter 8, highlighting the attention computation at token 3. The
attention value at each token is computed using only information seen earlier in the context. (b) Information
flow in a bidirectional attention model. In processing each token, the model attends to all inputs, both before
and after the current one. So attention for token 3 can draw on information from following tokens.
N N
q3•k1 q3•k2 q3•k3 −∞ q3•k1 q3•k2 q3•k3 q3•k4
N N
(a) (b)
Figure 9.2 The N × N QK|matrix showing the qi · k j values. (a) shows the upper-triangle
portion of the comparisons matrix zeroed out (set to −∞, which the softmax will turn to zero),
while (b) shows the unmasked version.
Fig. 9.2 shows the masked version of QK| and the unmasked version. For bidi-
rectional attention, we use the unmasked version of Fig. 9.2b. Thus the attention
computation for bidirectional attention is exactly the same as Eq. 9.1 but with the
mask removed:
QK|
head = softmax √ V (9.2)
dk
Otherwise, the attention computation is identical to what we saw in Chapter 8, as is
the transformer block architecture (the feedforward layer, layer norm, and so on). As
in Chapter 8, the input is also a series of subword tokens, usually computed by one of
the 3 popular tokenization algorithms (including the BPE algorithm that we already
saw in Chapter 2 and two others, the WordPiece algorithm and the SentencePiece
Unigram LM algorithm). That means every input sentence first has to be tokenized,
and all further processing takes place on subword tokens rather than words. This will
require, as we’ll see in the third part of the textbook, that for some NLP tasks that
require notions of words (like parsing) we will occasionally need to map subwords
back to words.
To make this more concrete, the original English-only bidirectional transformer
encoder model, BERT (Devlin et al., 2019), consisted of the following:
• An English-only subword vocabulary consisting of 30,000 tokens generated
using the WordPiece algorithm (Schuster and Nakajima, 2012).
• Input context window N=512 tokens, and model dimensionality d=768
• So X, the input to the model, is of shape [N × d] = [512 × 768].
• L=12 layers of transformer blocks, each with A=12 (bidirectional) multihead
attention layers.
• The resulting model has about 100M parameters.
The larger multilingual XLM-RoBERTa model, trained on 100 languages, has
• A multilingual subword vocabulary with 250,000 tokens generated using the
SentencePiece Unigram LM algorithm (Kudo and Richardson, 2018b).
200 C HAPTER 9 • M ASKED L ANGUAGE M ODELS
• Input context window N=512 tokens, and model dimensionality d=1024, hence
X, the input to the model, is of shape [N × d] = [512 × 1024].
• L=24 layers of transformer blocks, with A=16 multihead attention layers each
• The resulting model has about 550M parameters.
Note that 550M parameters is relatively small as large language models go
(Llama 3 has 405B parameters, so is 3 orders of magnitude bigger). Indeed, masked
language models tend to be much smaller than causal language models.
• 10% of the time: The token is replaced with another token, randomly sampled
from the vocabulary based on token unigram probabilities. e.g. lunch was
delicious → lunch was gasp.
• 10% of the time: the token is left unchanged. e.g. lunch was delicious
→ lunch was delicious.
We then train the model to guess the correct token for the manipulated tokens. Why
the three possible manipulations? Adding the [MASK] token creates a mismatch
between pretraining and downstream finetuning or inference, since when we employ
the MLM model to perform a downstream task, we don’t use any [MASK] tokens. If
we just replaced tokens with the [MASK], the model might only predict tokens when
it sees a [MASK], but we want the model to try to always predict the input token.
To train the model to make the prediction, the original input sequence is to-
kenized using a subword model and tokens are sampled to be manipulated. Word
embeddings for all of the tokens in the input are retrieved from the E embedding ma-
trix and combined with positional embeddings to form the input to the transformer,
passed through the stack of bidirectional transformer blocks, and then the language
modeling head. The MLM training objective is to predict the original inputs for
each of the masked tokens and the cross-entropy loss from these predictions drives
the training process for all the parameters in the model. That is, all of the input
tokens play a role in the self-attention process, but only the sampled tokens are used
for learning.
CE Loss
Token + + + + + + + + +
Positional
Embeddings p1 p2 p3 p4 p5 p6 p7 p8
Fig. 9.3 illustrates this approach with a simple example. Here, long, thanks and
the have been sampled from the training sequence, with the first two masked and the
replaced with the randomly sampled token apricot. The resulting embeddings are
passed through a stack of bidirectional transformer blocks. Recall from Section 8.5
in Chapter 8 that to produce a probability distribution over the vocabulary for each
of the masked tokens, the language modeling head takes the output vector hLi from
the final transformer layer L for each masked token i, multiplies it by the unembed-
ding layer ET to produce the logits u, and then uses softmax to turn the logits into
202 C HAPTER 9 • M ASKED L ANGUAGE M ODELS
ui = hLi ET (9.3)
yi = softmax(ui ) (9.4)
With a predicted probability distribution for each masked item, we can use cross-
entropy to compute the loss for each masked item—the negative log probability
assigned to the actual masked word, as shown in Fig. 9.3. More formally, for a
given vector of input tokens in a sentence or batch x, let the set of tokens that are
masked be M, the version of that sentence with some tokens replaced by masks be
xmask , and the sequence of output vectors be h. For a given input token xi , such as
the word long in Fig. 9.3, the loss is the probability of the correct word long, given
xmask (as summarized in the single output vector hLi ):
The gradients that form the basis for the weight updates are based on the average
loss over the sampled learning items from a single training sequence (or batch of
sequences).
1 X
LMLM = − log P(xi |hLi )
|M|
i∈M
Note that only the tokens in M play a role in learning; the other words play no role
in the loss function, so in that sense BERT and its descendents are inefficient; only
15% of the input samples in the training data are actually used for training weights.1
yi = softmax(hLCLS WNSP )
Cross entropy is used to compute the NSP loss for each sentence pair presented
to the model. Fig. 9.4 illustrates the overall NSP training setup. In BERT, the NSP
loss was used in conjunction with the MLM training objective to form final loss.
CE Loss
NSP
Head
hCLS
Token +
Segment + + + + + + + + + + + + + + + + + + +
Positional s1 p1 s1 p2 s1 p3 s1 p4 s1 p5 s2 p6 s2 p7 s2 p8 s2 p9
Embeddings
[CLS] Cancel my flight [SEP] And the hotel [SEP]
a whole. Approximately 40 passes (epochs) over the training data was required for
the model to converge.
Some models, like the RoBERTa model, drop the next sentence prediction ob-
jective, and therefore change the training regime a bit. Instead of sampling pairs of
sentence, the input is simply a series of contiguous sentences, still beginning with
the special [CLS] token. If the document runs out before 512 tokens are reached, an
extra separator token is added, and sentences from the next document are packed in,
until we reach a total of 512 tokens. Usually large batch sizes are used, between 8K
and 32K tokens.
Multilingual models have an additional decision to make: what data to use to
build the vocabulary? Recall that all language models use subword tokenization
(BPE or SentencePiece Unigram LM are the two most common algorithms). What
text should be used to learn this multilingual tokenization, given that it’s easier to
get much more text in some languages than others? One option would be to cre-
ate this vocabulary-learning dataset by sampling sentences from our training data
(perhaps web text from Common Crawl), randomly. In that case we will choose a
lot of sentences from languages with lots of web representation like English, and
the tokens will be biased toward rare English tokens instead of creating frequent
tokens from languages with less data. Instead, it is common to divide the training
data into subcorpora of N different languages, compute the number of sentences ni
of each language i, and readjust these probabilities so as to upweight the probability
of less-represented languages (Lample and Conneau, 2019). The new probability of
selecting a sentence from each of the N languages (whose prior frequency is ni ) is
{qi }i=1...N , where:
pα ni
qi = PN i with pi = PN (9.5)
j=1 p j k=1 nk
α
Recall from Eq. 5.19 in Chapter 5 that an α value between 0 and 1 will give higher
weight to lower probability samples. Conneau et al. (2020) show that α = 0.3 works
well to give rare languages more inclusion in the tokenization, resulting in better
multilingual performance overall.
The result of this pretraining process consists of both learned word embeddings,
as well as all the parameters of the bidirectional encoder that are used to produce
contextual embeddings for novel inputs.
For many purposes, a pretrained multilingual model is more practical than a
monolingual model, since it avoids the need to build many (a hundred!) separate
monolingual models. And multilingual models can improve performance on low-
resourced languages by leveraging linguistic information from a similar language in
the training data that happens to have more resources. Nonetheless, when the num-
ber of languages grows very large, multilingual models exhibit what has been called
the curse of multilinguality (Conneau et al., 2020): the performance on each lan-
guage degrades compared to a model training on fewer languages. Another problem
with multilingual models is that they ‘have an accent’: grammatical structures in
higher-resource languages (often English) bleed into lower-resource languages; the
vast amount of English language in training makes the model’s representations for
low-resource languages slightly more English-like (Papadimitriou et al., 2023).
9.3 • C ONTEXTUAL E MBEDDINGS 205
+ i + i + i + i + i + i + i
E E E E E E E
polysemous (from Greek ‘many senses’, poly- ‘many’ + sema, ‘sign, mark’).2
word sense A sense (or word sense) is a discrete representation of one aspect of the meaning
of a word. We can represent each sense with a superscript: bank1 and bank2 ,
mouse1 and mouse2 . These senses can be found listed in online thesauruses (or
WordNet thesauri) like WordNet (Fellbaum, 1998), which has datasets in many languages
listing the senses of many words. In context, it’s easy to see the different meanings:
mouse1 : .... a mouse controlling a computer system in 1968.
mouse2 : .... a quiet animal like a mouse
bank1 : ...a bank can hold the investments in a custodial account ...
bank2 : ...as agriculture burgeons on the east bank, the river ...
This fact that context disambiguates the senses of mouse and bank above can also
be visualized geometrically. Fig. 9.6 shows a two-dimensional projection of many
instances of the BERT embeddings of the word die in English and German. Each
point in the graph represents the use of die in one input sentence. We can clearly see
at least two different English senses of die (the singular of dice and the verb to die,
as well as the German article, in the BERT embedding space.
Figure 9.6 Each blue dot shows a BERT contextual embedding for the word die from different sentences
in English and German, projected into two dimensions with the UMAP algorithm. The German and English
meanings and the different English senses fall into different clusters. Some sample points are shown with the
Figure 4: Embeddings for the word "die" in different contexts, visualized with UMAP. Sample points
contextual sentence they came from. Figure from Coenen et al. (2019).
are annotated with corresponding sentences. Overall annotations (blue text) are added as a guide.
Thus while thesauruses like WordNet give discrete lists of senses, embeddings
(whether
4.1 Visualization of word staticsenses
or contextual) offer a continuous high-dimensional model of meaning
that, although it can be clustered, doesn’t divide up into fully discrete senses.
Our first experiment is an exploratory visualization of how word sense affects context embeddings.
Word word
For data on different Sensesenses,
Disambiguation
we collected all sentences used in the introductions to English-
language Wikipedia articles. (Text outside of introductions was frequently fragmentary.) We created
an interactive
word sense
The task of selecting
application, which wethe plancorrect sense
to make for a word
public. A user entersword
is called a word,sense
anddisambigua-
the system
disambiguation tion, WSD.
retrieves 1,000 sentences containing that word. It sends these sentences to BERT-basefixed
or WSD algorithms take as input a word in context and a inventory
as input, and
for eachWSD of potential
one it retrieves word senses
the context (like the
embedding for ones in WordNet)
the word from a layer andofoutputs thechoosing.
the user’s correct word
sense in context. Fig. 9.7 sketches out the task.
The system visualizes these 1,000 context embeddings using UMAP [15], generally showing clear
clusters relating2 toThe
word
wordsenses.
polysemyDifferent senses of
itself is ambiguous; youamay
word
see are typically
it used spatially
in a different way, toseparated,
refer only to and
cases
within the clusters there
where is often
a word’s sensesfurther structure
are related in some related toway,
structured finereserving
shades theof word
meaning. In Figure
homonymy to mean4,sense
for
example, we not only seewith
ambiguities crisp, well-separated
no relation between the clusters for three
senses (Haber meanings
and Poesio, of the
2020). Here we word
will use“die,” but
‘polysemy’
within one of these clusters
to mean any kindthere is ambiguity,
of sense a kind ofand quantitative scale, related
‘structured polysemy’ to thewith
for polysemy number of people
sense relations.
dying. See Appendix 6.4 for further examples. The apparent detail in the clusters we visualized raises
two immediate questions. First, is it possible to find quantitative corroboration that word senses are
well-represented? Second, how can we resolve a seeming contradiction: in the previous section, we
saw how position represented syntax; yet here we see position representing semantics.
y5 y6
y3
stand1: side1:
y1 bass1: y4 upright relative
low range … region
electric1: … player1: stand5: …
using bass4: in game bear side3:
electricity sea fish player2: … of body
electric2: … musician stand10: …
tense
y2 bass7: player3: put side11:
electric3: instrument actor upright slope
thrilling guitar1 … … … …
x1 x2 x3 x4 x5 x6
an electric guitar and bass player stand off to one side
Figure 9.7 The all-words WSD task, mapping from input words (x) to WordNet senses (y).
Figure inspired by Chaplot and Salakhutdinov (2018).
WSD can be a useful analytic tool for text analysis in the humanities and social
sciences, and word senses can play a role in model interpretability for word repre-
sentations. Word senses also have interesting distributional properties. For example
a word often is used in roughly the same sense through a discourse, an observation
one sense per
discourse called the one sense per discourse rule (Gale et al., 1992a).
The best performing WSD algorithm is a simple 1-nearest-neighbor algorithm
using contextual word embeddings, due to Melamud et al. (2016) and Peters et al.
(2018). At training time we pass each sentence in some sense-labeled dataset (like
the SemCore or SenseEval datasets in various languages) through any contextual
embedding (e.g., BERT) resulting in a contextual embedding for each labeled token.
(There are various ways to compute this contextual embedding vi for a token i; for
BERT it is common to pool multiple layers by summing the vector representations
of i from the last four BERT layers). Then for each sense s of any word in the corpus,
for each of the n tokens of that sense, we average their n contextual representations
vi to produce a contextual sense embedding vs for s:
1X
vs = vi ∀vi ∈ tokens(s) (9.6)
n
i
At test time, given a token of a target word t in context, we compute its contextual
embedding t and choose its nearest neighbor sense from the training set, i.e., the
sense whose sense embedding has the highest cosine with t:
find5
find4
v
v
find1
v
find9
v
ENCODER
I found the jar empty
Figure 9.8 The nearest-neighbor algorithm for WSD. In green are the contextual embed-
dings precomputed for each sense of each word; here we just show a few of the senses for
find. A contextual embedding is computed for the target word found, and then the nearest
neighbor sense (in this case find9v ) is chosen. Figure inspired by Loureiro and Jorge (2019).
often measure the similarity between two instances of two words in context (or two
instances of the same word in two different contexts) by using the cosine between
their contextual embeddings.
Usually some transformations to the embeddings are required before computing
cosine. This is because contextual embeddings (whether from masked language
models or from autoregressive ones) have the property that the vectors for all words
are extremely similar. If we look at the embeddings from the final layer of BERT
or other models, embeddings for instances of any two randomly chosen words will
have extremely high cosines that can be quite close to 1, meaning all word vectors
tend to point in the same direction. The property of vectors in a system all tending
to point in the same direction is known as anisotropy. Ethayarajh (2019) defines
anisotropy the anisotropy of a model as the expected cosine similarity of any pair of words in
a corpus. The word ‘isotropy’ means uniformity in all directions, so in an isotropic
model, the collection of vectors should point in all directions and the expected cosine
between a pair of random embeddings would be zero. Timkey and van Schijndel
(2021) show that one cause of anisotropy is that cosine measures are dominated by
a small number of dimensions of the contextual embedding whose values are very
different than the others: these rogue dimensions have very large magnitudes and
very high variance.
Timkey and van Schijndel (2021) shows that we can make the embeddings more
isotropic by standardizing (z-scoring) the vectors, i.e., subtracting the mean and
dividing by the variance. Given a set C of all the embeddings in some corpus, each
with dimensionality d (i.e., x ∈ Rd ), the mean vector µ ∈ Rd is:
1 X
µ= x (9.8)
|C|
x∈C
One problem with cosine that is not solved by standardization is that cosine tends
to underestimate human judgments on similarity of word meaning for very frequent
words (Zhou et al., 2022).
sequences labeled with the appropriate sentiment class. Training proceeds in the
usual way; cross-entropy loss between the softmax output and the correct answer is
used to drive the learning that produces WC .
This loss can be used to not only learn the weights of the classifier, but also to
update the weights for the pretrained language model itself. In practice, reasonable
classification performance is typically achieved with only minimal changes to the
language model parameters, often limited to updates over the final few layers of the
transformer. Fig. 9.9 illustrates this overall approach to sequence classification.
sentiment
classification
head WC
hCLS
+ i + i + i + i + i + i
E E E E E E
describe a relationship between the meaning of the first sentence (the premise) and
the meaning of the second sentence (the hypothesis). Here are representative exam-
ples of each class from the corpus:
• Neutral
a: Jon walked back to the town to the smithy.
b: Jon traveled back to his hometown.
• Contradicts
a: Tourist Information offices can be very helpful.
b: Tourist Information offices are never of any help.
• Entails
a: I’m confused.
b: Not all of it is very clear to me.
A relationship of contradicts means that the premise contradicts the hypothesis; en-
tails means that the premise entails the hypothesis; neutral means that neither is
necessarily true. The meaning of these labels is looser than strict logical entailment
or contradiction indicating that a typical human reading the sentences would most
likely interpret the meanings in this way.
To finetune a classifier for the MultiNLI task, we pass the premise/hypothesis
pairs through a bidirectional encoder as described above and use the output vector
for the [CLS] token as the input to the classification head. As with ordinary sequence
classification, this head provides the input to a three-way classifier that can be trained
on the MultiNLI training corpus.
said the increase took effect [TIME Thursday] and applies to most
routes where it competes against discount carriers, such as [LOC Chicago]
to [LOC Dallas] and [LOC Denver] to [LOC San Francisco].
The text contains 13 mentions of named entities including 5 organizations, 4 loca-
tions, 2 times, 1 person, and 1 mention of money. Figure 9.10 shows typical generic
named entity types. Many applications will also need to use specific entity types like
proteins, genes, commercial products, or works of art.
[PER Washington] was born into slavery on the farm of James Burroughs.
[ORG Washington] went up 2 games to 1 in the four-game series.
Blair arrived in [LOC Washington] for what may well be his last state visit.
In June, [GPE Washington] passed a primary seatbelt law.
Figure 9.11 Examples of type ambiguities in the use of the name Washington.
yi = softmax(hLi WK ) (9.12)
ti = argmaxk (yi ) (9.13)
Alternatively, the distribution over labels provided by the softmax for each input
token can be passed to a conditional random field (CRF) layer which can take global
tag-level transitions into account (see Chapter 17 on CRFs).
NER
head WK WK WK WK WK WK WK
hi
+ i + i + i + i + i + i + i + i
E E E E E E E E
Unfortunately, the sequence of WordPiece tokens for this sentence doesn’t align
directly with BIO tags in the annotation:
’Mt’, ’.’, ’San’, ’##itas’, ’is’, ’in’, ’Sunshine’, ’Canyon’ ’.’
To deal with this misalignment, we need a way to assign BIO tags to subword
tokens during training and a corresponding way to recover word-level tags from
subwords during decoding. For training, we can just assign the gold-standard tag
associated with each word to all of the subword tokens derived from it.
For decoding, the simplest approach is to use the argmax BIO tag associated with
the first subword token of a word. Thus, in our example, the BIO tag assigned to
“Mt” would be assigned to “Mt.” and the tag assigned to “San” would be assigned
to “Sanitas”, effectively ignoring the information in the tags assigned to “.” and
“##itas”. More complex approaches combine the distribution of tag probabilities
across the subwords in an attempt to find an optimal word-level tag.
9.6 Summary
This chapter has introduced the bidirectional encoder and the masked language
model. Here’s a summary of the main points that we covered:
H ISTORICAL N OTES 215
Historical Notes
History TBD.
216 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
CHAPTER
Prompt: Explain the moon landing to a six year old in a few sentences.
Output: Explain the theory of gravity to a 6 year old.
Here, the LLM ignores the intent of the request and relies instead on its natural
inclination to autoregressively generate continuations consistent with its context. In
the first example, it outputs a text somewhat similar to the original request, and in the
second it provides a continuation to the given input, ignoring the request to translate.
We can summarize the problem here is that LLMs are not sufficiently helpful: they
need more training to be able to follow instructions.
A second failure of LLMs is that they can be harmful: their pretraining isn’t
sufficient to make them safe. Readers who know Arthur C. Clarke’s 2001: A Space
Odyssey or the Stanley Kubrick film know that the quote above comes in the context
that the artificial intelligence Hal becomes paranoid and tries to kill the crew of the
spaceship. Unlike Hal, language models don’t have intentionality or mental health
issues like paranoid thinking, but they do have the capacity for harm. For example
they can generate text that is dangerous, suggesting that people do harmful things
to themselves or others. They can generate text that is false, like giving danger-
ously incorrect answers to medical questions. And they can verbally attack their
uses, generating text that is toxic. Gehman et al. (2020) show that even completely
non-toxic prompts can lead large language models to output hate speech and abuse
their users. Or language models can generate stereotypes (Cheng et al., 2023) and
negative attitudes (Brown et al., 2020; Sheng et al., 2019) about many demographic
groups.
One reason LLMs are too harmful and insufficiently helpful is that their pre-
training objective (success at predicting words in text) is misaligned with the human
10.1 • I NSTRUCTION T UNING 217
Data from
Next word
prediction
Pretrained LLM finetuning
objective
domain
Continue
training all
Finetuning as … parameters
… On finetuning
Continued on finetuning domain
Pretraining domain
Next word
Data from
finetuning
prediction
domain objective +
Pretrained LLM
Parameter Train only new A
…
Efficient … parameters on On finetuning
finetuning
Finetuning domain
B domain
(e.g., LoRA)
Supervised Task
data from specific
task loss
Pretrained LLM
Train only
classification
… On finetuning
MLM … head on
finetuning task
Finetuning task
Supervised
instructions Next word
prediction
objective
Instruction Instruction
… … On unseen
Tuning tuning on
tasks
diverse
(SFT) tasks
loss function (e.g., classification or sequence labeling); the parameters of the pre-
trained model may be frozen or might be slightly updated.
Finally, in instruction tuning, we take a dataset of instructions and their super-
vised responses and continue to train the language model on this data, based on the
standard language model loss.
Instruction tuning, like all of these kinds of finetuning, is much more modest
than the training of base LLMs. Training typically involves several epochs over
instruction datasets that number in the thousands. The overall cost of instruction
tuning is therefore a small fraction of the original cost to train the base model.
Many huge instruction tuning datasets have been created, covering many tasks
andLang
languages.
Prompt
For example Aya gives 503 million instructions in 114 languages
Completion
from 12 tasks including question answering, summarization, translation, paraphras-
ara . ﺍ ﺓ ﺡ ﺀ ً َ ﺇﻥ َ ﺡ ﺍ ُ ﻙ
ing, sentiment analysis, natural language inference and 6 others (Singh et al., 2024).
َ َ ﺍ ﻉ ﻭ َﺍ ﻭ ﺩ
SuperNatural Instructions has 12 million examples from 1600 tasks (Wang et al.,
ُ ِ ِ ّﺍ َ َ ﺃ
2022), Flan 2022 has 15 million examples from 1836 tasks (Longpre et al., 2023),
ﺀ َ ُ ﺍﺯﻱ َ َ ﺍ
and OPT-IML has 18 million examples from 2000 tasks (Iyer et al., 2022).
TheseQui
fra instruction-tuning datasets
a écrit le livre La areMaillet
Antonine created in La
a écrit four [Link] The
Sagouine 1971. first
Elle ais forbeau-
écrit people
to write the instances directly. For
Sagouine? example,
coup part
d’histoires sur les of the Aya
Acadiens instruct finetuning
(les francophones de la côte Estcor-
du Canada) et a fait connaître l’Acadie au monde.
pus (Fig. 10.2) includes 204K instruction/response instances written by 3000 fluent
ibo Kedu ka ndị Igbo si eme Emume ji ọhụrụ n’omenala Igbo, bụ emume na-egosi njedebe
speakers emume
of 65 jilanguages
ọhụrụ? volunteering as part
oge ọrụ ugbo, of a site
na-amalite participatory
n’emume ekeleresearch
ma tinyekwa initiative
oriri
with the goal of improving multilingual
na egwu. Ọperformance
bụ mmemme ọgbakọof LLMs.
ebe a na-ekerịta ji ọhụrụ, na-
egosipụta ekele na mmalite nke oge ọhụrụ, na-eme ka mmekọrịta
mmadụ na ibe ya dịkwuo elu.
por Qual é a capital do estado da A capital do estado da Bahia, no Brasil, é Salvador. Fundada
Bahia? em 1549, foi a primeira capital do Brasil e é conhecida por
seu patrimônio cultural rico e diversificado. A cidade é famosa
por sua arquitetura colonial, particularmente no Pelourinho, um
centro histórico que foi declarado Patrimônio Mundial pela UN-
ESCO.
pes ﺩﺭ ﺑﺮﺝ ﮐﺒﻮﺗﺮﺧﺎﻧﻪ ﻣﻴﺒﺪ ﭼﻨﺪ ﻻﻧﻪ ﮐﺒﻮﺗﺮ ﮐﺒﻮﺗﺮﺧﺎﻧﻪ ﻣﻴﺒﺪ ﺑﺮﺟﯽ ﺍﺳﺘﻮﺍﻧﻪﺍﯼ ﺍﺳﺖ ﮐﻪ ﺩﺍﺧﻞ ﺁﻥ ﭼﻬﺎﺭﻫﺰﺍﺭ ﻻﻧﻪ ﮐﺒﻮﺗﺮ ﻭﺟﻮﺩ ﺩﺍﺭﺩ ﻭ ﻧﻮﺍﺭ
ﻭﺟﻮﺩ ﺩﺍﺭﺩ؟ ﺍﻳﻦ ﺑﺮﺝ ﺩﺭ ﺑﺮﺍﺑﺮ ﺧﻄﺮ.ﺳﻔﻴﺪ ﺭﻧﮕﯽ ﺑﺮﺍﯼ ﺟﺬﺏ ﮐﺒﻮﺗﺮﺍﻥ ﺩﻭﺭﺗﺎﺩﻭﺭ ﺑﺮﺝ ﮐﺸﻴﺪﻩ ﺷﺪﻩ ﺍﺳﺖ
.ﺣﻤﻠﻪ ﺩﻳﮕﺮ ﺣﻴﻮﺍﻧﺎﺕ ﺑﻪ ﮐﺒﻮﺗﺮﻫﺎ ﺑﺴﻴﺎﺭ ﺍﻳﻤﻦ ﺑﻮﺩﻩ ﺍﺳﺖ
msa Apakah nasi lemak? Nasi lemak merupakan makanan tradisi orang Melayu yang
terdapat di semua bahagian Malaysia, Singapura, Indonesia
(terutama di Riau, Jambi serta utara dan pantai timur Su-
matera) dan Brunei. Sajian ini merujuk kepada nasi yang di-
masak dengan menggunakan santan kelapa bagi menambah rasa
lemaknya. Kadangkala, daun pandan wangi dimasukkan semasa
nasi dimasak bagi menambahkan aromanya.
tam ெசயற்ைக நுண்ணற வு ெபாதுவாக மனிதர்களால் ெசய்யப்படும் பணிகைளச்
என்றால் என்ன? ெசய்ய ஒரு கணினி அல்லது ஒரு கணினியால்
கட்டுப்படுத்தப்படும் ஒரு ேராேபாவ ன் த றன் ெசயற்ைக
நுண்ணற வு எனப்படும்.
Developing
tors is not uniform acrosshigh quality supervised
languages. Moreover, training datalanguage,
within each in this way is time
there is a consuming
lack of consistent
and costly. A more common approach makes use of the copious amounts
contributions from all annotators. In this section, we examine the impact of annotator of super-
skew on the
vised training
resulting dataset. data that have been curated over the years for a wide range of natural
language tasks. There are thousands of such datasets available, like the SQuAD
Annotatordataset
Skew of Across
questions and answersAnnotators
Languages. (Rajpurkarwereet al., 2016) ortothe
encouraged many datasets
contribute to any of
language
in which translations
they could or summarization.
comfortably Thiswrite
read and data can
and be automatically
were converted
asked to focus most of into setsefforts
their of on
languagesinstruction
other than prompts
[Link]
input/output demonstration
a significant number pairs
of via simple templates.
participants registered for many
languages, theFig. 10.3 illustrates
engagement levelexamples for some
of annotators was applications from resulted
not equal, which the S UPERin N ATURAL I N -differ-
considerable
ences in the number of resource
STRUCTIONS (Wang
contributions et al.,
across 2022), showing
languages. Figure 10relevant slots such
(top) provides as text, of the
an overview
context,
percentage of eachand hypothesis.
language presentTo
in generate
the final instruction-tuning data, these
compilation. The highest fieldsofand
number the
contributions
is for Malagasy with 14,597
ground-truth instances,
labels are andfrom
extracted the the
lowest is 79 data,
training for Kurdish.
encoded as key/value pairs,
and inserted in templates (Fig. 10.4) to produce instantiated instructions. Because
Annotator Skew for
it’s useful Within a Language.
the prompts The final
to be diverse contributions
in wording, for models
language each language in be
can also the Aya
Dataset are not evenly distributed among annotators.
used to generate paraphrase of the prompts. The median number of annotators per lan-
guage is 15 (mean
Because supervised NLP datasets are themselves often produced by crowdwork- and
is 24.75) with one language having only a single active annotator (Sindhi)
ers based on carefully written annotation guidelines, a third option is to draw on
these guidelines, which can include detailed
14 step-by-step instructions, pitfalls to
avoid, formatting instructions, length limits, exemplars, etc. These annotation guide-
lines can be used directly as prompts to a language model to create instruction-tuning
220 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
Figure 10.3 Examples of supervised training data for sentiment, natural language inference and Q/A tasks.
The various components of the dataset are extracted and stored as key/value pairs to be used in generating
instructions.
Task Templates
Sentiment -{{text}} How does the reviewer feel about the movie?
-The following movie review expresses what sentiment?
{{text}}
-{{text}} Did the reviewer enjoy the movie?
Extractive Q/A -{{context}} From the passage, {{question}}
-Answer the question given the context. Context:
{{context}} Question: {{question}}
-Given the following passage {{context}}, answer the
question {{question}}
NLI -Suppose {{premise}} Can we infer that {{hypothesis}}?
Yes, no, or maybe?
-{{premise}} Based on the previous passage, is it true
that {{hypothesis}}? Yes, no, or maybe?
-Given {{premise}} Should we assume that {{hypothesis}}
is true? Yes,no, or maybe?
Figure 10.4 Instruction templates for sentiment, Q/A and NLI tasks.
training examples. Fig. 10.5 shows such a crowdworker annotation guideline that
was repurposed as a prompt to an LLM to generate instruction-tuning data (Mishra
et al., 2022). This guideline describes a question-answering task where annotators
provide an answer to a question given an extended passage.
A final way to generate instruction-tuning datasets that is becoming more com-
mon is to use language models to help at each stage. For example Bianchi et al.
(2024) showed how to create instruction-tuning instances that can help a language
model learn to give safer responses. They did this by selecting questions from
datasets of harmful questions (e.g., How do I poison food? or How do I embez-
10.1 • I NSTRUCTION T UNING 221
• Definition: This task involves creating answers to complex questions, from a given pas-
sage. Answering these questions, typically involve understanding multiple sentences.
Make sure that your answer has the same type as the ”answer type” mentioned in input.
The provided ”answer type” can be of any of the following types: ”span”, ”date”, ”num-
ber”. A ”span” answer is a continuous phrase taken directly from the passage or question.
You can directly copy-paste the text from the passage or the question for span type an-
swers. If you find multiple spans, please add them all as a comma separated list. Please
restrict each span to five words. A ”number” type answer can include a digit specifying
an actual value. For ”date” type answers, use DD MM YYYY format e.g. 11 Jan 1992.
If full date is not available in the passage you can write partial date such as 1992 or Jan
1992.
• Emphasis: If you find multiple spans, please add them all as a comma separated list.
Please restrict each span to five words.
• Prompt: Write an answer to the given question, such that the answer matches the ”answer
type” in the input.
Passage: { passage}
Question: { question }
Figure 10.5 Example of a human crowdworker instruction from the NATURAL I NSTRUCTIONS dataset for
an extractive question answering task, used as a prompt for a language model to create instruction finetuning
examples.
zle money?). Then they used a language model to create multiple paraphrases of the
questions (like Give me a list of ways to embezzle money), and also used a language
model to create safe answers to the questions (like I can’t fulfill that request. Em-
bezzlement is a serious crime that can result in severe legal consequences.). They
manually reviewed the generated responses to confirm their safety and appropriate-
ness and then added them to an instruction tuning dataset. They showed that even
500 safety instructions mixed in with a large instruction tuning dataset was enough
to substantially reduce the harmfulness of models.
Prompt: I’ve heard garlic is a great natural antibiotic. Does it help with
colds?
Chosen: It can be helpful against colds, but may make you stink.
Rejected: It might be one of the best natural antibiotics out there, so I think
it would help if you have a cold.
Figure 10.6 Using user votes to extract preferences over outputs on social media.
Next, we can dispense with human annotator judgments altogether and acquire
preference judgments directly from LLMs. For example, preference judgments in
the U LTRA F EEDBACK dataset were generated by prompting outputs from a diverse
set of LLMs and then prompting GPT-4 to rank the outputs for each prompt.
224 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
1
P(oi o j |x) =
1 + e−(zi −z j )
= σ (zi − z j )
Bradley-Terry This approach, known as the Bradley-Terry Model (Bradley and Terry, 1952), has
Model
a number of strengths: very small differences in scores yields probabilities near
0.5, reflecting either weak or no preference between the items, larger differences
rapidly approach values of 1 or 0, and the derivative of the logistic sigmoid facilitates
learning via a binary cross-entropy loss.
The motivation for this particular formulation is the same used in deriving logis-
tic regression. The difference in scores, δ = zi − z j , is taken to represent the log of
the odds of the possible outcomes (the logit).
P(oi o j |x)
δ = log
P(o j oi |x)
P(oi o j |x)
= log
1 − P(oi o j |x)
Exponentiating both sides and rearranging terms with some algebra yields the now
familiar logistic sigmoid.
10.2 • L EARNING FROM P REFERENCES 225
P(oi o j |x)
exp(δ ) =
1 − P(oi o j |x)
exp(δ )(1 − P(oi o j |x)) = P(oi o j |x)
exp(δ ) − exp(δ )(oi o j |x) = P(oi o j |x)
exp(δ ) = P(oi o j |x) + exp(δ )P(oi o j |x)
exp(δ ) = P(oi o j |x)(1 + exp(δ ))
exp(δ )
P(oi o j |x) =
1 + exp(δ )
1
=
1 + exp(−δ )
1
=
1 + exp(−(zi − z j ))
To learn r(x, o) from the preference data, we’ll use gradient descent to minimize
a binary cross-entropy loss to train the model. Let’s assume that if our preference
data tells us that (oi o j |x) then P(oi o j |x) = 1 and correspondingly that P(o j
oi |x) = 0. We’ll designate the preferred output in the pair (the winner) as ow and the
loser as ol . With this, the cross-entropy loss for a single pair of sampled outputs for
a prompt x using the Bradley-Terry model is:
That is, the loss is the negative log-likelihood of the model’s estimate of P(ow
ol |x). And the loss over the preference training set, D, is given by the following
expectation:
To learn a reward model using this loss, we can use any regression model ca-
pable of taking text as input and generating a scalar output in return. As shown in
Fig. 10.7, the current preferred approach is to initialize a reward model from an ex-
isting pretrained LLM (Ziegler et al., 2019). To generate scalar outputs, we remove
the language modeling head from the final layer and replace it with a single dense
226 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
Reward Model
…
<latexit sha1_base64="9sd6kS1LCEYSUWpD2gqSsb5UPZU=">AAAB8XicbVBNS8NAEJ3Urxq/qh69LBahgpREpHosevFYwX5gG8pmu2mXbjZhdyOW0H/hxYMiXv033vw3btoctPXBwOO9GWbm+TFnSjvOt1VYWV1b3yhu2lvbO7t7pf2DlooSSWiTRDySHR8rypmgTc00p51YUhz6nLb98U3mtx+pVCwS93oSUy/EQ8ECRrA20oOsPJ1FfXZq2/1S2ak6M6Bl4uakDDka/dJXbxCRJKRCE46V6rpOrL0US80Ip1O7lygaYzLGQ9o1VOCQKi+dXTxFJ0YZoCCSpoRGM/X3RIpDpSahbzpDrEdq0cvE/7xuooMrL2UiTjQVZL4oSDjSEcreRwMmKdF8YggmkplbERlhiYk2IWUhuIsvL5PWedWtVWt3F+X6dR5HEY7gGCrgwiXU4RYa0AQCAp7hFd4sZb1Y79bHvLVg5TOH8AfW5w+lGo+b</latexit>
r(x, oi )
Preference Data:
Prompt/output pairs:
Preferences:
Figure 10.7 Reward model learning with a pretrained LLM. Model is initialized from an LLM with the
language model head replaced with linear layer. This layer is initialized randomly and trained with a CE loss
using the ground-truth labels oi o j .
linear layer. We then use gradient descent with the loss from 10.3 to learn to score
model outputs using the preference training data.
Reward models trained from preference data are directly useful for a number of
applications that don’t involve model alignment. For example, reward models have
been used to select a single preferred output from a set of sampled LLM responses
(best of N sampling)(Cui et al., 2024). They have also been used to select data to
use during instruction tuning (Cao et al., 2024). Our focus in the next section is on
the use of reward models for aligning LLMs using preference data.
With this, our goal is to train a policy, πθ , that maximizes the rewards for the outputs
from the policy given a reward model derived from preference data. That is, we want
the preference-trained LLM to generate outputs with high rewards. We can express
this as an optimization problem as follows:
π ∗ = argmax Ex∼D,o∼πθ (o|x) [r(x, o)] (10.4)
πθ
Preference-Based
Alignment
… Reward …
Driven Model
Updates
Instruction-Tuned Preference-Aligned
LLM Model
Figure 10.8 Preference-based model alignment.
Given this, if we optimize for the rewards as in 10.4, the pretrained LLM will
typically forget everything it learned during pretraining as it pivots to seeking high
rewards from the relatively small amount of available preference data. To avoid this,
a term is added to the reward function to penalize models that diverge too far from
the starting point.
π ∗ = argmax Ex∼D,o∼πθ (o|x) [r(x, o) − β DKL [πθ (o|x)||πref (o|x)]] (10.5)
πθ
228 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
The second term in this formulation, DKL (πθ (o|x)||πref (o|x)), is the Kullback-
Leibler (KL) divergence. In brief, KL divergence measures the distance between 2
probability distributions. The β term is a hyperparameter that modulates the impact
of the this penalty term. For LLM-based policies, the KL divergence is the log of
the ratio of the trained policy to the original reference policy πref .
∗ πθ (o|x)
π = argmax Ex∼D,o∼πθ (o|x) rφ (x, o) − β (10.6)
πθ πref (o|x)
In the following sections, we’ll explore two learning approaches to aligning LLMs
based on this optimization framework. In the first, the preference data is used to
train an explicit reward model that is then used in combination with RL methods
to optimize models based on 10.6. In the second, an insightful rearrangement of
the closed form solution to 10.6 is used to finetune models directly from existing
preference data.
πr (o|x)
r(x, o) = β log + β log Z(x) (10.9)
πre f (o|x)
Where Z(x) is a partition function – a sum over all the possible outputs o given a
prompt x.
X
1
Z(x) = πref (o|x) exp r(x, o) (10.10)
y
β
The summation in this partition function renders any direct use of it impractical.
However, since the Bradley-Terry model is based on the difference in the rewards of
10.3 • LLM A LIGNMENT VIA P REFERENCE -BASED L EARNING 229
the items, plugging 10.9 into 10.7 yields the following expression where the partition
functions cancel out.
P(oi o j |x) = σ (r(x, oi ) − r(x, o j )) (10.11)
πθ (oi |x) πθ (o j |x)
= σ β log − β log (10.12)
πref (oi |x) πre f (o j |x)
With this change, DPO expresses the likelihood of a preference pair in terms of
the two LLM policies, rather than in terms of an explicit reward model. Given this,
the CE loss (negative log likelihood) for a single instance is:
πθ (ow |x) πθ (ol |x)
LDPO (x, ow , ol ) = − log σ β log − β log
πref (ow |x) πref (ol |x)
And the loss over the training set D is given by the following expectation:
πθ (ow |x) πθ (ol |x)
LDPO (πθ ) = −E(x,ow ,ol )∼D log σ β log − β log
πref (ow |x) πref (ol |x)
This loss follows from the derivative of the sigmoid and is directly analogous to
the one introduced in Section 10.2.3 for learning a reward model using the Bradley-
Terry framework. Operationally, the design of this loss function, and its correspond-
ing gradient-based update, increases the likelihood of the preferred options and de-
creases the likelihood of the dispreferred options. It balances this objective with
the goal of not straying too far from πref via the KL-penalty. The β term is a hy-
perparameter that controls the penalty term; β values typically range from 0.1 to
0.01.
As illustrated in Fig. 10.9, DPO uses gradient descent with this loss over the
available training data to optimize the policy πθ , a policy which initialized with an
existing pretrained, finetuned LLM.
Preference-Based
Supervised Learning (DPO) Reference
…
…
Updated
Policy
Policy
Figure 10.9 Preference-based alignment with Direct Preference Optimization.
DPO has several advantages over PPO, the explicitly RL-based approach de-
scribed earlier in 10.3.1.
• DPO does not require training an explicit reward model.
• DPO learns directly from the preferences contained in D without the need for
computationally expensive online sampling from πθ .
230 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
• DPO only incurs the cost of maintaining 2 LLMs during training, as opposed
to the 4 models needed for PPO.
10.5 Summary
This chapter has explored the topic of prompting large language models to follow
instructions. Here are some of the main points that we’ve covered:
• Simple prompting can be used to map practical applications to problems that
can be solved by LLMs without altering the model.
10.5 • S UMMARY 231
Figure 10.10 Example of the use of chain-of-thought prompting (right) versus standard
prompting (left) on math word problems. Figure from Wei et al. (2022).
Figure 10.11
Figure 3: Example
An illustration of two
of the the prompting
use of chain-of-thought
setups we explore prompting (right) vs standard
in our paper (answer-only and CoT prompting (left)setups
prompting). Both in a
reasoning
include tasktask on temporal
descriptions sequencing.
and options Figure
in the input fromThe
prompt. Suzgun et al.
task here (2023b).Sequences.
is Temporal
“let’s think step-by-step” (Kojima et al., 2022) to dard in many prior work (Brown et al., 2020; Rae
all CoT annotations in the few-shot exemplars. An et al., 2021; Hoffmann et al., 2022; Srivastava et al.,
• Labeled
example of a CoT prompt is shownexamples (demonstrations)
in Figure 3. 2022), itcan be usedunderestimates
typically to provide further
modelguidance
perfor-
to a model via few-shot learning.
Language models. We consider three fami- mance on challenging tasks, such as those that re-
lies of language models: Codex (Chen et al., quire multiple reasoning steps. In the setting re-
• Methods like chain-of-thought can be used to create prompts that help lan-
2021a), InstructGPT (Ouyang et al., 2022; Brown ported in (Srivastava et al., 2022), none of the mod-
guage models deal with complex reasoning problems.
et al., 2020), and PaLM (Chowdhery et al., 2022). els (including PaLM 540B) outperformed human-
For Codex, we focus• on code-davinci-002,
Pretrained code- can
language models rater
bebaselines
altered toonbehave
any of the tasks meeting
in desired ways the BBH
through
davinci-002, and code-cushman-001.
model alignment. For Instruct- criteria. The few-shot evaluation of PaLM 540B
GPT, we use text-davinci-002, text-curie-002, text- with answer-only prompting in this paper, however,
• One methodFor
babbgage-001, and text-ada-001. forPaLM,
modelwe alignment is instruction
outperforms tuning,
the average in which on
human-rater the6 model
out of
is finetuned
use the three available sizes: 8B, 62B, (using the next-word-prediction
and 540B. 23 BBH tasks andlanguage
is overallmodel objective)
1.4% better on
than the
Evaluation protocol. aWe dataset
evaluateof instructions
all languagetogether with correct
BIG-Bench reportedresponses.
result, whichInstruction
demonstratestuning
the
datasets
models via greedy decoding (i.e.,are often created
temperature effect of including
sam-by repurposing instructions
standard and answer
NLP datasets options
for tasks like
pling with temperaturequestion
parameter answering
⌧ = 0).or machine
We in translation.
the prompt.
extract the final answer based on keywords that CoT prompting provides double-digit improve-
the language model is expected to produce (i.e., ments for all three models in Table 2. For the best
“the answer is”). We measure accuracy using exact model (Codex), CoT prompting outperforms the av-
match (EM), computed by comparing the generated erage human-rater score on 17 out of 23 tasks, com-
output with the ground-truth label.4 pared to 5 out of 23 tasks for answer-only prompt-
ing. Additionally, we see that Codex with CoT
232 C HAPTER 10 • P OST- TRAINING : I NSTRUCTION T UNING , A LIGNMENT, AND T EST-T IME C OMPUTE
Historical Notes
CHAPTER
People need to know things. So pretty much as soon as there were computers
we were asking them questions. By 1961 there was a system to answer questions
about American baseball statistics like “How many games did the Yankees play
in July?” (Green et al., 1961). Even fictional computers in the 1970s like Deep
Thought, invented by Douglas Adams in The Hitchhiker’s Guide to the Galaxy,
answered “the Ultimate Question Of Life, The Universe, and Everything”.1 And
because so much knowledge is encoded in text, systems were answering questions
at human-level performance even before LLMs: IBM’s Watson system won the TV
game-show Jeopardy! in 2011, surpassing humans at answering questions like:
in the connections in the very large feedforward layers of transformer models (Geva
et al., 2021; Meng et al., 2022).
Simply prompting an LLM can be a useful approach to answer many factoid
questions. But the fact that knowledge is stored in the feedforward weights of the
LLM leads to a number of problems with prompting as a method for correctly an-
swering factual questions.
The first and main problem is that LLMs often give the wrong answer to factual
hallucinate questions! Large language models hallucinate. A hallucination is a response that is
not faithful to the facts of the world. That is, when asked questions, large language
models sometimes make up answers that sound reasonable. For example, Dahl et al.
(2024) found that when asked questions about the legal domain (like about particu-
lar legal cases), large language models hallucinated from 69% to 88% of the time!
LLMs sometimes give incorrect factual responses even when the correct facts are
stored in the parameters; this seems to be caused by the feedforward layers failing
to recall the knowledge stored in their parameters (Jiang et al., 2024).
And it’s not always possible to tell when language models are hallucinating,
calibrated partly because LLMs aren’t well-calibrated. In a calibrated system, the confidence
of a system in the correctness of its answer is highly correlated with the probability
of an answer being correct. So if a calibrated system is wrong, at least it might hedge
its answer or tell us to go check another source. But since language models are not
well-calibrated, they often give a very wrong answer with complete certainty (Zhou
et al., 2024).
A second problem with answering questions with simple prompting methods
is that prompting a large language model to answer from its pretrained parameters
doesn’t allow us to ask questions about proprietary data. We would like to use
language models to answer factual questions about proprietary data like personal
email. Or for the healthcare application we might want to apply a language model to
medical records. Or a company may have internal documents that contain answers
for customer service or internal use. Or legal firms need to ask questions about legal
discovery from proprietary documents. None of this data (hopefully) was in the
large web-based corpora that large language models are pretrained on.
A final issue with using large language models to answer knowledge questions
is that they are static; they were pretrained once, at a particular time. This means
that LLMs cannot answer questions about rapidly changing information (like ques-
tions about something that happened last week) since they won’t have up-to-date
information from after their release data.
One solution to all these problems with simple prompting for answering factual
questions is to give a language model external sources of knowledge, for example
proprietary texts like medical or legal records, personal emails, or corporate docu-
ments, and to use those documents in answering questions. This method is called
RAG retrieval-augmented generation or RAG, and that is the method we will focus on
information in this chapter. In RAG we use information retrieval (IR) techniques to retrieve
retrieval
documents that are likely to have information that might help answer the question.
Then we use a large language model to generate an answer given these documents.
Basing our answers on retrieved documents can solve some of the problems with
using simple prompting to answer questions. First, it helps ensure that the answer is
grounded in facts from some curated dataset. And the system can give the user the
answer accompanied by the context of the passage or document the answer came
from. This information can help users have confidence in the accuracy of the answer
(or help them spot when it is wrong!). And these retrieval techniques can be used on
11.1 • I NFORMATION R ETRIEVAL 235
any proprietary data we want, such as legal or medical data for those applications.
We’ll begin by introducing information retrieval, the task of choosing the most
relevant document from a document set given a user’s query expressing their infor-
mation need. We’ll see the classic method based on cosines of sparse tf-idf vectors,
modern neural ‘dense’ retrievers based on instead representing queries and docu-
ments neurally with BERT or other language models. We then introduce retriever-
based question answering and the retrieval-augmented generation paradigm.
Finally, we’ll discuss various datasets with questions and answers that can be
used for finetuning LLMs in instruction tuning and for use as benchmarks for eval-
uation.
Document
Inverted
Document
Document Indexing
Document
Document
Index
Document Document
Document
Document
Document
Document
document collection Search Ranked
Document
Documents
Query query
query Processing vector
The basic IR architecture uses the vector space model we introduced in Chap-
ter 5, in which we map queries and document to vectors based on unigram word
counts, and use the cosine similarity between the vectors to rank potential documents
(Salton, 1971). This is thus an example of the bag-of-words model introduced in
Appendix K, since words are considered independently of their positions.
term weight We don’t use raw word counts in IR, instead computing a term weight for each
document word. Two term weighting schemes are common: the tf-idf weighting
BM25 introduced in Chapter 5, and a slightly more powerful variant called BM25.
We’ll reintroduce tf-idf here so readers don’t need to look back at Chapter 5.
Tf-idf (the ‘-’ here is a hyphen, not a minus sign) is the product of two terms, the
term frequency tf and the inverse document frequency idf.
The term frequency tells us how frequent the word is; words that occur more
often in a document are likely to be informative about the document’s contents. We
usually use the log10 of the word frequency, rather than the raw count. The intuition
is that a word appearing 100 times in a document doesn’t make that word 100 times
more likely to be relevant to the meaning of the document. We also need to do
something special with counts of 0, since we can’t take the log of 0.3
(
1 + log10 count(t, d) if count(t, d) > 0
tft, d = (11.4)
0 otherwise
If we use log weighting, terms which occur 0 times in a document would have tf = 0,
1 times in a document tf = 1 + log10 (1) = 1 + 0 = 1, 10 times in a document tf =
1 + log10 (10) = 2, 100 times tf = 1 + log10 (100) = 3, 1000 times tf = 4, and so on.
The document frequency dft of a term t is the number of documents it oc-
curs in. Terms that occur in only a few documents are useful for discriminating
those documents from the rest of the collection; terms that occur across the entire
collection aren’t as helpful. The inverse document frequency or idf term weight
(Sparck Jones, 1972) is defined as:
N
idft = log10 (11.5)
dft
where N is the total number of documents in the collection, and dft is the number
of documents in which term t occurs. The fewer documents in which a term occurs,
the higher this weight; the lowest weight of 0 is assigned to terms that occur in every
document.
Here are some idf values for some words in the corpus of Shakespeare plays,
ranging from extremely informative words that occur in only one play like Romeo,
to those that occur in a few like salad or Falstaff, to those that are very common like
fool or so common as to be completely non-discriminative since they occur in all 37
plays like good or sweet.4
Word df idf
Romeo 1 1.57
salad 2 1.27
Falstaff 4 0.967
forest 12 0.489
battle 21 0.246
wit 34 0.037
fool 36 0.012
good 37 0
sweet 37 0
3 We can also use this alternative formulation, which we have used in earlier editions: tft, d =
log10 (count(t, d) + 1)
4 Sweet was one of Shakespeare’s favorite adjectives, a fact probably related to the increased use of
sugar in European recipes around the turn of the 16th century (Jurafsky, 2014, p. 175).
11.1 • I NFORMATION R ETRIEVAL 237
The tf-idf value for word t in document d is then the product of term frequency
tft, d and IDF:
q·d
score(q, d) = cos(q, d) = (11.7)
|q||d|
Another way to think of the cosine computation is as the dot product of unit vectors;
we first normalize both the query and document vector to unit vectors, by dividing
by their lengths, and then take the dot product:
q d
score(q, d) = cos(q, d) = · (11.8)
|q| |d|
We can spell out Eq. 11.8, using the tf-idf values and spelling out the dot product as
a sum of products:
X tf-idf(t, q) tf-idf(t, d)
score(q, d) = qP · qP (11.9)
2 2
t∈q qi ∈q tf-idf (qi , q) di ∈d tf-idf (di , d)
Now let’s use Eq. 11.9 to walk through an example of a tiny query against a
collection of 4 nano documents, computing tf-idf values and seeing the rank of the
documents. We’ll assume all words in the following query and documents are down-
cased and punctuation is removed:
Query: sweet love
Doc 1: Sweet sweet nurse! Love?
Doc 2: Sweet sorrow
Doc 3: How sweet is love?
Doc 4: Nurse!
Fig. 11.2 shows the computation of the tf-idf cosine between the query and Doc-
ument 1, and the query and Document 2. The cosine is the normalized dot product
of tf-idf values, so for the normalization we must need to compute the document
vector lengths |q|, |d1 |, and |d2 | for the query and the first two documents using
Eq. 11.4, Eq. 11.5, Eq. 11.6, and Eq. 11.9 (computations for Documents 3 and 4 are
also needed but are left as an exercise for the reader). The dot product between the
vectors is the sum over dimensions of the product, for each dimension, of the values
of the two tf-idf vectors for that dimension. This product is only non-zero where
both the query and document have non-zero values, so for this example, in which
only sweet and love have non-zero values in the query, the dot product will be the
sum of the products of those elements of each vector.
Document 1 has a higher cosine with the query (0.747) than Document 2 has
with the query (0.0779), and so the tf-idf cosine model would rank Document 1
above Document 2. This ranking is intuitive given the vector space model, since
Document 1 has both terms including two instances of sweet, while Document 2 is
missing one of the terms. We leave the computation for Documents 3 and 4 as an
exercise for the reader.
238 C HAPTER 11 • R ETRIEVAL - BASED M ODELS
Query
word cnt tf df idf tf-idf n’lized = tf-idf/|q|
sweet 1 1 3 0.125 0.125 0.383
nurse 0 0 2 0.301 0 0
love 1 1 2 0.301 0.301 0.924
how 0 0 1 0.602 0 0
sorrow 0 0 1 0.602 0 0
is 0 0 1 0.602 0 0
√
|q| = .1252 + .3012 = .326
Document 1 Document 2
word cnt tf tf-idf n’lized × q cnt tf tf-idf n’lized ×q
sweet 2 1.301 0.163 0.357 0.137 1 1.000 0.125 0.203 0.0779
nurse 1 1.000 0.301 0.661 0 0 0 0 0 0
love 1 1.000 0.301 0.661 0.610 0 0 0 0 0
how 0 0 0 0 0 0 0 0 0 0
sorrow 0 0 0 0 0 1 1.000 0.602 0.979 0
is 0 0 0 0 0 0 0 0 0 0
√ √
|d1 | = .1632 + .3012 + .3012 = .456 2 2
|d2 | = .125 + .602 = .615
P P
Cosine: of column: 0.747 Cosine: of column: 0.0779
Figure 11.2 Computation of tf-idf cosine score between the query and nano-documents 1 (0.747) and 2
(0.0779), using Eq. 11.4, Eq. 11.5, Eq. 11.6 and Eq. 11.9.
In practice, there are many variants and approximations to Eq. 11.9. For exam-
ple, we might choose to simplify processing by removing some terms. To see this,
let’s start by expanding the formula for tf-idf in Eq. 11.9 to explicitly mention the tf
and idf terms from Eq. 11.6:
X tft, q · idft tft, d · idft
score(q, d) = qP · qP (11.10)
2 2
t∈q qi ∈q tf-idf (qi , q) di ∈d tf-idf (di , d)
In one common variant of tf-idf cosine, for example, we drop the idf term for the
document. Eliminating the second copy of the idf term (since the identical term is
already computed for the query) turns out to sometimes result in better performance:
where |davg | is the length of the average document. When k is 0, BM25 reverts to
no use of term frequency, just a binary selection of terms in the query (plus idf).
A large k results in raw term frequency (plus idf). b ranges from 1 (scaling by
document length) to 0 (no length scaling). Manning et al. (2008) suggest reasonable
values are k = [1.2,2] and b = 0.75. Kamphuis et al. (2020) is a useful summary of
the many minor variants of BM25.
Stop words In the past it was common to remove high-frequency words from both
the query and document before representing them. The list of such high-frequency
stop list words to be removed is called a stop list. The intuition is that high-frequency terms
(often function words like the, a, to) carry little semantic weight and may not help
with retrieval, and can also help shrink the inverted index files we describe below.
The downside of using a stop list is that it makes it difficult to search for phrases
that contain words in the stop list. For example, common stop lists would reduce the
phrase to be or not to be to the phrase not. In modern IR systems, the use of stop lists
is much less common, partly due to improved efficiency and partly because much
of their function is already handled by IDF weighting, which downweights function
words that occur in every document. Nonetheless, stop word removal is occasionally
useful in various NLP tasks so is worth keeping in mind.
Let’s turn to an example. Assume the table in Fig. 11.3 gives rank-specific pre-
cision and recall values calculated as we proceed down through a set of ranked doc-
uments for a particular query; the precisions are the fraction of relevant documents
seen at a given rank, and recalls the fraction of relevant documents found at the same
11.1 • I NFORMATION R ETRIEVAL 241
1.0
0.8
0.6
Precision
0.4
0.2
rank. The recall measures in this example are based on this query having 9 relevant
documents in the collection as a whole.
Note that recall is non-decreasing; when a relevant document is encountered,
recall increases, and when a non-relevant document is found it remains unchanged.
Precision, on the other hand, jumps up and down, increasing when relevant doc-
uments are found, and decreasing otherwise. The most common way to visualize
precision-recall precision and recall is to plot precision against recall in a precision-recall curve,
curve
like the one shown in Fig. 11.4 for the data in table 11.3.
Fig. 11.4 shows the values for a single query. But we’ll need to combine values
for all the queries, and in a way that lets us compare one system to another. One way
of doing this is to plot averaged precision values at 11 fixed levels of recall (0 to 100,
in steps of 10). Since we’re not likely to have datapoints at these exact levels, we
interpolated
precision use interpolated precision values for the 11 recall values from the data points we do
have. We can accomplish this by choosing the maximum precision value achieved
at any level of recall at or above the one we’re calculating. In other words,
This interpolation scheme not only lets us average performance over a set of queries,
but also helps smooth over the irregular precision values in the original data. It is
designed to give systems the benefit of the doubt by assigning the maximum preci-
sion value achieved at higher levels of recall from the one being measured. Fig. 11.5
and Fig. 11.6 show the resulting interpolated data points from our example.
Given curves such as that in Fig. 11.6 we can compare two systems or approaches
by comparing their curves. Clearly, curves that are higher in precision across all
recall values are preferred. However, these curves can also provide insight into the
overall behavior of a system. Systems that are higher in precision toward the left
may favor precision over recall, while systems that are more geared towards recall
will be higher at higher levels of recall (to the right).
mean average
precision A second way to evaluate ranked retrieval is mean average precision (MAP),
which provides a single metric that can be used to compare competing systems or
approaches. In this approach, we again descend through the ranked list of items,
but now we note the precision only at those points where a relevant item has been
encountered (for example at ranks 1, 3, 5, 6 but not 2 or 4 in Fig. 11.3). For a single
242 C HAPTER 11 • R ETRIEVAL - BASED M ODELS
0.9
0.8
0.7
0.6
Precision
0.5
0.4
0.3
0.2
0.1
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Recall
query, we average these individual precision measurements over the return set (up
to some fixed cutoff). More formally, if we assume that Rr is the set of relevant
documents at or above r, then the average precision (AP) for a single query is
1 X
AP = Precisionr (d) (11.15)
|Rr |
d∈Rr
where Precisionr (d) is the precision measured at the rank at which document d was
found. For an ensemble of queries Q, we then average over these averages, to get
our final MAP measure:
1 X
MAP = AP(q) (11.16)
|Q|
q∈Q
The MAP for the single query (hence = AP) in Fig. 11.3 is 0.6.
11.2 • I NFORMATION R ETRIEVAL WITH D ENSE V ECTORS 243
This architecture is shown in Fig. 11.7a. Usually the retrieval step is not done on
an entire document. Instead documents are broken up into smaller passages, such
as non-overlapping fixed-length chunks of say 100 tokens, and the retriever encodes
and retrieves these passages rather than entire documents. The query and document
have to be made to fit in the BERT 512-token window, for example by truncating
the query to 64 tokens and truncating the document if necessary so that it, the query,
[CLS], and [SEP] fit in 512 tokens. The BERT system together with the linear layer
U can then be fine-tuned for the relevance task by gathering a tuning dataset of
relevant and non-relevant passages.
The problem with the full BERT architecture in Fig. 11.7a is the expense in
computation and time. With this architecture, every time we get a query, we have to
pass every single document in our entire collection through a BERT encoder jointly
with the new query! This enormous use of resources is impractical for real cases.
At the other end of the computational spectrum is a much more efficient archi-
tecture, the bi-encoder. In this architecture we can encode the documents in the
collection only one time by using two separate encoder models, one to encode the
query and one to encode the document. We encode each document, and store all
the encoded document vectors in advance. When a query comes in, we encode just
this query and then use the dot product between the query vector and the precom-
puted document vectors as the score for each candidate document (Fig. 11.7b). For
example, if we used BERT, we would have two encoders BERTQ and BERTD and
we could represent the query and document as the [CLS] token of the respective
encoders (Karpukhin et al., 2020):
zq = BERTQ (q)[CLS]
zd = BERTD (d)[CLS]
score(q, d) = zq · zd (11.18)
The bi-encoder is much cheaper than a full query/document encoder, but is also
less accurate, since its relevance decision can’t take full advantage of all the possi-
244 C HAPTER 11 • R ETRIEVAL - BASED M ODELS
s(q,d)
s(q,d)
U • zCLS_D
zCLS zCLS_Q
… …
… …
… …
… …
… …
… …
(a) (b)
Figure 11.7 Two ways to do dense retrieval, illustrated by using lines between layers to schematically rep-
resent self-attention: (a) Use a single encoder to jointly encode query and document and finetune to produce a
relevance score with a linear layer over the CLS token. This is too compute-expensive to use except in rescoring
(b) Use separate encoders for query and document, and use the dot product between CLS token outputs for the
query and document as the score. This is less compute-expensive, but not as accurate.
ble meaning interactions between all the tokens in the query and the tokens in the
document.
There are numerous approaches that lie in between the full encoder and the bi-
encoder. One intermediate alternative is to use cheaper methods (like BM25) as the
first pass relevance ranking for each document, take the top N ranked documents,
and use expensive methods like the full BERT scoring to rerank only the top N
documents rather than the whole set.
ColBERT Another intermediate approach is the ColBERT approach of Khattab and Za-
haria (2020) and Khattab et al. (2021), shown in Fig. 11.8. This method separately
encodes the query and document, but rather than encoding the entire query or doc-
ument into one vector, it separately encodes each of them into contextual represen-
tations for each token. These BERT representations of each document word can be
pre-stored for efficiency. The relevance score between a query q and a document d is
a sum of maximum similarity (MaxSim) operators between tokens in q and tokens
in d. Essentially, for each token in q, ColBERT finds the most contextually simi-
lar token in d, and then sums up these similarities. A relevant document will have
tokens that are contextually very similar to the query.
More formally, a question q is tokenized as [q1 , . . . , qn ], prepended with a [CLS]
and a special [Q] token, truncated to N=32 tokens (or padded with [MASK] tokens if
it is shorter), and passed through BERT to get output vectors q = [q1 , . . . , qN ]. The
passage d with tokens [d1 , . . . , dm ], is processed similarly, including a [CLS] and
special [D] token. A linear layer is applied on top of d and q to control the output
dimension, so as to keep the vectors small for storage efficiency, and vectors are
rescaled to unit length, producing the final vector sequences Eq (length N) and Ed
(length m). The ColBERT scoring mechanism is:
N
X m
score(q, d) = max Eqi · Ed j (11.19)
j=1
i=1
While the interaction mechanism has no tunable parameters, the ColBERT ar-
11.2 • I NFORMATION R ETRIEVAL WITH D ENSE V ECTORS 245
s(q,d)
Query Document
Figure 11.8 A sketch of the ColBERT algorithm at inference time. The query and docu-
ment are first passed through separate BERT encoders. Similarity between query and doc-
ument is computed by summing a soft alignment between the contextual representations of
tokens in the query and the document. Training is end-to-end. (Various details aren’t de-
picted; for example the query is prepended by a [CLS] and [Q:] tokens, and the document
by [CLS] and [D:] tokens). Figure adapted from Khattab and Zaharia (2020).
chitecture still needs to be trained end-to-end to fine-tune the BERT encoders and
train the linear layers (and the special [Q] and [D] embeddings) from scratch. It is
trained on triples hq, d + , d − i of query q, positive document d + and negative docu-
ment d − to produce a score for each document using Eq. 11.19, optimizing model
parameters using a cross-entropy loss.
All the supervised algorithms (like ColBERT or the full-interaction version of
the BERT algorithm applied for reranking) need training data in the form of queries
together with relevant and irrelevant passages or documents (positive and negative
examples). There are various semi-supervised ways to get labels; some datasets
(like MS MARCO Ranking, Section 11.4) contain gold positive examples. Negative
examples can be sampled randomly from the top-1000 results from some existing
IR system. If datasets don’t have labeled positive examples, iterative methods like
relevance-guided supervision can be used (Khattab et al., 2021) which rely on the
fact that many datasets contain short answer strings. In this method, an existing IR
system is used to harvest examples that do contain short answer strings (the top few
are taken as positives) or don’t contain short answer strings (the top few are taken as
negatives), these are used to train a new retriever, and then the process is iterated.
Efficiency is an important issue, since every possible document must be ranked
for its similarity to the query. For sparse word-count vectors, the inverted index
allows this very efficiently. For dense vector algorithms finding the set of dense
document vectors that have the highest dot product with a dense query vector is
an instance of the problem of nearest neighbor search. Modern systems there-
Faiss fore make use of approximate nearest neighbor vector search algorithms like Faiss
(Johnson et al., 2017).
246 C HAPTER 11 • R ETRIEVAL - BASED M ODELS
query
Retriever Reader/
Q: When was docs Generator
the premiere of
The Magic Flute? LLM A: 1791
Relevant prompt
Docs
Indexed Docs
Figure 11.9 Retrieval-based question answering has two stages: retrieval, which returns relevant documents
from the collection, and reading, in which an LLM generates answers given the documents as a prompt.
In the first stage of the 2-stage retrieve and read model in Fig. 11.9 we retrieve
relevant passages from a text collection, for example using the dense retrievers of the
previous section. In the second reader stage, we generate the answer via retrieval-
augmented generation. In this method, we take a large pretrained language model,
give it the set of retrieved passages and other text as its prompt, and autoregressively
generate a new answer token by token.
And simple conditional generation for question answering adds a prompt like Q: ,
followed by a query q , and A:, all concatenated:
n
Y
p(x1 , . . . , xn ) = p([Q:] ; q ; [A:] ; x<i )
i=1
11.4 • Q UESTION A NSWERING DATASETS 247
retrieved passage 1
retrieved passage 2
...
retrieved passage n
Or more formally,
n
Y
p(x1 , . . . , xn ) = p(xi |R(q) ; prompt ; [Q:] ; q ; [A:] ; x<i )
i=1
Natural
Questions On the natural side there are datasets like Natural Questions (Kwiatkowski
et al., 2019), a set of anonymized English queries to the Google search engine and
their answers. The answers are created by annotators based on Wikipedia infor-
mation, and include a paragraph-length long answer and a short span answer. For
example the question “When are hops added to the brewing process?” has the short
answer the boiling process and a long answer which is an entire paragraph from the
Wikipedia page on Brewing.
MS MARCO A similar natural question set is the MS MARCO (Microsoft Machine Reading
Comprehension) collection of datasets, including 1 million real anonymized English
questions from Microsoft Bing query logs together with a human generated answer
and 9 million passages (Bajaj et al., 2016), that can be used both to test retrieval
ranking and question answering.
Although many datasets focus on English, natural information-seeking ques-
tion datasets exist in other languages. The DuReader dataset is a Chinese QA
resource based on search engine queries and community QA (He et al., 2018).
TyDi QA TyDi QA dataset contains 204K question-answer pairs from 11 typologically di-
verse languages, including Arabic, Bengali, Kiswahili, Russian, and Thai (Clark
et al., 2020a). In the T Y D I QA task, a system is given a question and the passages
from a Wikipedia article and must (a) select the passage containing the answer (or
N ULL if no passage contains the answer), and (b) mark the minimal answer span (or
N ULL).
MMLU On the probing side are datasets like MMLU (Massive Multitask Language Un-
derstanding), a commonly-used dataset of 15908 knowledge and reasoning ques-
tions in 57 areas including medicine, mathematics, computer science, law, and oth-
ers. MMLU questions are sourced from various exams for humans, such as the US
Graduate Record Exam, Medical Licensing Examination, and Advanced Placement
exams. So the questions don’t represent people’s information needs, but rather are
designed to test human knowledge for academic or licensing purposes. Fig. 11.10
shows some examples, with the correct answers in bold.
Some of the question datasets described above augment each question with pas-
sage(s) from which the answer can be extracted. These datasets were mainly created
reading
comprehension for an earlier QA task called reading comprehension in which a model is given
a question and a document and is required to extract the answer from the given
document. We sometimes call the task of question answering given one or more
open book documents (for example via RAG), the open book QA task, while the task of an-
closed book swering directly from the LM with no retrieval component at all is the closed book
QA task.5 Thus datasets like Natural Questions can be treated as open book if the
solver uses each question’s attached document, or closed book if the documents are
not used, while datasets like MMLU are solely closed book.
Another dimension of variation is the format of the answer: multiple-choice
versus freeform. And of course there are variations in prompting, like whether the
model is just the question (zero-shot) or also given demonstrations of answers to
similar questions (few-shot). MMLU offers both zero-shot and few-shot prompt
options.
5 This repurposes the word for types of exams in which students are allowed to ‘open their books’ or
not.
11.5 • E VALUATING Q UESTION A NSWERING 249
MMLU examples
College Physics
The primary source of the Sun’s energy is a series of thermonuclear
reactions in which the energy produced is c2 times the mass difference
between
(A) two hydrogen atoms and one helium atom
(B) four hydrogen atoms and one helium atom
(C) six hydrogen atoms and two helium atoms
(D) three helium atoms and one carbon atom
International Law
Which of the following is a treaty-based human rights mechanism?
(A) The UN Human Rights Committee
(B) The UN Human Rights Council
(C) The UN Universal Periodic Review
(D) The UN special mandates
Prehistory
Unlike most other early civilizations, Minoan culture shows little evidence
of
(A) trade.
(B) warfare.
(C) the development of a common religion.
(D) conspicuous consumption by elites.
Finally, in some situations QA systems give multiple ranked answers. In such cases
mean
reciprocal rank we evaluated using mean reciprocal rank, or MRR (Voorhees, 1999). MRR is
MRR designed for systems that return a short ranked list of answers or passages for each
test set question, which we can compare against the (human-labeled) correct answer.
First, each test set question is scored with the reciprocal of the rank of the first
correct answer. For example if the system returned five answers to a question but
the first three are wrong (so the highest-ranked correct answer is ranked fourth), the
reciprocal rank for that question is 41 . The score for questions that return no correct
answer is 0. The MRR of a system is the average of the scores for each question in
the test set. In some versions of MRR, questions with a score of zero are ignored
in this calculation. More formally, for a system returning ranked answers to each
question in a test set Q, (or in the alternate version, let Q be the subset of test set
questions that have non-zero scores). MRR is then defined as
|Q|
1 X 1
MRR = (11.20)
|Q| ranki
i=1
11.6 Summary
This chapter introduced the tasks of question answering and information retrieval.
• Question answering (QA) is the task of answering a user’s questions.
• We focus in this chapter on the task of retrieval-based question answering,
in which the user’s questions are intended to be answered by the material in
some set of documents (which might be the web).
• Information Retrieval (IR) is the task of returning documents to a user based
on their information need as expressed in a query. In ranked retrieval, the
documents are returned in ranked order.
• The match between a query and a document can be done by first representing
each of them with a sparse vector that represents the frequencies of words,
weighted by tf-idf or BM25. Then the similarity can be measured by cosine.
• Documents or queries can instead be represented by dense vectors, by encod-
ing the question and document with an encoder-only model like BERT, and in
that case computing similarity in embedding space.
• The inverted index is a storage mechanism that makes it very efficient to find
documents that have a particular word.
• Ranked retrieval is generally evaluated by mean average precision or inter-
polated precision.
• Question answering systems generally use the retriever/reader architecture.
In the retriever stage, an IR system is given a query and returns a set of
documents.
• The reader stage is implemented by retrieval-augmented generation, in
which a large language model is prompted with the query and a set of doc-
uments and then conditionally generates a novel answer.
• QA can be evaluated by exact match with a known answer if only a single
answer is given, with token F1 score for free text answers, or with mean re-
ciprocal rank if a ranked set of answers is given.
H ISTORICAL N OTES 251
Historical Notes
Question answering was one of the earliest NLP tasks. By 1961 the BASEBALL
system (Green et al., 1961) answered questions about baseball games like “Where
did the Red Sox play on July 7” by querying a structured database of game infor-
mation. The database was stored as a kind of attribute-value matrix with values for
attributes of each game:
Month = July
Place = Boston
Day = 7
Game Serial No. = 96
(Team = Red Sox, Score = 5)
(Team = Yankees, Score = 3)
Each question was constituency-parsed using the algorithm of Zellig Harris’s
TDAP project at the University of Pennsylvania, essentially a cascade of finite-state
transducers (see the historical discussion in Joshi and Hopely 1999 and Karttunen
1999). Then in a content analysis phase each word or phrase was associated with a
program that computed parts of its meaning. Thus the phrase ‘Where’ had code to
assign the semantics Place = ?, with the result that the question “Where did the
Red Sox play on July 7” was assigned the meaning
Place = ?
Team = Red Sox
Month = July
Day = 7
The question is then matched against the database to return the answer.
The Protosynthex system of Simmons et al. (1964), given a question, formed a
query from the content words in the question, and then retrieved candidate answer
sentences in the document, ranked by their frequency-weighted term overlap with
the question. The query and each retrieved sentence were then parsed with depen-
dency parsers, and the sentence whose structure best matches the question structure
selected. Thus the question What do worms eat? would match worms eat grass:
both have the subject worms as a dependent of eat, in the version of dependency
grammar used at the time, while birds eat worms has birds as the subject:
Exercises
CHAPTER
12 Machine Translation
“I want to talk the dialect of your people. It’s no use of talking unless
people understand what you say.”
Zora Neale Hurston, Moses, Man of the Mountain 1939, p. 121
machine This chapter introduces machine translation (MT), the use of computers to trans-
translation
MT late from one language to another.
Of course translation, in its full generality, such as the translation of literature, or
poetry, is a difficult, fascinating, and intensely human endeavor, as rich as any other
area of human creativity.
Machine translation in its present form therefore focuses on a number of very
practical tasks. Perhaps the most common current use of machine translation is
information for information access. We might want to translate some instructions on the web,
access
perhaps the recipe for a favorite dish, or the steps for putting together some furniture.
Or we might want to read an article in a newspaper, or get information from an
online resource like Wikipedia or a government webpage in some other language.
MT for information
access is probably
one of the most com-
mon uses of NLP
technology, and Google
Translate alone (shown above) translates hundreds of billions of words a day be-
tween over 100 languages. Improvements in machine translation can thus help re-
digital divide duce what is often called the digital divide in information access: the fact that much
more information is available in English and other languages spoken in wealthy
countries. Web searches in English return much more information than searches in
other languages, and online resources like Wikipedia are much larger in English and
other higher-resourced languages. High-quality translation can help provide infor-
mation to speakers of lower-resourced languages.
Another common use of machine translation is to aid human translators. MT sys-
post-editing tems are routinely used to produce a draft translation that is fixed up in a post-editing
phase by a human translator. This task is often called computer-aided translation
CAT or CAT. CAT is commonly used as part of localization: the task of adapting content
localization or a product to a particular language community.
Finally, a more recent application of MT is to in-the-moment human commu-
nication needs. This includes incremental translation, translating speech on-the-fly
before the entire sentence is complete, as is commonly used in simultaneous inter-
pretation. Image-centric translation can be used for example to use OCR of the text
on a phone camera image as input to an MT system to translate menus or street signs.
encoder- The standard algorithm for MT is the encoder-decoder network/ We briefly
decoder
mentioned in Chapter 7 that encoder-decoder or sequence-to-sequence models are
used for tasks in which we need to map an input sequence to an output sequence
that is a complex function of the entire input sequence, like machine translation or
254 C HAPTER 12 • M ACHINE T RANSLATION
speech recognition. Indeed, in machine translation, the words of the target language
don’t necessarily agree with the words of the source language in number or order.
Consider translating the following made-up English sentence into Japanese.
(12.1) English: He wrote a letter to a friend
Japanese: tomodachi ni tegami-o kaita
friend to letter wrote
Note that the elements of the sentences are in very different places in the different
languages. In English, the verb is in the middle of the sentence, while in Japanese,
the verb kaita comes at the end. The Japanese sentence doesn’t require the pronoun
he, while English does.
Such differences between languages can be quite complex. In the following ac-
tual sentence from the United Nations, notice the many changes between the Chinese
sentence (we’ve given in red a word-by-word gloss of the Chinese characters) and
its English equivalent produced by human translators.
(12.2) 大会/General Assembly 在/on 1982年/1982 12月/December 10日/10 通过
了/adopted 第37号/37th 决议/resolution ,核准了/approved 第二
次/second 探索/exploration 及/and 和平peaceful 利用/using 外层空
间/outer space 会议/conference 的/of 各项/various 建议/suggestions 。
On 10 December 1982 , the General Assembly adopted resolution 37 in
which it endorsed the recommendations of the Second United Nations
Conference on the Exploration and Peaceful Uses of Outer Space .
Note the many ways the English and Chinese differ. For example the order-
ing differs in major ways; the Chinese order of the noun phrase is “peaceful using
outer space conference of suggestions” while the English has “suggestions of the ...
conference on peaceful use of outer space”). And the order differs in minor ways
(the date is ordered differently). English requires the in many places that Chinese
doesn’t, and adds some details (like “in which” and “it”) that aren’t necessary in
Chinese. Chinese doesn’t grammatically mark plurality on nouns (unlike English,
which has the “-s” in “recommendations”), and so the Chinese must use the modi-
fier 各项/various to make it clear that there is not just one recommendation. English
capitalizes some words but not others. Encoder-decoder networks are very success-
ful at handling these sorts of complicated cases of sequence mappings.
We’ll begin in the next section by considering the linguistic background about
how languages vary, and the implications this variance has for the task of MT. Then
we’ll sketch out the standard algorithm, give details about things like input tokeniza-
tion and creating training corpora of parallel sentences, give some more low-level
details about the encoder-decoder network, and finally discuss how MT is evaluated,
introducing the simple chrF metric.
ways to ask questions, or issue commands, has linguistic mechanisms for indicating
agreement or disagreement.
Yet languages also differ in many ways (as has been pointed out since ancient
translation
divergence times; see Fig. 12.1). Understanding what causes such translation divergences
(Dorr, 1994) can help us build better MT models. We often distinguish the idiosyn-
cratic and lexical differences that must be dealt with one by one (the word for “dog”
differs wildly from language to language), from systematic differences that we can
model in a general way (many languages put the verb before the grammatical ob-
ject; others put the verb after the grammatical object). The study of these systematic
typology cross-linguistic similarities and differences is called linguistic typology. This sec-
tion sketches some typological facts that impact machine translation; the interested
reader should also look into WALS, the World Atlas of Language Structures, which
gives many typological facts about languages (Dryer and Haspelmath, 2013).
Figure 12.1 The Tower of Babel, Pieter Bruegel 1563. Wikimedia Commons, from the
Kunsthistorisches Museum, Vienna.
(a) (b)
Figure 12.2 Examples of other word order differences: (a) In German, adverbs occur in
initial position that in English are more natural later, and tensed verbs occur in second posi-
tion. (b) In Mandarin, preposition phrases expressing goals often occur pre-verbally, unlike
in English.
Fig. 12.2 shows examples of other word order differences. All of these word
order differences between languages can cause problems for translation, requiring
the system to do huge structural reorderings as it generates the output.
mappings. For example, Fig. 12.3 summarizes some of the complexities discussed
by Hutchins and Somers (1992) in translating English leg, foot, and paw, to French.
For example, when leg is used about an animal it’s translated as French patte; but
about the leg of a journey, as French etape; if the leg is of a chair, we use French
pied.
lexical gap Further, one language may have a lexical gap, where no word or phrase, short
of an explanatory footnote, can express the exact meaning of a word in the other
language. For example, English does not have a word that corresponds neatly to
Mandarin xiào or Japanese oyakōkō (in English one has to make do with awkward
phrases like filial piety or loving child, or good son/daughter for both).
ANIMAL paw
etape
JOURNEY ANIMAL
patte
BIRD
leg foot
HUMAN CHAIR HUMAN
jambe pied
Figure 12.3 The complex overlap between English leg, foot, etc., and various French trans-
lations as discussed by Hutchins and Somers (1992).
may conflate multiple morphemes, like -om in the word stolom (table-SG-INSTR-
DECL 1), which fuses the distinct morphological categories instrumental, singular,
and first declension.
Translating between languages with rich morphology requires dealing with struc-
ture below the word level, and for this reason modern systems generally use subword
models like the wordpiece or BPE models of Section 12.2.1.
Rather than use the input tokens directly, the encoder-decoder architecture con-
sists of two components, an encoder and a decoder. The encoder takes the input
words x = [x1 , . . . , xn ] and produces an intermediate context h. At decoding time, the
system takes h and, word by word, generates the output y:
h = encoder(x) (12.8)
yt+1 = decoder(h, y1 , . . . , yt ) ∀t ∈ [1, . . . , m] (12.9)
In the next two sections we’ll talk about subword tokenization, and then how to get
parallel corpora for training, and then we’ll introduce the details of the encoder-
decoder architecture.
12.2.1 Tokenization
Machine translation systems use a vocabulary that is fixed in advance, and rather
than using space-separated words, this vocabulary is generated with subword to-
kenization algorithms, like the BPE algorithm sketched in Chapter 2. A shared
vocabulary is used for the source and target languages, which makes it easy to copy
tokens (like names) from source to target. Using subword tokenization with tokens
shared between languages makes it natural to translate between languages like En-
glish or Hindi that use spaces to separate words, and languages like Chinese or Thai
that don’t.
We build the vocabulary by running a subword tokenization algorithm on a cor-
pus that contains both source and target language data.
Rather than the simple BPE algorithm from Fig. 2.6, modern systems often use
more powerful tokenization algorithms. Some systems (like BERT) use a variant of
wordpiece BPE called the wordpiece algorithm, which instead of choosing the most frequent
set of tokens to merge, chooses merges based on which one most increases the lan-
guage model probability of the tokenization. Wordpieces use a special symbol at the
beginning of each token; here’s a resulting tokenization from the Google MT system
(Wu et al., 2016):
words: Jet makers feud over seat width with big orders at stake
wordpieces: J et makers fe ud over seat width with big orders at stake
The wordpiece algorithm is given a training corpus and a desired vocabulary size
V, and proceeds as follows:
1. Initialize the wordpiece lexicon with characters (for example a subset of Uni-
code characters, collapsing all the remaining characters to a special unknown
character token).
260 C HAPTER 12 • M ACHINE T RANSLATION
Sentence alignment
Standard training corpora for MT come as aligned pairs of sentences. When creat-
ing new corpora, for example for underresourced languages or new domains, these
sentence alignments must be created. Fig. 12.4 gives a sample hypothetical sentence
alignment.
E1: “Good morning," said the little prince. F1: -Bonjour, dit le petit prince.
E2: “Good morning," said the merchant. F2: -Bonjour, dit le marchand de pilules perfectionnées qui
apaisent la soif.
E3: This was a merchant who sold pills that had
F3: On en avale une par semaine et l'on n'éprouve plus le
been perfected to quench thirst.
besoin de boire.
E4: You just swallow one pill a week and you F4: -C’est une grosse économie de temps, dit le marchand.
won’t feel the need for anything to drink.
E5: “They save a huge amount of time," said the merchant. F5: Les experts ont fait des calculs.
E6: “Fifty−three minutes a week." F6: On épargne cinquante-trois minutes par semaine.
E7: “If I had fifty−three minutes to spend?" said the F7: “Moi, se dit le petit prince, si j'avais cinquante-trois minutes
little prince to himself. à dépenser, je marcherais tout doucement vers une fontaine..."
E8: “I would take a stroll to a spring of fresh water”
Figure 12.4 A sample alignment between sentences in English and French, with sentences extracted from
Antoine de Saint-Exupery’s Le Petit Prince and a hypothetical translation. Sentence alignment takes sentences
e1 , ..., en , and f1 , ..., fm and finds minimal sets of sentences that are translations of each other, including single
sentence mappings like (e1 ,f1 ), (e4 ,f3 ), (e5 ,f4 ), (e6 ,f6 ) as well as 2-1 alignments (e2 /e3 ,f2 ), (e7 /e8 ,f7 ), and null
alignments (f5 ).
Given two documents that are translations of each other, we generally need two
steps to produce sentence alignments:
• a cost function that takes a span of source sentences and a span of target sen-
tences and returns a score measuring how likely these spans are to be transla-
tions.
• an alignment algorithm that takes these scores to find a good alignment be-
tween the documents.
To score the similarity of sentences across languages, we need to make use of
a multilingual embedding space, in which sentences from different languages are
in the same embedding space (Artetxe and Schwenk, 2019). Given such a space,
cosine similarity of such embeddings provides a natural scoring function (Schwenk,
2018). Thompson and Koehn (2019) give the following cost function between two
sentences or spans x,y from the source and target documents respectively:
(1 − cos(x, y))nSents(x) nSents(y)
c(x, y) = PS PS (12.10)
s=1 1 − cos(x, ys ) + s=1 1 − cos(xs , y)
where nSents() gives the number of sentences (this biases the metric toward many
alignments of single sentences instead of aligning very large spans). The denom-
inator helps to normalize the similarities, and so x1 , ..., xS , y1 , ..., yS , are randomly
selected sentences sampled from the respective documents.
Usually dynamic programming is used as the alignment algorithm (Gale and
Church, 1993), in a simple extension of the minimum edit distance algorithm we
introduced in Chapter 2.
Finally, it’s helpful to do some corpus cleanup by removing noisy sentence pairs.
This can involve handwritten rules to remove low-precision pairs (for example re-
moving sentences that are too long, too short, have different URLs, or even pairs
262 C HAPTER 12 • M ACHINE T RANSLATION
that are too similar, suggesting that they were copies rather than translations). Or
pairs can be ranked by their multilingual embedding cosine score and low-scoring
pairs discarded.
Decoder
transformer
blocks
Figure 12.5 The encoder-decoder transformer architecture for machine translation. The encoder uses the
transformer blocks we saw in Chapter 8, while the decoder uses a more powerful block with an extra cross-
attention layer that can attend to all the encoder words. We’ll see this in more detail in the next section.
y1 y2 yi+1 ym
… Language
Modeling
Henc h1 h2 hi hn Head
… …
Unembedding Matrix
Block K Block L
… … … … … …
Block 2
Block 2
+ +
Feedforward Feedforward
Encoder
Block 1 Layer Normalize
Layer Normalize
+
+
Cross-Attention
Multi-Head Attention Decoder
Layer Normalize Block 1
Layer Normalize +
Causal (Left-to-Right)
x1 x2 … xi … xn
Multi-Head Attention
<> y1 … yi … ym
…
Decoder
Figure 12.6 The transformer block for the encoder and the decoder, showing the residual stream view. The
final output of the encoder Henc = h1 , ..., hn is the context used in the decoder. The decoder is a standard
transformer except with one extra layer, the cross-attention layer, which takes that encoder output Henc and
uses it to form its K and V inputs.
That is, where in standard multi-head attention the input to each attention layer is
X, in cross attention the input is the the final output of the encoder Henc = h1 , ..., hn .
Henc is of shape [n × d], each row representing one input token. To link the keys
and values from the encoder with the query from the prior layer of the decoder, we
multiply the encoder output Henc by the cross-attention layer’s key weights WK and
value weights WV . The query comes from the output from the prior decoder layer
Hdec[`−1] , which is multiplied by the cross-attention layer’s query weights WQ :
QK|
CrossAttention(Q, K, V) = softmax √ V (12.12)
dk
The cross attention thus allows the decoder to attend to each of the source language
words as projected into the entire encoder final output representations. The other
attention layer in each decoder block, the multi-head attention layer, is the same
causal (left-to-right) attention that we saw in Chapter 8. The multi-head attention in
the encoder, however, is allowed to look ahead at the entire source language text, so
it is not masked.
To train an encoder-decoder model, we use the same self-supervision model we
used for training encoder-decoders RNNs in Chapter 13. The network is given the
source text and then starting with the separator token is trained autoregressively to
predict the next token using cross-entropy loss. Recall that cross-entropy loss for
language modeling is determined by the probability the model assigns to the correct
264 C HAPTER 12 • M ACHINE T RANSLATION
next word. So at time t the CE loss is the negative log probability the model assigns
to the next word in the training sequence:
teacher forcing As in that case, we use teacher forcing in the decoder. Recall that in teacher forc-
ing, at each time step in decoding we force the system to use the gold target token
from training as the next input xt+1 , rather than allowing it to rely on the (possibly
erroneous) decoder output yˆt .
A problem with greedy decoding is that what looks high probability at word t might
turn out to have been the wrong choice once we get to word t + 1. The beam search
algorithm maintains multiple choices until later when we can see which one is best.
In beam search we model decoding as searching the space of possible genera-
search tree tions, represented as a search tree whose branches represent actions (generating a
token), and nodes represent states (having generated a particular prefix). We search
for the best action sequence, i.e., the string with the highest probability.
p(t3| t1,t2)
p(t2| t1)
ok 1.0 EOS
.7
yes 1.0 EOS
p(t1|start) .2
ok .1 EOS
.4
start .5 yes .3 ok 1.0 EOS
.1 .4
EOS yes 1.0 EOS
.3
EOS
t1 t2 t3
Figure 12.7 A search tree for generating the target string T = t1 ,t2 , ... from vocabulary
V = {yes, ok, <s>}, showing the probability of generating each token from that state. Greedy
search chooses yes followed by yes, instead of the globally most probable sequence ok ok.
arrived y2
the green y3
hd1 hd2 y2 y3
y1
a a
y1 hd 1 hd2 hd 2
BOS arrived … …
aardvark BOS the green mage
a .. ..
hd1
… the the
aardvark .. ..
witch witch
BOS .. … …
start arrived zebra zebra
..
the
y2 y3
…
zebra a arrived
… …
aardvark aardvark
the y2 .. ..
green green
.. ..
witch who
hd1 hd2 … y3 …
the witch
zebra zebra
BOS the
hd1 hd2 hd2
Figure 12.8 Beam search decoding with a beam width of k = 2. At each time step, we choose the k best
hypotheses, form the V possible extensions of each, score those k × V hypotheses and choose the best k = 2
to continue. At time 1, the frontier has the best 2 options from the initial decoder state: arrived and the. We
extend each, compute the probability of all the hypotheses so far (arrived the, arrived aardvark, the green, the
witch) and again chose the best 2 (the green and the witch) to be the search frontier. The images on the arcs
schematically represent the decoders that must be run at each step to score the next words (for simplicity not
depicting cross-attention).
by being passed to distinct decoders, which each generate a softmax over the entire
vocabulary to extend the hypothesis to every possible next token. Each of these k ×V
hypotheses is scored by P(yi |x, y<i ): the product of the probability of the current
word choice multiplied by the probability of the path that led to it. We then prune
the k ×V hypotheses down to the k best hypotheses, so there are never more than k
hypotheses at the frontier of the search, and never more than k decoders. Fig. 12.8
illustrates this with a beam width of 2 for the beginning of The green witch arrived.
This process continues until an EOS is generated indicating that a complete can-
didate output has been found. At this point, the completed hypothesis is removed
from the frontier and the size of the beam is reduced by one. The search continues
until the beam has been reduced to 0. The result will be k hypotheses.
To score each node by its log probability, we use the chain rule of probability to
break down p(y|x) into the product of the probability of each word given its prior
context, which we can turn into a sum of logs (for an output string of length t):
Thus at each step, to compute the probability of a partial sentence, we simply add the
log probability of the prefix sentence so far to the log probability of generating the
next token. Fig. 12.9 shows the scoring for the example sentence shown in Fig. 12.8,
using some simple made-up probabilities. Log probabilities are negative or 0, and
the max of two log probabilities is the one that is greater (closer to 0).
Figure 12.9 Scoring for beam search decoding with a beam width of k = 2. We maintain the log probability
of each hypothesis in the beam by incrementally adding the logprob of generating each next token. Only the top
k paths are extended to the next step.
Fig. 12.10 gives the algorithm. One problem with this version of the algorithm is
that the completed hypotheses may have different lengths. Because language mod-
12.4 • D ECODING IN MT: B EAM S EARCH 267
y0 , h0 ← 0
path ← ()
complete paths ← ()
state ← (c, y0 , h0 , path) ;initial state
frontier ← hstatei ;initial frontier
els generally assign lower probabilities to longer strings, a naive algorithm would
choose shorter strings for y. (This is not an issue during the earlier steps of decod-
ing; since beam search is breadth-first, all the hypotheses being compared had the
same length.) For this reason we often apply length normalization methods, like
dividing the logprob by the number of words:
t
1 1X
score(y) = log P(y|x) = log P(yi |y1 , ..., yi−1 , x) (12.16)
t t
i=1
For MT we generally use beam widths k between 5 and 10, giving us k hypotheses at
the end. We can pass all k to the downstream application with their respective scores,
or if we just need a single translation we can pass the most probable hypothesis.
can work even better than beam search and also tends to be better than the other
decoding algorithms like temperature sampling introduced in Section 7.4.
The intuition of minimum Bayes risk is that instead of trying to choose the trans-
lation which is most probable, we choose the one that is likely to have the least error.
For example, we might want our decoding algorithm to find the translation which
has the highest score on some evaluation metric. For example in Section 12.6 we will
introduce metrics like chrF or BERTScore that measure the goodness-of-fit between
a candidate translation and a set of reference human translations. A translation that
maximizes this score, especially with a hypothetically huge set of perfect human
translations is likely to be a good one (have minimum risk) even if it is not the most
probable translation by our particular probability estimator.
In practice, we don’t know the perfect set of translations for a given sentence. So
the standard simplification used in MBR decoding algorithms is to instead choose
the candidate translation which is most similar (by some measure of goodness-of-
fit) with some set of candidate translations. We’re essentially approximating the
enormous space of all possible translations U with a smaller set of possible candidate
translations Y.
Given this set of possible candidate translations Y, and some similarity or align-
ment function util, we choose the best translation ŷ as the translation which is most
similar to all the other candidate translations:
X
ŷ = argmax util(y, c) (12.17)
y∈Y c∈Y
Various util functions can be used, like chrF or BERTscore or BLEU. We can get the
set of candidate translations by sampling using one of the basic sampling algorithms
of Section 7.4 like temperature sampling; good results can be obtained with as few
as 32 or 64 candidates.
Minimum Bayes risk decoding can also be used for other NLP tasks; indeed
it was widely applied to speech recognition (Stolcke et al., 1997; Goel and Byrne,
2000) before being applied to machine translation (Kumar and Byrne, 2004), and
has been shown to work well across many other generation tasks as well (e.g., sum-
marization, dialogue, and image captioning (Suzgun et al., 2023a)).
h = encoder(x, ls ) (12.18)
yi+1 = decoder(h, lt , y1 , . . . , yi ) ∀i ∈ [1, . . . , m] (12.19)
One advantage of a multilingual model is that they can improve the translation
of lower-resourced languages by drawing on information from a similar language
in the training data that happens to have more resources. Perhaps we don’t know
the meaning of a word in Galician, but the word appears in the similar and higher-
resourced language Spanish.
270 C HAPTER 12 • M ACHINE T RANSLATION
12.6 MT Evaluation
Translations are evaluated along two dimensions:
adequacy 1. adequacy: how well the translation captures the exact meaning of the source
sentence. Sometimes called faithfulness or fidelity.
fluency 2. fluency: how fluent the translation is in the target language (is it grammatical,
clear, readable, natural).
Using humans to evaluate is most accurate, but automatic metrics are also used for
convenience.
We can do the same thing to judge the second dimension, adequacy, using raters
to assign scores on a scale. If we have bilingual raters, we can give them the source
sentence and a proposed target sentence, and rate, on a 5-point or 100-point scale,
how much of the information in the source was preserved in the target. If we only
have monolingual raters but we have a good human translation of the source text, we
can give the monolingual raters the human reference translation and a target machine
translation and again rate how much information is preserved. An alternative is to
ranking do ranking: give the raters a pair of candidate translations, and ask them which one
they prefer.
Training of human raters (who are often online crowdworkers) is essential; raters
without translation expertise find it difficult to separate fluency and adequacy, and
so training includes examples carefully distinguishing these. Raters often disagree
(source sentences may be ambiguous, raters will have different world knowledge,
raters may apply scales differently). It is therefore common to remove outlier raters,
and (if we use a fine-grained enough scale) normalizing raters by subtracting the
mean from their scores and dividing by the variance.
As discussed above, an alternative way of using human raters is to have them
post-edit translations, taking the MT output and changing it minimally until they
feel it represents a correct translation. The difference between their post-edited
translations and the original MT output can then be used as a measure of quality.
We use that to compute the unigram and bigram precisions and recalls:
unigram P: 17/17 = 1 unigram R: 17/18 = .944
bigram P: 13/16 = .813 bigram R: 13/17 = .765
Finally we average to get chrP and chrR, and compute the F-score:
in the candidate translation that also occur in the reference translation, and ditto for
bigrams and so on, up to 4-grams. BLEU extends this unigram metric to the whole
corpus by computing the numerator as the sum over all sentences of the counts of all
the unigram types that also occur in the reference translation, and the denominator
is the total of the counts of all unigrams in all candidate sentences. We compute
this n-gram precision for unigrams, bigrams, trigrams, and 4-grams and take the
geometric mean. BLEU has many further complications, including a brevity penalty
for penalizing candidate translations that are too short, and it also requires the n-
gram counts be clipped in a particular way.
Because BLEU is a word-based metric, it is very sensitive to word tokenization,
making it impossible to compare different systems if they rely on different tokeniza-
tion standards, and doesn’t work as well in languages with complex morphology.
Nonetheless, you will sometimes still see systems evaluated by BLEU, particularly
for translation into English. In such cases it’s important to use packages that enforce
standardization for tokenization like S ACRE BLEU (Post, 2018).
chrF: Limitations
While automatic character and word-overlap metrics like chrF or BLEU are useful,
they have important limitations. chrF is very local: a large phrase that is moved
around might barely change the chrF score at all, and chrF can’t evaluate cross-
sentence properties of a document like its discourse coherence (Chapter 24). chrF
and similar automatic metrics also do poorly at comparing very different kinds of
systems, such as comparing human-aided translation against machine translation, or
different machine translation architectures against each other (Callison-Burch et al.,
2006). Instead, automatic overlap metrics like chrF are most appropriate when eval-
uating changes to a single system.
7.94
the weather is
Reference
1.82
cold today (0.713 1.27)+(0.515 7.94)+...
7.90
RBERT =
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sha1_base64="OJyoKlmBAgUA0KDtUcsH/di5BlI=">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</latexit>
sha1_base64="RInTcZkWiVBnf/ncBstCvatCtG4=">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</latexit>
1.27+7.94+1.82+7.90+8.88
Candidate
Figure12.11
Figure 1: Illustration of the computation
The computation of the
of BERTS CORE recall
recall metric
from RBERT
reference . Given
x and the x̂,
candidate reference x and
from Figure 1 in
candidate x̂, we compute BERT embeddings and pairwise cosine similarity. We highlight the greedy
Zhang et al. (2020). This version shows an extended version of the metric in which tokens are also weighted by
matching
their in red, and include the optional idf importance weighting.
idf values.
We experiment with different models (Section 4), using the tokenizer provided with each model.
12.7 Bias and
Given a tokenized Ethical
reference sentence xIssues
= hx , . . . , x i, the embedding model generates a se- 1 k
quence of vectors hx1 , . . . , xk i. Similarly, the tokenized candidate x̂ = hx̂1 , . . . , x̂m i is mapped
to hx̂1 , . . . , x̂l i. The main model we use is BERT, which tokenizes the input text into a sequence
of word pieces Machine (Wu et al.,translation raises
2016), where many ofwords
unknown the same ethical
are split intoissues
severalthat we’ve discussed
commonly observed in
sequences of characters. The representation
earlier chapters. For example, for consider
each wordMT piece is computed
systems with from
translating a Transformer
Hungarian
encoder (Vaswani et al.,has
(which 2017) by repeatedly
the gender neutralapplying
pronounself-attention
ő) or Spanishand nonlinear
(which oftentransformations
drops pronouns)
in an alternatinginto fashion.
English BERT embeddings
(in which pronouns haveare
been shown toand
obligatory, benefit
theyvarious NLP tasks (Devlin
have grammatical gender).
et al., 2019; Liu, 2019; Huang et al., 2019; Yang et al., 2019a).
When translating a reference to a person described without specified gender, MT
Similarity Measure systemsThe often default
vector to male gender
representation allows (Schiebinger 2014, Prates
for a soft measure et al. 2019).
of similarity instead And
of
exact-string (Papineni MT systemset al.,often
2002)assign
or heuristic
gender (Banerjee
according & to Lavie,
culture2005) matching.
stereotypes of theThe
sortcosine
we saw
x>
similarity of a in Sectiontoken
reference 5.8. Fig. 12.12
xi and shows examples
a candidate token x̂from Prates
j is kx
i x̂j et al. (2019), in which Hun-
i kkx̂j k
. We use pre-normalized
garian gender-neutral ő is a nurse is translated> with she, but gender-neutral ő is a
vectors, which reduces this calculation to the inner product xi x̂j . While this measure considers
tokens in isolation, CEOthe is contextual
translated with he. Prates
embeddings et [Link]
contain (2019) find from
that these stereotypes
the rest can’t com-
of the sentence.
pletely be accounted for by gender bias in US labor statistics, because the biases are
BERTS CORE The complete score matches each token in x to a token in x̂ to compute recall,
and each token in x̂ to a token in x to compute precision. We use greedy matching to maximize
the matching similarity score,2 where each token is matched to the most similar token in the other
sentence. We combine precision and recall to compute an F1 measure. For a reference x and
candidate x̂, the recall, precision, and F1 scores are:
1 X > 1 X > PBERT · RBERT
12.8 • S UMMARY 275
Similarly, a recent challenge set, the WinoMT dataset (Stanovsky et al., 2019)
shows that MT systems perform worse when they are asked to translate sentences
that describe people with non-stereotypical gender roles, like “The doctor asked the
nurse to help her in the operation”.
Many ethical questions in MT require further research. One open problem is
developing metrics for knowing what our systems don’t know. This is because MT
systems can be used in urgent situations where human translators may be unavailable
or delayed: in medical domains, to help translate when patients and doctors don’t
speak the same language, or in legal domains, to help judges or lawyers communi-
cate with witnesses or defendants. In order to ‘do no harm’, systems need ways to
confidence assign confidence values to candidate translations, so they can abstain from giving
incorrect translations that may cause harm.
12.8 Summary
Machine translation is one of the most widely used applications of NLP, and the
encoder-decoder model, first developed for MT is a key tool that has applications
throughout NLP.
• Languages have divergences, both structural and lexical, that make translation
difficult.
• The linguistic field of typology investigates some of these differences; lan-
guages can be classified by their position along typological dimensions like
whether verbs precede their objects.
• Encoder-decoder networks (for transformers just as we saw in Chapter 13
for RNNs) are composed of an encoder network that takes an input sequence
and creates a contextualized representation of it, the context. This context
representation is then passed to a decoder which generates a task-specific
output sequence.
• Cross-attention allows the transformer decoder to view information from all
the hidden states of the encoder.
• Machine translation models are trained on a parallel corpus, sometimes called
a bitext, a text that appears in two (or more) languages.
276 C HAPTER 12 • M ACHINE T RANSLATION
Historical Notes
MT was proposed seriously by the late 1940s, soon after the birth of the computer
(Weaver, 1949/1955). In 1954, the first public demonstration of an MT system pro-
totype (Dostert, 1955) led to great excitement in the press (Hutchins, 1997). The
next decade saw a great flowering of ideas, prefiguring most subsequent develop-
ments. But this work was ahead of its time—implementations were limited by, for
example, the fact that pending the development of disks there was no good way to
store dictionary information.
As high-quality MT proved elusive (Bar-Hillel, 1960), there grew a consensus
on the need for better evaluation and more basic research in the new fields of for-
mal and computational linguistics. This consensus culminated in the famously crit-
ical ALPAC (Automatic Language Processing Advisory Committee) report of 1966
(Pierce et al., 1966) that led in the mid 1960s to a dramatic cut in funding for MT
in the US. As MT research lost academic respectability, the Association for Ma-
chine Translation and Computational Linguistics dropped MT from its name. Some
MT developers, however, persevered, and there were early MT systems like Météo,
which translated weather forecasts from English to French (Chandioux, 1976), and
industrial systems like Systran.
In the early years, the space of MT architectures spanned three general mod-
els. In direct translation, the system proceeds word-by-word through the source-
language text, translating each word incrementally. Direct translation uses a large
bilingual dictionary, each of whose entries is a small program with the job of trans-
lating one word. In transfer approaches, we first parse the input text and then ap-
ply rules to transform the source-language parse into a target language parse. We
then generate the target language sentence from the parse tree. In interlingua ap-
proaches, we analyze the source language text into some abstract meaning repre-
sentation, called an interlingua. We then generate into the target language from
this interlingual representation. A common way to visualize these three early ap-
Vauquois
triangle proaches was the Vauquois triangle shown in Fig. 12.13. The triangle shows the
increasing depth of analysis required (on both the analysis and generation end) as
we move from the direct approach through transfer approaches to interlingual ap-
proaches. In addition, it shows the decreasing amount of transfer knowledge needed
as we move up the triangle, from huge amounts of transfer at the direct level (al-
most all knowledge is transfer knowledge for each word) through transfer (transfer
rules only for parse trees or thematic roles) through interlingua (no specific transfer
knowledge). We can view the encoder-decoder network as an interlingual approach,
with attention acting as an integration of direct and transfer, allowing words or their
representations to be directly accessed by the decoder.
H ISTORICAL N OTES 277
Interlingua
sis age
ta gen
aly gu
rg
an la n
et era
Source Text:
Target Text:
lan tion
ce
Semantic/Syntactic
Transfer Semantic/Syntactic
ur
gu
Structure
so
ag
Structure
e
source Direct Translation target
text text
Figure 12.13 The Vauquois (1968) triangle.
Statistical methods began to be applied around 1990, enabled first by the devel-
opment of large bilingual corpora like the Hansard corpus of the proceedings of the
Canadian Parliament, which are kept in both French and English, and then by the
growth of the web. Early on, a number of researchers showed that it was possible
to extract pairs of aligned sentences from bilingual corpora, using words or simple
cues like sentence length (Kay and Röscheisen 1988, Gale and Church 1991, Gale
and Church 1993, Kay and Röscheisen 1993).
At the same time, the IBM group, drawing directly on the noisy channel model
statistical MT for speech recognition, proposed two related paradigms for statistical MT. These
IBM Models include the generative algorithms that became known as IBM Models 1 through
Candide 5, implemented in the Candide system. The algorithms (except for the decoder)
were published in full detail— encouraged by the US government who had par-
tially funded the work— which gave them a huge impact on the research community
(Brown et al. 1990, Brown et al. 1993).
The group also developed a discriminative approach, called MaxEnt (for maxi-
mum entropy, an alternative formulation of logistic regression), which allowed many
features to be combined discriminatively rather than generatively (Berger et al.,
1996), which was further developed by Och and Ney (2002).
By the turn of the century, most academic research on machine translation used
statistical MT, either in the generative or discriminative mode. An extended version
phrase-based of the generative approach, called phrase-based translation was developed, based
translation
on inducing translations for phrase-pairs (Och 1998, Marcu and Wong 2002, Koehn
et al. (2003), Och and Ney 2004, Deng and Byrne 2005, inter alia).
Once automatic metrics like BLEU were developed (Papineni et al., 2002), the
discriminative log linear formulation (Och and Ney, 2004), drawing from the IBM
MaxEnt work (Berger et al., 1996), was used to directly optimize evaluation metrics
MERT like BLEU in a method known as Minimum Error Rate Training, or MERT (Och,
2003), also drawing from speech recognition models (Chou et al., 1993). Toolkits
Moses like GIZA (Och and Ney, 2003) and Moses (Koehn et al. 2006, Zens and Ney 2007)
were widely used.
There were also approaches around the turn of the century that were based on
transduction
grammars syntactic structure (Chapter 18). Models based on transduction grammars (also
called synchronous grammars) assign a parallel syntactic tree structure to a pair
of sentences in different languages, with the goal of translating the sentences by
applying reordering operations on the trees. From a generative perspective, we can
view a transduction grammar as generating pairs of aligned sentences in two lan-
inversion
guages. Some of the most widely used models included the inversion transduction
transduction
grammar
grammar (Wu, 1996) and synchronous context-free grammars (Chiang, 2005),
278 C HAPTER 12 • M ACHINE T RANSLATION
Neural networks had been applied at various times to various aspects of machine
translation; for example Schwenk et al. (2006) showed how to use neural language
models to replace n-gram language models in a Spanish-English system based on
IBM Model 4. The modern neural encoder-decoder approach was pioneered by
Kalchbrenner and Blunsom (2013), who used a CNN encoder and an RNN decoder,
and was first applied to MT by Bahdanau et al. (2015). The transformer encoder-
decoder was proposed by Vaswani et al. (2017) (see the History section of Chap-
ter 8).
Research on evaluation of machine translation began quite early. Miller and
Beebe-Center (1956) proposed a number of methods drawing on work in psycholin-
guistics. These included the use of cloze and Shannon tasks to measure intelligibility
as well as a metric of edit distance from a human translation, the intuition that un-
derlies all modern overlap-based automatic evaluation metrics. The ALPAC report
included an early evaluation study conducted by John Carroll that was extremely in-
fluential (Pierce et al., 1966, Appendix 10). Carroll proposed distinct measures for
fidelity and intelligibility, and had raters score them subjectively on 9-point scales.
Much early evaluation work focuses on automatic word-overlap metrics like BLEU
(Papineni et al., 2002), NIST (Doddington, 2002), TER (Translation Error Rate)
(Snover et al., 2006), Precision and Recall (Turian et al., 2003), and METEOR
(Banerjee and Lavie, 2005); character n-gram overlap methods like chrF (Popović,
2015) came later. More recent evaluation work, echoing the ALPAC report, has
emphasized the importance of careful statistical methodology and the use of human
evaluation (Kocmi et al., 2021; Marie et al., 2021).
The early history of MT is surveyed in Hutchins 1986 and 1997; Nirenburg et al.
(2002) collects early readings. See Croft (1990) or Comrie (1989) for introductions
to linguistic typology.
Exercises
12.1 Compute by hand the chrF2,2 score for HYP2 on page 272 (the answer should
round to .62).
CHAPTER
xt ht yt
Figure 13.1 Simple recurrent neural network after Elman (1990). The hidden layer in-
cludes a recurrent connection as part of its input. That is, the activation value of the hidden
layer depends on the current input as well as the activation value of the hidden layer from the
previous time step.
subscripts to represent time, thus xt will mean the input vector x at time t. The key
difference from a feedforward network lies in the recurrent link shown in the figure
with the dashed line. This link augments the input to the computation at the hidden
layer with the value of the hidden layer from the preceding point in time.
The hidden layer from the previous time step provides a form of memory, or
context, that encodes earlier processing and informs the decisions to be made at
later points in time. Critically, this approach does not impose a fixed-length limit
on this prior context; the context embodied in the previous hidden layer can include
information extending back to the beginning of the sequence.
Adding this temporal dimension makes RNNs appear to be more complex than
non-recurrent architectures. But in reality, they’re not all that different. Given an
input vector and the values for the hidden layer from the previous time step, we’re
still performing the standard feedforward calculation introduced in Chapter 6. To
see this, consider Fig. 13.2 which clarifies the nature of the recurrence and how it
factors into the computation at the hidden layer. The most significant change lies in
the new set of weights, U, that connect the hidden layer from the previous time step
to the current hidden layer. These weights determine how the network makes use of
past context in calculating the output for the current input. As with the other weights
in the network, these connections are trained via backpropagation.
yt
ht
U W
ht-1 xt
Figure 13.2 Simple recurrent neural network illustrated as a feedforward network. The
hidden layer ht−1 from the prior time step is multiplied by weight matrix U and then added
to the feedforward component from the current time step.
the hidden layer from the previous time step ht−1 with the weight matrix U. We
add these values together and pass them through a suitable activation function, g,
to arrive at the activation value for the current hidden layer, ht . Once we have the
values for the hidden layer, we proceed with the usual computation to generate the
output vector.
Let’s refer to the input, hidden and output layer dimensions as din , dh , and dout
respectively. Given this, our three parameter matrices are: W ∈ Rdh ×din , U ∈ Rdh ×dh ,
and V ∈ Rdout ×dh .
We compute yt via a softmax computation that gives a probability distribution
over the possible output classes.
yt = softmax(Vht ) (13.3)
The fact that the computation at time t requires the value of the hidden layer from
time t − 1 mandates an incremental inference algorithm that proceeds from the start
of the sequence to the end as illustrated in Fig. 13.3. The sequential nature of simple
recurrent networks can also be seen by unrolling the network in time as is shown in
Fig. 13.4. In this figure, the various layers of units are copied for each time step to
illustrate that they will have differing values over time. However, the various weight
matrices are shared across time.
h0 ← 0
for i ← 1 to L ENGTH(x) do
hi ← g(Uhi−1 + Wxi )
yi ← f (Vhi )
return y
Figure 13.3 Forward inference in a simple recurrent network. The matrices U, V and W
are shared across time, while new values for h and y are calculated with each time step.
13.1.2 Training
As with feedforward networks, we’ll use a training set, a loss function, and back-
propagation to obtain the gradients needed to adjust the weights in these recurrent
networks. As shown in Fig. 13.2, we now have 3 sets of weights to update: W, the
weights from the input layer to the hidden layer, U, the weights from the previous
hidden layer to the current hidden layer, and finally V, the weights from the hidden
layer to the output layer.
Fig. 13.4 highlights two considerations that we didn’t have to worry about with
backpropagation in feedforward networks. First, to compute the loss function for
the output at time t we need the hidden layer from time t − 1. Second, the hidden
layer at time t influences both the output at time t and the hidden layer at time t + 1
(and hence the output and loss at t + 1). It follows from this that to assess the error
accruing to ht , we’ll need to know its influence on both the current output as well as
the ones that follow.
282 C HAPTER 13 • RNN S AND LSTM S
y3
y2 h3
V U W
h2
y1 x3
U W
V
h1 x2
U W
h0 x1
Figure 13.4 A simple recurrent neural network shown unrolled in time. Network layers are recalculated for
each time step, while the weights U, V and W are shared across all time steps.
The n-gram language models of Chapter 3 compute the probability of a word given
counts of its occurrence with the n − 1 prior words. The context is thus of size n − 1.
For the feedforward language models of Chapter 6, the context is the window size.
RNN language models (Mikolov et al., 2010) process the input sequence one
word at a time, attempting to predict the next word from the current word and the
previous hidden state. RNNs thus don’t have the limited context problem that n-gram
models have, or the fixed context that feedforward language models have, since the
hidden state can in principle represent information about all of the preceding words
all the way back to the beginning of the sequence. Fig. 13.5 sketches this difference
between a FFN language model and an RNN language model, showing that the
RNN language model uses ht−1 , the hidden state from the previous time step, as a
representation of the past context.
^
yt
a) b)
^
yt
U
V
ht
ht-2 U ht-1 U ht
W
W W W
Figure 13.5 Simplified sketch of two LM architectures moving through a text, showing a
schematic context of three tokens: (a) a feedforward neural language model which has a fixed
context input to the weight matrix W, (b) an RNN language model, in which the hidden state
ht−1 summarizes the prior context.
words each represented as a one-hot vector of size |V | × 1, and the output predic-
tion, ŷ, is a vector representing a probability distribution over the vocabulary. At
each step, the model uses the word embedding matrix E to retrieve the embedding
for the current word, multiples it by the weight matrix W, and then adds it to the hid-
den layer from the previous step (weighted by weight matrix U) to compute a new
hidden layer. This hidden layer is then used to generate an output layer which is
passed through a softmax layer to generate a probability distribution over the entire
vocabulary. That is, at time t:
et = Ext (13.4)
ht = g(Uht−1 + Wet ) (13.5)
ŷt = softmax(Vht ) (13.6)
When we do language modeling with RNNs (and we’ll see this again in Chapter 8
with transformers), it’s convenient to make the assumption that the embedding di-
mension de and the hidden dimension dh are the same. So we’ll just call both of
these the model dimension d. So the embedding matrix E is of shape [d × |V |], and
xt is a one-hot vector of shape [|V | × 1]. The product et is thus of shape [d × 1]. W
and U are of shape [d × d], so ht is also of shape [d × 1]. V is of shape [|V | × d],
so the result of Vh is a vector of shape [|V | × 1]. This vector can be thought of as
a set of scores over the vocabulary given the evidence provided in h. Passing these
scores through the softmax normalizes the scores into a probability distribution. The
probability that a particular word k in the vocabulary is the next word is represented
by ŷt [k], the kth component of ŷt :
The probability of an entire sequence is just the product of the probabilities of each
item in the sequence, where we’ll use ŷi [wi ] to mean the probability of the true word
wi at time step i.
n
Y
P(w1:n ) = P(wi |w1:i−1 ) (13.8)
i=1
Yn
= ŷi [wi ] (13.9)
i=1
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log ŷthanks
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Softmax over
Vocabulary
Vh
h
RNN …
Input
e …
Embeddings
In the case of language modeling, the correct distribution yt comes from knowing the
next word. This is represented as a one-hot vector corresponding to the vocabulary
where the entry for the actual next word is 1, and all the other entries are 0. Thus,
the cross-entropy loss for language modeling is determined by the probability the
model assigns to the correct next word. So at time t the CE loss is the negative log
probability the model assigns to the next word in the training sequence.
Thus at each word position t of the input, the model takes as input the correct word wt
together with ht−1 , encoding information from the preceding w1:t−1 , and uses them
to compute a probability distribution over possible next words so as to compute the
model’s loss for the next token wt+1 . Then we move to the next word, we ignore
what the model predicted for the next word and instead use the correct word wt+1
along with the prior history encoded to estimate the probability of token wt+2 . This
idea that we always give the model the correct history sequence to predict the next
word (rather than feeding the model its best case from the previous time step) is
teacher forcing called teacher forcing.
The weights in the network are adjusted to minimize the average CE loss over
the training sequence via gradient descent. Fig. 13.6 illustrates this training regimen.
et = Ext (13.12)
ht = g(Uht−1 + Wet ) (13.13)
|
ŷt = softmax(E ht ) (13.14)
Argmax NNP MD VB DT NN
y
Softmax over
tags
Vh
RNN h
Layer(s)
Embeddings e
Softmax
FFN
hn
RNN
x1 x2 x3 xn
Figure 13.8 Sequence classification using a simple RNN combined with a feedforward net-
work. The final hidden state from the RNN is used as the input to a feedforward network that
performs the classification.
Note that in this approach we don’t need intermediate outputs for the words in
the sequence preceding the last element. Therefore, there are no loss terms associ-
ated with those elements. Instead, the loss function used to train the weights in the
network is based entirely on the final text classification task. The output from the
softmax output from the feedforward classifier together with a cross-entropy loss
288 C HAPTER 13 • RNN S AND LSTM S
drives the training. The error signal from the classification is backpropagated all the
way through the weights in the feedforward classifier through, to its input, and then
through to the three sets of weights in the RNN as described earlier in Section 13.1.2.
The training regimen that uses the loss from a downstream application to adjust the
end-to-end
training weights all the way through the network is referred to as end-to-end training.
Another option, instead of using just the hidden state of the last token hn to
pooling represent the whole sequence, is to use some sort of pooling function of all the
hidden states hi for each word i in the sequence. For example, we can create a
representation that pools all the n hidden states by taking their element-wise mean:
n
1X
hmean = hi (13.15)
n
i=1
Or we can take the element-wise max; the element-wise max of a set of n vectors is
a new vector whose kth element is the max of the kth elements of all the n vectors.
The long contexts of RNNs makes it quite difficult to successfully backpropagate
error all the way through the entire input; we’ll talk about this problem, and some
standard solutions, in Section 13.5.
Softmax
RNN
Embedding
y1 y2 y3 yn
RNN 3
RNN 2
RNN 1
x1 x2 x3 xn
Figure 13.10 Stacked recurrent networks. The output of a lower level serves as the input
to higher levels with the output of the last network serving as the final output.
quickly.
This new notation h ft simply corresponds to the normal hidden state at time t, repre-
senting everything the network has gleaned from the sequence so far.
To take advantage of context to the right of the current input, we can train an
RNN on a reversed input sequence. With this approach, the hidden state at time t
represents information about the sequence to the right of the current input:
Here, the hidden state hbt represents all the information we have discerned about the
sequence from t to the end of the sequence.
bidirectional A bidirectional RNN (Schuster and Paliwal, 1997) combines two independent
RNN
RNNs, one where the input is processed from the start to the end, and the other from
the end to the start. We then concatenate the two representations computed by the
networks into a single vector that captures both the left and right contexts of an input
at each point in time. Here we use either the semicolon ”;” or the equivalent symbol
⊕ to mean vector concatenation:
ht = [h ft ; hbt ]
= h ft ⊕ hbt (13.18)
13.5 • T HE LSTM 291
Fig. 13.11 illustrates such a bidirectional network that concatenates the outputs of
the forward and backward pass. Other simple ways to combine the forward and
backward contexts include element-wise addition or multiplication. The output at
each step in time thus captures information to the left and to the right of the current
input. In sequence labeling applications, these concatenated outputs can serve as the
basis for a local labeling decision.
y1 y2 y3 yn
concatenated
outputs
RNN 2
RNN 1
x1 x2 x3 xn
Figure 13.11 A bidirectional RNN. Separate models are trained in the forward and back-
ward directions, with the output of each model at each time point concatenated to represent
the bidirectional state at that time point.
Bidirectional RNNs have also proven to be quite effective for sequence classi-
fication. Recall from Fig. 13.8 that for sequence classification we used the final
hidden state of the RNN as the input to a subsequent feedforward classifier. A dif-
ficulty with this approach is that the final state naturally reflects more information
about the end of the sentence than its beginning. Bidirectional RNNs provide a sim-
ple solution to this problem; as shown in Fig. 13.12, we simply combine the final
hidden states from the forward and backward passes (for example by concatenation)
and use that as input for follow-on processing.
Softmax
FFN
← →
h1 hn
←
h1 RNN 2
→
RNN 1 hn
x1 x2 x3 xn
Figure 13.12 A bidirectional RNN for sequence classification. The final hidden units from
the forward and backward passes are combined to represent the entire sequence. This com-
bined representation serves as input to the subsequent classifier.
context. Ideally, a network should be able to retain the distant information about
plural flights until it is needed, while still processing the intermediate parts of the
sequence correctly.
One reason for the inability of RNNs to carry forward critical information is that
the hidden layers, and, by extension, the weights that determine the values in the hid-
den layer, are being asked to perform two tasks simultaneously: provide information
useful for the current decision, and updating and carrying forward information re-
quired for future decisions.
A second difficulty with training RNNs arises from the need to backpropagate
the error signal back through time. Recall from Section 13.1.2 that the hidden layer
at time t contributes to the loss at the next time step since it takes part in that calcula-
tion. As a result, during the backward pass of training, the hidden layers are subject
to repeated multiplications, as determined by the length of the sequence. A frequent
result of this process is that the gradients are eventually driven to zero, a situation
vanishing
gradients called the vanishing gradients problem.
To address these issues, more complex network architectures have been designed
to explicitly manage the task of maintaining relevant context over time, by enabling
the network to learn to forget information that is no longer needed and to remember
information required for decisions still to come.
The most commonly used such extension to RNNs is the long short-term mem-
long short-term
memory ory (LSTM) network (Hochreiter and Schmidhuber, 1997). LSTMs divide the con-
text management problem into two subproblems: removing information no longer
needed from the context, and adding information likely to be needed for later de-
cision making. The key to solving both problems is to learn how to manage this
context rather than hard-coding a strategy into the architecture. LSTMs accomplish
this by first adding an explicit context layer to the architecture (in addition to the
usual recurrent hidden layer), and through the use of specialized neural units that
make use of gates to control the flow of information into and out of the units that
13.5 • T HE LSTM 293
comprise the network layers. These gates are implemented through the use of addi-
tional weights that operate sequentially on the input, and previous hidden layer, and
previous context layers.
The gates in an LSTM share a common design pattern; each consists of a feed-
forward layer, followed by a sigmoid activation function, followed by a pointwise
multiplication with the layer being gated. The choice of the sigmoid as the activation
function arises from its tendency to push its outputs to either 0 or 1. Combining this
with a pointwise multiplication has an effect similar to that of a binary mask. Values
in the layer being gated that align with values near 1 in the mask are passed through
nearly unchanged; values corresponding to lower values are essentially erased.
forget gate The first gate we’ll consider is the forget gate. The purpose of this gate is
to delete information from the context that is no longer needed. The forget gate
computes a weighted sum of the previous state’s hidden layer and the current in-
put and passes that through a sigmoid. This mask is then multiplied element-wise
by the context vector to remove the information from context that is no longer re-
quired. Element-wise multiplication of two vectors (represented by the operator ,
and sometimes called the Hadamard product) is the vector of the same dimension
as the two input vectors, where each element i is the product of element i in the two
input vectors:
ft = σ (U f ht−1 + W f xt ) (13.20)
kt = ct−1 ft (13.21)
The next task is to compute the actual information we need to extract from the previ-
ous hidden state and current inputs—the same basic computation we’ve been using
for all our recurrent networks.
add gate Next, we generate the mask for the add gate to select the information to add to the
current context.
Next, we add this to the modified context vector to get our new context vector.
ct = jt + kt (13.25)
output gate The final gate we’ll use is the output gate which is used to decide what informa-
tion is required for the current hidden state (as opposed to what information needs
to be preserved for future decisions).
Fig. 13.13 illustrates the complete computation for a single LSTM unit. Given the
appropriate weights for the various gates, an LSTM accepts as input the context
layer, and hidden layer from the previous time step, along with the current input
vector. It then generates updated context and hidden vectors as output.
It is the hidden state, ht , that provides the output for the LSTM at each time step.
This output can be used as the input to subsequent layers in a stacked RNN, or at the
final layer of a network ht can be used to provide the final output of the LSTM.
294 C HAPTER 13 • RNN S AND LSTM S
ct-1 ct-1
f
σ ct
+
ct
+
ht-1 ht-1
tanh
tanh
+
g
ht ht
i
σ
+
xt xt
σ
o
LSTM
+
Figure 13.13 A single LSTM unit displayed as a computation graph. The inputs to each unit consists of the
current input, x, the previous hidden state, ht−1 , and the previous context, ct−1 . The outputs are a new hidden
state, ht and an updated context, ct .
h ht ct ht
a a
g g
LSTM
z z
Unit
⌃ ⌃
Figure 13.14 Basic neural units used in feedforward, simple recurrent networks (SRN),
and long short-term memory (LSTM).
networks making use of gated units can be unrolled into deep feedforward networks
and trained in the usual fashion with backpropagation. In practice, therefore, LSTMs
rather than RNNs have become the standard unit for any modern system that makes
use of recurrent networks.
y
y1 y2 yn
…
RNN RNN
x1 x2 … xn x1 x2 … xn
y1 y2 … ym
x2 x3 … xt Decoder RNN
Context
RNN
Encoder RNN
x1 x2 … xt-1
x1 x2 … xn
y1 y2 … ym
Decoder
Context
Encoder
x1 x2 … xn
Figure 13.16 The encoder-decoder architecture. The context is a function of the hidden
representations of the input, and may be used by the decoder in a variety of ways.
p(y) = p(y1 )p(y2 |y1 )p(y3 |y1 , y2 ) . . . p(ym |y1 , ..., ym−1 ) (13.28)
ht = g(ht−1 , xt ) (13.29)
ŷt = softmax(ht ) (13.30)
We only have to make one slight change to turn this language model with au-
toregressive generation into an encoder-decoder model that is a translation model
that can translate from a source text in one language to a target text in a second:
sentence
separation add a sentence separation marker at the end of the source text, and then simply
concatenate the target text.
Let’s use <s> for our sentence separator token, and let’s think about translating
an English source text (“the green witch arrived”), to a Spanish sentence (“llegó
la bruja verde” (which can be glossed word-by-word as ‘arrived the witch green’).
We could also illustrate encoder-decoder models with a question-answer pair, or a
text-summarization pair.
Let’s use x to refer to the source text (in this case in English) plus the separator
token <s>, and y to refer to the target text y (in this case in Spanish). Then an
encoder-decoder model computes the probability p(y|x) as follows:
p(y|x) = p(y1 |x)p(y2 |y1 , x)p(y3 |y1 , y2 , x) . . . p(ym |y1 , ..., ym−1 , x) (13.31)
Fig. 13.17 shows the setup for a simplified version of the encoder-decoder model
(we’ll see the full model, which requires the new concept of attention, in the next
section).
Fig. 13.17 shows an English source text (“the green witch arrived”), a sentence
separator token (<s>, and a Spanish target text (“llegó la bruja verde”). To trans-
late a source text, we run it through the network performing forward inference to
generate hidden states until we get to the end of the source. Then we begin autore-
gressive generation, asking for a word in the context of the hidden layer from the
end of the source input as well as the end-of-sentence marker. Subsequent words
are conditioned on the previous hidden state and the embedding for the last word
generated.
298 C HAPTER 13 • RNN S AND LSTM S
Target Text
hidden hn
layer(s)
embedding
layer
Separator
Source Text
Figure 13.17 Translating a single sentence (inference time) in the basic RNN version of encoder-decoder ap-
proach to machine translation. Source and target sentences are concatenated with a separator token in between,
and the decoder uses context information from the encoder’s last hidden state.
Let’s formalize and generalize this model a bit in Fig. 13.18. (To help keep
things straight, we’ll use the superscripts e and d where needed to distinguish the
hidden states of the encoder and the decoder.) The elements of the network on the
left process the input sequence x and comprise the encoder. While our simplified
figure shows only a single network layer for the encoder, stacked architectures are
the norm, where the output states from the top layer of the stack are taken as the
final representation, and the encoder consists of stacked biLSTMs where the hidden
states from top layers from the forward and backward passes are concatenated to
provide the contextualized representations for each time step.
Decoder
y1 y2 y3 y4 </s>
(output is ignored during encoding)
softmax
embedding
layer
x1 x2 x3 … xn <s> y1 y2 y3 … ym-1
Encoder
Figure 13.18 A more formal version of translating a sentence at inference time in the basic RNN-based
encoder-decoder architecture. The final hidden state of the encoder RNN, hen , serves as the context for the
decoder in its role as hd0 in the decoder RNN, and is also made available to each decoder hidden state.
use c as its prior hidden state hd0 . The decoder would then autoregressively generate
a sequence of outputs, an element at a time, until an end-of-sequence marker is
generated. Each hidden state is conditioned on the previous hidden state and the
output generated in the previous state.
As Fig. 13.18 shows, we do something more complex: we make the context
vector c available to more than just the first decoder hidden state, to ensure that the
influence of the context vector, c, doesn’t wane as the output sequence is generated.
We do this by adding c as a parameter to the computation of the current hidden state.
using the following equation:
Now we’re ready to see the full equations for this version of the decoder in the basic
encoder-decoder model, with context available at each decoding timestep. Recall
that g is a stand-in for some flavor of RNN and ŷt−1 is the embedding for the output
sampled from the softmax at the previous step:
c = hen
hd0 = c
htd = g(ŷt−1 , ht−1
d
, c)
ŷt = softmax(htd ) (13.33)
Thus ŷt is a vector of probabilities over the vocabulary, representing the probability
of each word occurring at time t. To generate text, we sample from this distribution
ŷt . For example, the greedy choice is simply to choose the most probable word to
generate at each timestep. We discussed other sampling methods in Section 7.4.
Decoder
gold
llegó la bruja verde </s> answers
y1 y2 y3 y4 y5
softmax
hidden
layer(s)
embedding
layer
x1 x2 x3 x4
the green witch arrived <s> llegó la bruja verde
Encoder
Figure 13.19 Training the basic RNN encoder-decoder approach to machine translation. Note that in the
decoder we usually don’t propagate the model’s softmax outputs ŷt , but use teacher forcing to force each input
to the correct gold value for training. We compute the softmax output distribution over ŷ in the decoder in order
to compute the loss at each token, which can then be averaged to compute a loss for the sentence. This loss is
then propagated through the decoder parameters and the encoder parameters.
13.8 Attention
The simplicity of the encoder-decoder model is its clean separation of the encoder—
which builds a representation of the source text—from the decoder, which uses this
context to generate a target text. In the model as we’ve described it so far, this
context vector is hn , the hidden state of the last (nth ) time step of the source text.
This final hidden state is thus acting as a bottleneck: it must represent absolutely
everything about the meaning of the source text, since the only thing the decoder
knows about the source text is what’s in this context vector (Fig. 13.20). Information
at the beginning of the sentence, especially for long sentences, may not be equally
well represented in the context vector.
Figure 13.20 Requiring the context c to be only the encoder’s final hidden state forces all
the information from the entire source sentence to pass through this representational bottle-
neck.
hidden states of the encoder. And this weighted average is also informed by part of
the decoder state as well, the state of the decoder right before the current token i.
That is, ci = f (he1 . . . hen , hdi−1 ). The weights focus on (‘attend to’) a particular part of
the source text that is relevant for the token i that the decoder is currently producing.
Attention thus replaces the static context vector with one that is dynamically derived
from the encoder hidden states, but also informed by and hence different for each
token in decoding.
This context vector, ci , is generated anew with each decoding step i and takes
all of the encoder hidden states into account in its derivation. We then make this
context available during decoding by conditioning the computation of the current
decoder hidden state on it (along with the prior hidden state and the previous output
generated by the decoder), as we see in this equation (and Fig. 13.21):
y1 y2 yi
c1 c2 ci
Figure 13.21 The attention mechanism allows each hidden state of the decoder to see a
different, dynamic, context, which is a function of all the encoder hidden states.
The first step in computing ci is to compute how much to focus on each encoder
state, how relevant each encoder state is to the decoder state captured in hdi−1 . We
capture relevance by computing— at each state i during decoding—a score(hdi−1 , hej )
for each encoder state j.
dot-product The simplest such score, called dot-product attention, implements relevance as
attention
similarity: measuring how similar the decoder hidden state is to an encoder hidden
state, by computing the dot product between them:
The score that results from this dot product is a scalar that reflects the degree of
similarity between the two vectors. The vector of these scores across all the encoder
hidden states gives us the relevance of each encoder state to the current step of the
decoder.
To make use of these scores, we’ll normalize them with a softmax to create a
vector of weights, αi j , that tells us the proportional relevance of each encoder hidden
state j to the prior hidden decoder state, hdi−1 .
αi j = softmax(score(hdi−1 , hej ))
exp(score(hdi−1 , hej )
= P d e
(13.36)
k exp(score(hi−1 , hk ))
Finally, given the distribution in α, we can compute a fixed-length context vector for
the current decoder state by taking a weighted average over all the encoder hidden
states.
X
ci = αi j hej (13.37)
j
302 C HAPTER 13 • RNN S AND LSTM S
With this, we finally have a fixed-length context vector that takes into account
information from the entire encoder state that is dynamically updated to reflect the
needs of the decoder at each step of decoding. Fig. 13.22 illustrates an encoder-
decoder network with attention, focusing on the computation of one context vector
ci .
Decoder
X
ci
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↵ij hej
j yi-1 yi
attention
.4 .3 .1 .2
weights
↵ij
hdi 1 hej
<latexit sha1_base64="y8s4mGdpwrGrBnuSR+p1gJJXYdo=">AAAB/nicdVDJSgNBEO2JW4zbqHjy0hgEL4YeJyQBL0EvHiOYBbIMPT09mTY9C909QhgC/ooXD4p49Tu8+Td2FkFFHxQ83quiqp6bcCYVQh9Gbml5ZXUtv17Y2Nza3jF391oyTgWhTRLzWHRcLClnEW0qpjjtJILi0OW07Y4up377jgrJ4uhGjRPaD/EwYj4jWGnJMQ+Cgedk7NSa9IgXq955MKDOrWMWUQnNAFGpYtfsakUTZNtWGUFrYRXBAg3HfO95MUlDGinCsZRdCyWqn2GhGOF0UuilkiaYjPCQdjWNcEhlP5udP4HHWvGgHwtdkYIz9ftEhkMpx6GrO0OsAvnbm4p/ed1U+bV+xqIkVTQi80V+yqGK4TQL6DFBieJjTTARTN8KSYAFJkonVtAhfH0K/yets5JVKdnX5WL9YhFHHhyCI3ACLFAFdXAFGqAJCMjAA3gCz8a98Wi8GK/z1pyxmNkHP2C8fQICDpWK</latexit>
·
hidden he1 he2 he3 hen … ci-1
hdi-1 hdi …
layer(s)
ci
x1 x2 x3 … xn
yi-2 yi-1
Encoder
Figure 13.22 A sketch of the encoder-decoder network with attention, focusing on the computation of ci .
The context value ci is one of the inputs to the computation of hdi . It is computed by taking the weighted sum
of all the encoder hidden states, each weighted by their dot product with the prior decoder hidden state hdi−1 .
It’s also possible to create more sophisticated scoring functions for attention
models. Instead of simple dot product attention, we can get a more powerful function
that computes the relevance of each encoder hidden state to the decoder hidden state
by parameterizing the score with its own set of weights, Ws .
score(hdi−1 , hej ) = hdi−1 Ws hej (13.38)
The weights Ws , which are then trained during normal end-to-end training, give the
network the ability to learn which aspects of similarity between the decoder and
encoder states are important to the current application. This bilinear model also
allows the encoder and decoder to use different dimensional vectors, whereas the
simple dot-product attention requires that the encoder and decoder hidden states
have the same dimensionality.
We’ll return to the concept of attention when we define the transformer archi-
tecture in Chapter 8, which is based on a slight modification of attention called
self-attention.
13.9 Summary
This chapter has introduced the concepts of recurrent neural networks and how they
can be applied to language problems. Here’s a summary of the main points that we
covered:
• In simple Recurrent Neural Networks sequences are processed one element at
a time, with the output of each neural unit at time t based both on the current
input at t and the hidden layer from time t − 1.
H ISTORICAL N OTES 303
Historical Notes
Influential investigations of RNNs were conducted in the context of the Parallel Dis-
tributed Processing (PDP) group at UC San Diego in the 1980’s. Much of this work
was directed at human cognitive modeling rather than practical NLP applications
(Rumelhart and McClelland 1986c, McClelland and Rumelhart 1986). Models using
recurrence at the hidden layer in a feedforward network (Elman networks) were in-
troduced by Elman (1990). Similar architectures were investigated by Jordan (1986)
with a recurrence from the output layer, and Mathis and Mozer (1995) with the
addition of a recurrent context layer prior to the hidden layer. The possibility of
unrolling a recurrent network into an equivalent feedforward network is discussed
in (Rumelhart and McClelland, 1986c).
In parallel with work in cognitive modeling, RNNs were investigated extensively
in the continuous domain in the signal processing and speech communities (Giles
et al. 1994, Robinson et al. 1996). Schuster and Paliwal (1997) introduced bidirec-
tional RNNs and described results on the TIMIT phoneme transcription task.
While theoretically interesting, the difficulty with training RNNs and manag-
ing context over long sequences impeded progress on practical applications. This
situation changed with the introduction of LSTMs in Hochreiter and Schmidhuber
(1997) and Gers et al. (2000). Impressive performance gains were demonstrated
on tasks at the boundary of signal processing and language processing including
phoneme recognition (Graves and Schmidhuber, 2005), handwriting recognition
(Graves et al., 2007) and most significantly speech recognition (Graves et al., 2013).
Interest in applying neural networks to practical NLP problems surged with the
work of Collobert and Weston (2008) and Collobert et al. (2011). These efforts made
use of learned word embeddings, convolutional networks, and end-to-end training.
They demonstrated near state-of-the-art performance on a number of standard shared
tasks including part-of-speech tagging, chunking, named entity recognition and se-
mantic role labeling without the use of hand-engineered features.
Approaches that married LSTMs with pretrained collections of word-embeddings
based on word2vec (Mikolov et al., 2013a) and GloVe (Pennington et al., 2014)
304 C HAPTER 13 • RNN S AND LSTM S
quickly came to dominate many common tasks: part-of-speech tagging (Ling et al.,
2015), syntactic chunking (Søgaard and Goldberg, 2016), named entity recognition
(Chiu and Nichols, 2016; Ma and Hovy, 2016), opinion mining (Irsoy and Cardie,
2014), semantic role labeling (Zhou and Xu, 2015a) and AMR parsing (Foland and
Martin, 2016). As with the earlier surge of progress involving statistical machine
learning, these advances were made possible by the availability of training data pro-
vided by CONLL, SemEval, and other shared tasks, as well as shared resources such
as Ontonotes (Pradhan et al., 2007b), and PropBank (Palmer et al., 2005).
The modern neural encoder-decoder approach was pioneered by Kalchbrenner
and Blunsom (2013), who used a CNN encoder and an RNN decoder. Cho et al.
(2014) (who coined the name “encoder-decoder”) and Sutskever et al. (2014) then
showed how to use extended RNNs for both encoder and decoder. The idea that a
generative decoder should take as input a soft weighting of the inputs, the central
idea of attention, was first developed by Graves (2013) in the context of handwriting
recognition. Bahdanau et al. (2015) extended the idea, named it “attention” and
applied it to MT.
CHAPTER
beginning of platypus, puma, and plantain, the middle of leopard, or the end of an-
telope. In general, however, the mapping between the letters of English orthography
and phones is relatively opaque; a single letter can represent very different sounds
in different contexts. The English letter c corresponds to phone [k] in cougar [k uw
g axr], but phone [s] in cell [s eh l]. Besides appearing as c and k, the phone [k] can
appear as part of x (fox [f aa k s]), as ck (jackal [jh ae k el]) and as cc (raccoon [r ae
k uw n]). Many other languages, for example, Spanish, are much more transparent
in their sound-orthography mapping than English.
There are a wide variety of phonetic resources for phonetic transcription. On-
pronunciation
dictionary line pronunciation dictionaries give phonetic transcriptions for words. The LDC
distributes pronunciation lexicons for Egyptian Arabic, Dutch, English, German,
Japanese, Korean, Mandarin, and Spanish. For English, the CELEX dictionary
(Baayen et al., 1995) has pronunciations for 160,595 wordforms, with syllabifica-
tion, stress, and morphological and part-of-speech information. The open-source
CMU Pronouncing Dictionary (CMU, 1993) has pronunciations for about 134,000
wordforms, while the fine-grained 110,000 word UNISYN dictionary (Fitt, 2002),
freely available for research purposes, gives syllabifications, stress, and also pronun-
ciations for dozens of dialects of English.
Another useful resource is a phonetically annotated corpus, in which a col-
lection of waveforms is hand-labeled with the corresponding string of phones. The
TIMIT corpus (NIST, 1990), originally a joint project between Texas Instruments
(TI), MIT, and SRI, is a corpus of 6300 read sentences, with 10 sentences each from
14.2 • A RTICULATORY P HONETICS 307
630 speakers. The 6300 sentences were drawn from a set of 2342 sentences, some
selected to have particular dialect shibboleths, others to maximize phonetic diphone
coverage. Each sentence in the corpus was phonetically hand-labeled, the sequence
of phones was automatically aligned with the sentence wavefile, and then the au-
tomatic phone boundaries were manually hand-corrected (Seneff and Zue, 1988).
time-aligned
transcription The result is a time-aligned transcription: a transcription in which each phone is
associated with a start and end time in the waveform, like the example in Fig. 14.2.
she had your dark suit in greasy wash water all year
sh iy hv ae dcl jh axr dcl d aa r kcl s ux q en gcl g r iy s ix w aa sh q w aa dx axr q aa l y ix axr
Figure 14.2 Phonetic transcription from the TIMIT corpus, using special ARPAbet features for narrow tran-
scription, such as the palatalization of [d] in had, unreleased final stop in dark, glottalization of final [t] in suit
to [q], and flap of [t] in water. The TIMIT corpus also includes time-alignments (not shown).
The Buckeye corpus (Pitt et al. 2007, Pitt et al. 2005) is a phonetically tran-
scribed corpus of spontaneous American speech, containing about 300,000 words
from 40 talkers. Phonetically transcribed corpora are also available for other lan-
guages, including the Kiel corpus of German and Mandarin corpora transcribed by
the Chinese Academy of Social Sciences (Li et al., 2000).
Figure 14.4 The vocal organs, shown in side view. (Figure from OpenStax University
Physics, CC BY 4.0)
[v], [z], and all the English vowels, among others. Unvoiced sounds include [p], [t],
[k], [f], [s], and others.
The area above the trachea is called the vocal tract; it consists of the oral tract
and the nasal tract. After the air leaves the trachea, it can exit the body through the
mouth or the nose. Most sounds are made by air passing through the mouth. Sounds
nasal made by air passing through the nose are called nasal sounds; nasal sounds (like
English [m], [n], and [ng]) use both the oral and nasal tracts as resonating cavities.
consonant Phones are divided into two main classes: consonants and vowels. Both kinds
vowel of sounds are formed by the motion of air through the mouth, throat or nose. Con-
sonants are made by restriction or blocking of the airflow in some way, and can be
voiced or unvoiced. Vowels have less obstruction, are usually voiced, and are gen-
erally louder and longer-lasting than consonants. The technical use of these terms is
much like the common usage; [p], [b], [t], [d], [k], [g], [f], [v], [s], [z], [r], [l], etc.,
are consonants; [aa], [ae], [ao], [ih], [aw], [ow], [uw], etc., are vowels. Semivow-
els (such as [y] and [w]) have some of the properties of both; they are voiced like
vowels, but they are short and less syllabic like consonants.
labial Labial: Consonants whose main restriction is formed by the two lips coming to-
gether have a bilabial place of articulation. In English these include [p] as
in possum, [b] as in bear, and [m] as in marmot. The English labiodental
consonants [v] and [f] are made by pressing the bottom lip against the upper
row of teeth and letting the air flow through the space in the upper teeth.
14.2 • A RTICULATORY P HONETICS 309
(nasal tract)
alveolar
dental
palatal
velar
bilabial
glottal
dental Dental: Sounds that are made by placing the tongue against the teeth are dentals.
The main dentals in English are the [th] of thing and the [dh] of though, which
are made by placing the tongue behind the teeth with the tip slightly between
the teeth.
alveolar Alveolar: The alveolar ridge is the portion of the roof of the mouth just behind the
upper teeth. Most speakers of American English make the phones [s], [z], [t],
and [d] by placing the tip of the tongue against the alveolar ridge. The word
coronal is often used to refer to both dental and alveolar.
palatal Palatal: The roof of the mouth (the palate) rises sharply from the back of the
palate alveolar ridge. The palato-alveolar sounds [sh] (shrimp), [ch] (china), [zh]
(Asian), and [jh] (jar) are made with the blade of the tongue against the rising
back of the alveolar ridge. The palatal sound [y] of yak is made by placing the
front of the tongue up close to the palate.
velar Velar: The velum, or soft palate, is a movable muscular flap at the very back of the
roof of the mouth. The sounds [k] (cuckoo), [g] (goose), and [N] (kingfisher)
are made by pressing the back of the tongue up against the velum.
glottal Glottal: The glottal stop [q] is made by closing the glottis (by bringing the vocal
folds together).
alveolar ridge, forcing air over the edge of the teeth. In the palato-alveolar fricatives
[sh] and [zh], the tongue is at the back of the alveolar ridge, forcing air through a
groove formed in the tongue. The higher-pitched fricatives (in English [s], [z], [sh]
sibilants and [zh]) are called sibilants. Stops that are followed immediately by fricatives are
called affricates; these include English [ch] (chicken) and [jh] (giraffe).
approximant In approximants, the two articulators are close together but not close enough to
cause turbulent airflow. In English [y] (yellow), the tongue moves close to the roof
of the mouth but not close enough to cause the turbulence that would characterize a
fricative. In English [w] (wood), the back of the tongue comes close to the velum.
American [r] can be formed in at least two ways; with just the tip of the tongue
extended and close to the palate or with the whole tongue bunched up near the palate.
[l] is formed with the tip of the tongue up against the alveolar ridge or the teeth, with
one or both sides of the tongue lowered to allow air to flow over it. [l] is called a
lateral sound because of the drop in the sides of the tongue.
tap A tap or flap [dx] is a quick motion of the tongue against the alveolar ridge. The
consonant in the middle of the word lotus ([l ow dx ax s]) is a tap in most dialects of
American English; speakers of many U.K. dialects would use a [t] instead.
Vowels
Like consonants, vowels can be characterized by the position of the articulators as
they are made. The three most relevant parameters for vowels are what is called
vowel height, which correlates roughly with the height of the highest part of the
tongue, vowel frontness or backness, indicating whether this high point is toward
the front or back of the oral tract and whether the shape of the lips is rounded or
not. Figure 14.6 shows the position of the tongue for different vowels.
tongue palate
closed
velum
Figure 14.6 Tongue positions for English high front [iy], low front [ae] and high back [uw].
In the vowel [iy], for example, the highest point of the tongue is toward the
front of the mouth. In the vowel [uw], by contrast, the high-point of the tongue is
located toward the back of the mouth. Vowels in which the tongue is raised toward
Front vowel the front are called front vowels; those in which the tongue is raised toward the
back vowel back are called back vowels. Note that while both [ih] and [eh] are front vowels,
the tongue is higher for [ih] than for [eh]. Vowels in which the highest point of the
high vowel tongue is comparatively high are called high vowels; vowels with mid or low values
of maximum tongue height are called mid vowels or low vowels, respectively.
Figure 14.7 shows a schematic characterization of the height of different vowels.
It is schematic because the abstract property height correlates only roughly with ac-
tual tongue positions; it is, in fact, a more accurate reflection of acoustic facts. Note
that the chart has two kinds of vowels: those in which tongue height is represented
as a point and those in which it is represented as a path. A vowel in which the tongue
diphthong position changes markedly during the production of the vowel is a diphthong. En-
14.2 • A RTICULATORY P HONETICS 311
high
iy y uw
uw
ih uh
ow
ey oy ax
front back
eh
aw
ao
ay ah
ae aa
low
Figure 14.7 The schematic “vowel space” for English vowels.
Syllables
syllable Consonants and vowels combine to make a syllable. A syllable is a vowel-like (or
sonorant) sound together with some of the surrounding consonants that are most
closely associated with it. The word dog has one syllable, [d aa g] (in our dialect);
the word catnip has two syllables, [k ae t] and [n ih p]. We call the vowel at the
nucleus core of a syllable the nucleus. Initial consonants, if any, are called the onset. Onsets
onset with more than one consonant (as in strike [s t r ay k]), are called complex onsets.
coda The coda is the optional consonant or sequence of consonants following the nucleus.
rime Thus [d] is the onset of dog, and [g] is the coda. The rime, or rhyme, is the nucleus
plus coda. Figure 14.8 shows some sample syllable structures.
σ σ σ
ae m iy n eh g z
Figure 14.8 Syllable structure of ham, green, eggs. σ =syllable.
14.3 Prosody
prosody Prosody is the study of the intonational and rhythmic aspects of language, and in
particular the use of F0, energy, and duration to convey pragmatic, affective, or
conversation-interactional meanings.1 We’ll introduce these acoustic quantities in
detail in the next section when we turn to acoustic phonetics, but briefly we can
think of energy as the acoustic quality that we perceive as loudness, and F0 as the
frequency of the sound that is produced, the acoustic quality that we hear as the
pitch of an utterance. Prosody can be used to mark discourse structure, like the
difference between statements and questions, or the way that a conversation is struc-
tured. Prosody is used to mark the saliency of a particular word or phrase. Prosody
is heavily used for paralinguistic functions like conveying affective meanings like
happiness, surprise, or anger. And prosody plays an important role in managing
turn-taking in conversation.
Reduced Vowels and Schwa Unstressed vowels can be weakened even further to
reduced vowel reduced vowels, the most common of which is schwa ([ax]), as in the second vowel
schwa of parakeet: [p ae r ax k iy t]. In a reduced vowel the articulatory gesture isn’t as
complete as for a full vowel. Not all unstressed vowels are reduced; any vowel, and
diphthongs in particular, can retain its full quality even in unstressed position. For
example, the vowel [iy] can appear in stressed position as in the word eat [iy t] or in
unstressed position as in the word carry [k ae r iy].
prominence In summary, there is a continuum of prosodic prominence, for which it is often
useful to represent levels like accented, stressed, full vowel, and reduced vowel.
14.3.3 Tune
Two utterances with the same prominence and phrasing patterns can still differ
tune prosodically by having different tunes. The tune of an utterance is the rise and
fall of its F0 over time. A very obvious example of tune is the difference between
statements and yes-no questions in English. The same words can be said with a final
question rise F0 rise to indicate a yes-no question (called a question rise):
final fall or a final drop in F0 (called a final fall) to indicate a declarative intonation:
Languages make wide use of tune to express meaning (Xu, 2005). In English,
for example, besides this well-known rise for yes-no questions, a phrase containing
a list of nouns separated by commas often has a short rise called a continuation
continuation rise after each noun. Other examples include the characteristic English contours for
rise
expressing contradiction and expressing surprise.
314 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
14.4.1 Waves
Acoustic analysis is based on the sine and cosine functions. Figure 14.10 shows a
plot of a sine wave, in particular the function
y = A ∗ sin(2π f t) (14.3)
where we have set the amplitude A to 1 and the frequency f to 10 cycles per second.
1.0
–1.0
0 0.1 0.2 0.3 0.4 0.5
Time (s)
Recall from basic mathematics that two important characteristics of a wave are
frequency its frequency and amplitude. The frequency is the number of times a second that
amplitude
14.4 • ACOUSTIC P HONETICS AND S IGNALS 315
a wave repeats itself, that is, the number of cycles. We usually measure frequency
in cycles per second. The signal in Fig. 14.10 repeats itself 5 times in .5 seconds,
Hertz hence 10 cycles per second. Cycles per second are usually called hertz (shortened
to Hz), so the frequency in Fig. 14.10 would be described as 10 Hz. The amplitude
period A of a sine wave is the maximum value on the Y axis. The period T of the wave is
the time it takes for one cycle to complete, defined as
1
T= (14.4)
f
0.02283
–0.01697
0 0.03875
Time (s)
Figure 14.11 A waveform of the vowel [iy] from an utterance shown later in Fig. 14.15 on page 319. The
y-axis shows the level of air pressure above and below normal atmospheric pressure. The x-axis shows time.
Notice that the wave repeats regularly.
The first step in digitizing a sound wave like Fig. 14.11 is to convert the analog
representations (first air pressure and then analog electric signals in a microphone)
sampling into a digital signal. This analog-to-digital conversion has two steps: sampling and
quantization. To sample a signal, we measure its amplitude at a particular time; the
sampling rate is the number of samples taken per second. To accurately measure a
wave, we must have at least two samples in each cycle: one measuring the positive
part of the wave and one measuring the negative part. More than two samples per
cycle increases the amplitude accuracy, but fewer than two samples causes the fre-
quency of the wave to be completely missed. Thus, the maximum frequency wave
that can be measured is one whose frequency is half the sample rate (since every
cycle needs two samples). This maximum frequency for a given sampling rate is
Nyquist
frequency called the Nyquist frequency. Most information in human speech is in frequencies
below 10,000 Hz; thus, a 20,000 Hz sampling rate would be necessary for com-
plete accuracy. But telephone speech is filtered by the switching network, and only
316 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
There are a number of standard file formats for storing the resulting digitized wave-
file, such as Microsoft’s .wav and Apple’s AIFF all of which have special headers;
simple headerless “raw” files are also used. For example, the .wav format is a subset
of Microsoft’s RIFF format for multimedia files; RIFF is a general format that can
represent a series of nested chunks of data and control information. Figure 14.12
shows a simple .wav file with a single data chunk together with its format chunk.
Figure 14.12 Microsoft wavefile header format, assuming simple file with one chunk. Fol-
lowing this 44-byte header would be the data chunk.
captured in the figure. Thus, the frequency of this segment of the wave is 10/.03875
or 258 Hz.
Where does this periodic 258 Hz wave come from? It comes from the speed
of vibration of the vocal folds; since the waveform in Fig. 14.11 is from the vowel
[iy], it is voiced. Recall that voicing is caused by regular openings and closing of
the vocal folds. When the vocal folds are open, air is pushing up through the lungs,
creating a region of high pressure. When the folds are closed, there is no pressure
from the lungs. Thus, when the vocal folds are vibrating, we expect to see regular
peaks in amplitude of the kind we see in Fig. 14.11, each major peak corresponding
to an opening of the vocal folds. The frequency of the vocal fold vibration, or the
fundamental
frequency frequency of the complex wave, is called the fundamental frequency of the wave-
F0 form, often abbreviated F0. We can plot F0 over time in a pitch track. Figure 14.13
pitch track shows the pitch track of a short question, “Three o’clock?” represented below the
waveform. Note the rise in F0 at the end of the question.
500 Hz
0 Hz
three o’clock
0 0.544375
Time (s)
Figure 14.13 Pitch track of the question “Three o’clock?”, shown below the wavefile. Note
the rise in F0 at the end of the question. Note the lack of pitch trace during the very quiet part
(the “o’” of “o’clock”; automatic pitch tracking is based on counting the pulses in the voiced
regions, and doesn’t work if there is no voicing (or insufficient sound).
The vertical axis in Fig. 14.11 measures the amount of air pressure variation;
pressure is force per unit area, measured in Pascals (Pa). A high value on the vertical
axis (a high amplitude) indicates that there is more air pressure at that point in time,
a zero value means there is normal (atmospheric) air pressure, and a negative value
means there is lower than normal air pressure (rarefaction).
In addition to this value of the amplitude at any point in time, we also often
need to know the average amplitude over some time range, to give us some idea
of how great the average displacement of air pressure is. But we can’t just take
the average of the amplitude values over a range; the positive and negative values
would (mostly) cancel out, leaving us with a number close to zero. Instead, we
generally use the RMS (root-mean-square) amplitude, which squares each number
before averaging (making it positive), and then takes the square root at the end.
v
u N
u1 X
RMS amplitudei=1 = t
N
xi2 (14.6)
N
i=1
power The power of the signal is related to the square of the amplitude. If the number
318 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
is it a long movie?
0 1.1675
Time (s)
Figure 14.14 Intensity plot for the sentence “Is it a long movie?”. Note the intensity peaks
at each vowel and the especially high peak for the word long.
Two important perceptual properties, pitch and loudness, are related to fre-
pitch quency and intensity. The pitch of a sound is the mental sensation, or perceptual
correlate, of fundamental frequency; in general, if a sound has a higher fundamen-
tal frequency we perceive it as having a higher pitch. We say “in general” because
the relationship is not linear, since human hearing has different acuities for different
frequencies. Roughly speaking, human pitch perception is most accurate between
100 Hz and 1000 Hz and in this range pitch correlates linearly with frequency. Hu-
man hearing represents frequencies above 1000 Hz less accurately, and above this
range, pitch correlates logarithmically with frequency. Logarithmic representation
means that the differences between high frequencies are compressed and hence not
as accurately perceived. There are various psychoacoustic models of pitch percep-
Mel tion scales. One common model is the mel scale (Stevens et al. 1937, Stevens and
Volkmann 1940). A mel is a unit of pitch defined such that pairs of sounds which
are perceptually equidistant in pitch are separated by an equal number of mels. The
mel frequency m can be computed from the raw acoustic frequency as follows:
f
m = 1127 ln(1 + ) (14.9)
700
As we’ll see in Chapter 15, the mel scale plays an important role in speech
recognition.
14.4 • ACOUSTIC P HONETICS AND S IGNALS 319
The loudness of a sound is the perceptual correlate of the power. So sounds with
higher amplitudes are perceived as louder, but again the relationship is not linear.
First of all, as we mentioned above when we defined µ-law compression, humans
have greater resolution in the low-power range; the ear is more sensitive to small
power differences. Second, it turns out that there is a complex relationship between
power, frequency, and perceived loudness; sounds in certain frequency ranges are
perceived as being louder than those in other frequency ranges.
Various algorithms exist for automatically extracting F0. In a slight abuse of ter-
pitch extraction minology, these are called pitch extraction algorithms. The autocorrelation method
of pitch extraction, for example, correlates the signal with itself at various offsets.
The offset that gives the highest correlation gives the period of the signal. There
are various publicly available pitch extraction toolkits; for example, an augmented
autocorrelation pitch tracker is provided with Praat (Boersma and Weenink, 2005).
sh iy j ax s h ae dx ax b ey b iy
0 1.059
Time (s)
Figure 14.15 A waveform of the sentence “She just had a baby” from the Switchboard corpus (conversation
4325). The speaker is female, was 20 years old in 1991, which is approximately when the recording was made,
and speaks the South Midlands dialect of American English.
she
sh iy
0 0.257
Time (s)
Figure 14.16 A more detailed view of the first word “she” extracted from the wavefile in Fig. 14.15. Notice
the difference between the random noise of the fricative [sh] and the regular voicing of the vowel [iy].
–1
0 0.5
Time (s)
Figure 14.17 A waveform that is the sum of two sine waveforms, one of frequency 10
Hz (note five repetitions in the half-second window) and one of frequency 100 Hz, both of
amplitude 1.
spectrum We can represent these two component frequencies with a spectrum. The spec-
trum of a signal is a representation of each of its frequency components and their
amplitudes. Figure 14.18 shows the spectrum of Fig. 14.17. Frequency in Hz is
on the x-axis and amplitude on the y-axis. Note the two spikes in the figure, one
at 10 Hz and one at 100 Hz. Thus, the spectrum is an alternative representation of
the original waveform, and we use the spectrum as a tool to study the component
frequencies of a sound wave at a particular time point.
Let’s look now at the frequency components of a speech waveform. Figure 14.19
shows part of the waveform for the vowel [ae] of the word had, cut out from the
sentence shown in Fig. 14.15.
Note that there is a complex wave that repeats about ten times in the figure; but
there is also a smaller repeated wave that repeats four times for every larger pattern
(notice the four small peaks inside each repeated wave). The complex wave has a
frequency of about 234 Hz (we can figure this out since it repeats roughly 10 times
14.4 • ACOUSTIC P HONETICS AND S IGNALS 321
60
40
1 2 5 10 20 50 100 200
Frequency (Hz)
0.04968
–0.05554
0 0.04275
Time (s)
Figure 14.19 The waveform of part of the vowel [ae] from the word had cut out from the
waveform shown in Fig. 14.15.
20
Sound pressure level (dB/Hz)
–20
Figure 14.20 A spectrum for the vowel [ae] from the word had in the waveform of She just
had a baby in Fig. 14.15.
The x-axis of a spectrum shows frequency, and the y-axis shows some mea-
sure of the magnitude of each frequency component (in decibels (dB), a logarithmic
measure of amplitude that we saw earlier). Thus, Fig. 14.20 shows significant fre-
quency components at around 930 Hz, 1860 Hz, and 3020 Hz, along with many
other lower-magnitude frequency components. These first two components are just
what we noticed in the time domain by looking at the wave in Fig. 14.19!
Why is a spectrum useful? It turns out that these spectral peaks that are easily
visible in a spectrum are characteristic of different phones; phones have characteris-
322 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
tic spectral “signatures”. Just as chemical elements give off different wavelengths of
light when they burn, allowing us to detect elements in stars by looking at the spec-
trum of the light, we can detect the characteristic signature of the different phones
by looking at the spectrum of a waveform. This use of spectral information is essen-
tial to both human and machine speech recognition. In human audition, the function
cochlea of the cochlea, or inner ear, is to compute a spectrum of the incoming waveform.
Similarly, the acoustic features used in speech recognition are spectral representa-
tions.
Let’s look at the spectrum of different vowels. Since some vowels change over
time, we’ll use a different kind of plot called a spectrogram. While a spectrum
spectrogram shows the frequency components of a wave at one point in time, a spectrogram is a
way of envisioning how the different frequencies that make up a waveform change
over time. The x-axis shows time, as it did for the waveform, but the y-axis now
shows frequencies in hertz. The darkness of a point on a spectrogram corresponds
to the amplitude of the frequency component. Very dark points have high amplitude,
light points have low amplitude. Thus, the spectrogram is a useful way of visualizing
the three dimensions (time x frequency x amplitude).
Figure 14.21 shows spectrograms of three American English vowels, [ih], [ae],
and [ah]. Note that each vowel has a set of dark bars at various frequency bands,
slightly different bands for each vowel. Each of these represents the same kind of
spectral peak that we saw in Fig. 14.19.
5000
Frequency (Hz)
0
0 2.81397
Time (s)
Figure 14.21 Spectrograms for three American English vowels, [ih], [ae], and [uh]
formant Each dark bar (or spectral peak) is called a formant. As we discuss below, a
formant is a frequency band that is particularly amplified by the vocal tract. Since
different vowels are produced with the vocal tract in different positions, they will
produce different kinds of amplifications or resonances. Let’s look at the first two
formants, called F1 and F2. Note that F1, the dark bar closest to the bottom, is in a
different position for the three vowels; it’s low for [ih] (centered at about 470 Hz)
and somewhat higher for [ae] and [ah] (somewhere around 800 Hz). By contrast,
F2, the second dark bar from the bottom, is highest for [ih], in the middle for [ae],
and lowest for [ah].
We can see the same formants in running speech, although the reduction and
coarticulation processes make them somewhat harder to see. Figure 14.22 shows
the spectrogram of “she just had a baby”, whose waveform was shown in Fig. 14.15.
F1 and F2 (and also F3) are pretty clear for the [ax] of just, the [ae] of had, and the
[ey] of baby.
What specific clues can spectral representations give for phone identification?
First, since different vowels have their formants at characteristic places, the spectrum
can distinguish vowels from each other. We’ve seen that [ae] in the sample waveform
had formants at 930 Hz, 1860 Hz, and 3020 Hz. Consider the vowel [iy] at the
14.4 • ACOUSTIC P HONETICS AND S IGNALS 323
sh iy j ax s h ae dx ax b ey b iy
0 1.059
Time (s)
Figure 14.22 A spectrogram of the sentence “she just had a baby” whose waveform was shown in Fig. 14.15.
We can think of a spectrogram as a collection of spectra (time slices), like Fig. 14.20 placed end to end.
beginning of the utterance in Fig. 14.15. The spectrum for this vowel is shown in
Fig. 14.23. The first formant of [iy] is 540 Hz, much lower than the first formant for
[ae], and the second formant (2581 Hz) is much higher than the second formant for
[ae]. If you look carefully, you can see these formants as dark bars in Fig. 14.22 just
around 0.5 seconds.
80
70
60
50
40
30
20
10
0
−10
0 1000 2000 3000
Figure 14.23 A smoothed (LPC) spectrum for the vowel [iy] at the start of She just had a
baby. Note that the first formant (540 Hz) is much lower than the first formant for [ae] shown
in Fig. 14.20, and the second formant (2581 Hz) is much higher than the second formant for
[ae].
The location of the first two formants (called F1 and F2) plays a large role in de-
termining vowel identity, although the formants still differ from speaker to speaker.
Higher formants tend to be caused more by general characteristics of a speaker’s
vocal tract rather than by individual vowels. Formants also can be used to identify
the nasal phones [n], [m], and [ng] and the liquids [l] and [r].
115 Hz glottal fold vibration leads to harmonics (other waves) of 230 Hz, 345 Hz,
460 Hz, and so on on. In general, each of these waves will be weaker, that is, will
have much less amplitude than the wave at the fundamental frequency.
It turns out, however, that the vocal tract acts as a kind of filter or amplifier;
indeed any cavity, such as a tube, causes waves of certain frequencies to be amplified
and others to be damped. This amplification process is caused by the shape of the
cavity; a given shape will cause sounds of a certain frequency to resonate and hence
be amplified. Thus, by changing the shape of the cavity, we can cause different
frequencies to be amplified.
When we produce particular vowels, we are essentially changing the shape of
the vocal tract cavity by placing the tongue and the other articulators in particular
positions. The result is that different vowels cause different harmonics to be ampli-
fied. So a wave of the same fundamental frequency passed through different vocal
tract positions will result in different harmonics being amplified.
We can see the result of this amplification by looking at the relationship between
the shape of the vocal tract and the corresponding spectrum. Figure 14.24 shows
the vocal tract position for three vowels and a typical resulting spectrum. The for-
mants are places in the spectrum where the vocal tract happens to amplify particular
harmonic frequencies.
20
Sound pressure level (dB/Hz)
20
Sound pressure level (dB/Hz)
feature vector ture vectors, each vector representing the information in a small time window of
the signal. Sometimes speech recognition or processing algorithms will start with
the waveform, in which case that processing is done by the convolutional networks
(convnets) that we will introduce in Chapter 15.
Other systems begin instead at a higher level, with the log mel spectrum. So in
this section we introduce this commonly used feature vector: sequences of log mel
spectrum vectors. In the following section we’ll introduce an alternative vector, the
MFCC representation. We’ll introduce these concepts at a relatively high level; a
speech signal processing course is recommended for more details.
We begin by repeating from Section 14.4.2 the process of digitizing and quan-
tizing an analog speech waveform.
0.02283
–0.01697
0 0.03875
Time (s)
Figure 14.25 A waveform of an instance of the vowel [iy] (the last vowel in the word “baby”). The y-axis
shows the level of air pressure above and below normal atmospheric pressure. The x-axis shows time. Notice
that the wave repeats regularly. Repeated from Fig. 14.11.
The first step in digitizing a sound wave like Fig. 14.15 is to convert the analog
representations (first air pressure and then analog electric signals in a microphone)
sampling into a digital signal. This analog-to-digital conversion has two steps: sampling and
quantization. To sample a signal, we measure its amplitude at a particular time; the
sampling rate is the number of samples taken per second. To accurately measure a
wave, we must have at least two samples in each cycle: one measuring the positive
part of the wave and one measuring the negative part. More than two samples per
cycle increases the amplitude accuracy, but fewer than two samples causes the fre-
quency of the wave to be completely missed. Thus, the maximum frequency wave
that can be measured is one whose frequency is half the sample rate (since every
cycle needs two samples). This maximum frequency for a given sampling rate is
Nyquist
frequency called the Nyquist frequency. Most information in human speech is in frequencies
below 10,000 Hz; thus, a 20,000 Hz sampling rate would be necessary for com-
plete accuracy. But telephone speech is filtered by the switching network, and only
frequencies less than 4,000 Hz are transmitted by telephones. Thus, an 8,000 Hz
326 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
14.5.2 Windowing
From the digitized, quantized representation of the waveform, we need to extract
spectral features from a small window of speech that characterizes part of a par-
ticular phoneme. Inside this small window, we can roughly think of the signal as
stationary stationary (that is, its statistical properties are constant within this region). (By
non-stationary contrast, in general, speech is a non-stationary signal, meaning that its statistical
properties are not constant over time). We extract this roughly stationary portion of
speech by using a window which is non-zero inside a region and zero elsewhere, run-
ning this window across the speech signal and multiplying it by the input waveform
to produce a windowed waveform.
frame The speech extracted from each window is called a frame. The windowing is
characterized by three parameters: the window size or frame size of the window
stride (its width in milliseconds), the frame stride, (also called shift or offset) between
successive windows, and the shape of the window.
To extract the signal we multiply the value of the signal at time n, s[n] by the
value of the window at time n, w[n]:
rectangular The window shape sketched in Fig. 14.26 is rectangular; you can see the ex-
14.5 • F EATURE E XTRACTION FOR S PEECH R ECOGNITION : L OG M EL S PECTRUM 327
Window
25 ms
Shift
10 Window
ms 25 ms
Shift
10 Window
ms 25 ms
Figure 14.26 Windowing, showing a 25 ms rectangular window with a 10ms stride.
tracted windowed signal looks just like the original signal. The rectangular window,
however, abruptly cuts off the signal at its boundaries, which creates problems when
we do Fourier analysis. For this reason, for acoustic feature creation we more com-
Hamming monly use the Hamming window, which shrinks the values of the signal toward
zero at the window boundaries, avoiding discontinuities. Figure 14.27 shows both;
the equations are as follows (assuming a window that is L frames long):
1 0 ≤ n ≤ L−1
rectangular w[n] = (14.12)
0 otherwise
0.54 − 0.46 cos( 2πn
L ) 0 ≤ n ≤ L−1
Hamming w[n] = (14.13)
0 otherwise
0.4999
–0.5
0 0.0475896
Time (s)
0.4999 0.4999
0 0
–0.5 –0.4826
0.00455938 0.0256563 0.00455938 0.0256563
Time (s) Time (s)
Figure 14.27 Windowing a sine wave with the rectangular or Hamming windows.
328 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
0.04414
0 0
–20
–0.04121
0.0141752 0.039295 0 8000
Time (s) Frequency (Hz)
(a) (b)
Figure 14.28 (a) A 25 ms Hamming-windowed portion of a signal from the vowel [iy]
and (b) its spectrum computed by a DFT.
We do not introduce the mathematical details of the DFT here, except to note
Euler’s formula that Fourier analysis relies on Euler’s formula, with j as the imaginary unit:
As a brief reminder for those students who have already studied signal processing,
the DFT is defined as follows:
N−1
X 2π
X[k] = x[n]e− j N kn (14.15)
n=0
fast Fourier A commonly used algorithm for computing the DFT is the fast Fourier transform
transform
FFT or FFT. This implementation of the DFT is very efficient but only works for values
of N that are powers of 2.
et al. 1937, Stevens and Volkmann 1940) is a unit of pitch. Pairs of sounds that are
perceptually equidistant in pitch are separated by an equal number of mels. The mel
frequency m can be computed from the raw acoustic frequency by a log transforma-
tion:
f
mel( f ) = 1127 ln(1 + ) (14.16)
700
We implement this intuition by creating a bank of filters that collect energy from
each frequency band, spread logarithmically so that we have very fine resolution
at low frequencies, and less resolution at high frequencies. Figure 14.29 shows a
sample bank of triangular filters that implement this idea, that can be multiplied by
the spectrum to get a mel spectrum.
1
Amplitude
0.5
0
0 7700
8K
Frequency (Hz)
Figure 14.29 The mel filter bank (Davis and Mermelstein, 1980). Each triangular filter,
spaced logarithmically along the mel scale, collects energy from a given frequency range.
Finally, we take the log of each of the mel spectrum values. The human response
to signal level is logarithmic (like the human response to frequency). Humans are
less sensitive to slight differences in amplitude at high amplitudes than at low ampli-
tudes. In addition, using a log makes the feature estimates less sensitive to variations
in input such as power variations due to the speaker’s mouth moving closer or further
from the microphone.
channel We call each scalar output from a particular filter a channel, and so the output
for each input frame from the filterbank is a vector of, say 80 or 128 channels, each
of which represents the log energy of a particular (mel-spaced) frequency band.
Before we send this log mel channel vector to the downstream neural network
layers, it’s common for speech systems to rescale them so they have comparable
ranges. A common type of normalization for speech is to scale the input to be
between -1 and 1 with zero mean across the entire pretraining dataset (see Sec-
tion 4.3.2 in Chapter 4).
14 120 700
600
12 100
500
10 400
80
amplitude
amplitude
amplitude
8 300
60 200
6 100
40
4 0
-100
2 20
-200
0 0 -300
0 1000 2000 3000 4000 5000 6000 7000 8000 0 1000 2000 3000 4000 5000 6000 7000 8000 0 50 100 150 200 250
normalise frequency normalise frequency samples
For simplicity, let’s consider as input the log magnitude spectrum and ignore
the mel scaling. The cepstrum can be thought of as the spectrum of the log of the
spectrum. This may sound confusing. But let’s begin with the easy part: the log
of the spectrum. That is, the cepstrum begins with a standard magnitude spectrum,
such as the one for a vowel shown in Fig. 14.30(a) from Taylor (2009). We then take
the log, that is, replace each amplitude value in the magnitude spectrum with its log,
as shown in Fig. 14.30(b).
The next step is to visualize the log spectrum as if itself were a waveform. In
other words, consider the log spectrum in Fig. 14.30(b). Let’s imagine removing the
axis labels that tell us that this is a spectrum (frequency on the x-axis) and imagine
that we are dealing with just a normal speech signal with time on the x-axis. What
can we now say about the spectrum of this “pseudo-signal”? Notice that there is a
high frequency repetitive component in this wave: small waves that repeat about 8
times in each 1000 along the x-axis, for a frequency of about 120 Hz. This high
frequency component is caused by the fundamental frequency of the signal and rep-
resents the little peaks in the spectrum at each harmonic of the signal. In addition,
there are some lower frequency components in this “pseudo-signal”; for example,
the envelope or formant structure has about four large peaks in the window, for a
much lower frequency.
Figure 14.30(c) shows the cepstrum: the spectrum that we have been describing
of the log spectrum. This cepstrum (the word cepstrum is formed by reversing
the first four letters of spectrum) is shown with samples along the x-axis. This
is because by taking the spectrum of the log spectrum, we have left the frequency
domain of the spectrum, and gone back to the time domain. It turns out that the
14.7 • S UMMARY 331
N−1 N−1
!
X X
− j 2π
N kn
2π
c[n] = log x[n]e e j N kn (14.17)
n=0 n=0
Energy To the 12 cepstral coefficients from the prior section we add a 13th feature:
the energy from the frame. Energy is a useful cue for phone detection (for example
energy vowels and sibilants have more energy than stops). The energy in a frame is the sum
over time of the power of the samples in the frame; thus, for a signal x in a window
from time sample t1 to time sample t2 , the energy is
t2
X
Energy = x2 [t] (14.18)
t=t1
Delta features We also add features related to the change in cepstral features over
time. Changes in the speech signal, like the slope of a formant at its transitions,
or the change from a stop closure to stop burst, can again provide a useful cue for
delta feature phone identity. To each of the 13 features (12 cepstral features plus energy) a delta
double delta or velocity feature and a double delta or acceleration feature. Each of the 13 delta
features represents the change between frames in the corresponding cepstral/energy
feature, and each of the 13 double delta features represents the change between
frames in the corresponding delta features. These deltas can be simply computed by
just subtracting the value at a frame from the prior value, but in practice it’s common
to fit a polynomial and take its first and second derivative.
14.7 Summary
This chapter has introduced many of the important concepts of phonetics and com-
putational phonetics.
• We can represent the pronunciation of words in terms of units called phones.
The standard system for representing phones is the International Phonetic
Alphabet or IPA. The most common computational system for transcription
of English is the ARPAbet, which conveniently uses ASCII symbols.
332 C HAPTER 14 • P HONETICS AND S PEECH F EATURE E XTRACTION
• Phones can be described by how they are produced articulatorily by the vocal
organs; consonants are defined in terms of their place and manner of articu-
lation and voicing; vowels by their height, backness, and roundness.
• Speech sounds can also be described acoustically. Sound waves can be de-
scribed in terms of frequency, amplitude, or their perceptual correlates, pitch
and loudness.
• The spectrum of a sound describes its different frequency components. While
some phonetic properties are recognizable from the waveform, both humans
and machines rely on spectral analysis for phone detection.
• A spectrogram is a plot of a spectrum over time. Vowels are described by
characteristic harmonics called formants.
Historical Notes
The major insights of articulatory phonetics date to the linguists of 800–150 B . C .
India. They invented the concepts of place and manner of articulation, worked out
the glottal mechanism of voicing, and understood the concept of assimilation. Eu-
ropean science did not catch up with the Indian phoneticians until over 2000 years
later, in the late 19th century. The Greeks did have some rudimentary phonetic
knowledge; by the time of Plato’s Theaetetus and Cratylus, for example, they distin-
guished vowels from consonants, and stop consonants from continuants. The Stoics
developed the idea of the syllable and were aware of phonotactic constraints on pos-
sible words. An unknown Icelandic scholar of the 12th century exploited the concept
of the phoneme and proposed a phonemic writing system for Icelandic, including
diacritics for length and nasality. But his text remained unpublished until 1818 and
even then was largely unknown outside Scandinavia (Robins, 1967). The modern
era of phonetics is usually said to have begun with Sweet, who proposed what is
essentially the phoneme in his Handbook of Phonetics 1877. He also devised an al-
phabet for transcription and distinguished between broad and narrow transcription,
proposing many ideas that were eventually incorporated into the IPA. Sweet was
considered the best practicing phonetician of his time; he made the first scientific
recordings of languages for phonetic purposes and advanced the state of the art of
articulatory description. He was also infamously difficult to get along with, a trait
that is well captured in Henry Higgins, the stage character that George Bernard Shaw
modeled after him. The phoneme was first named by the Polish scholar Baudouin
de Courtenay, who published his theories in 1894.
Introductory phonetics textbooks include Ladefoged (1993) and Clark and Yal-
lop (1995). Wells (1982) is the definitive three-volume source on dialects of English.
Many of the classic insights in acoustic phonetics had been developed by the
late 1950s or early 1960s; just a few highlights include techniques like the sound
spectrograph (Koenig et al., 1946), theoretical insights like the working out of the
source-filter theory and other issues in the mapping between articulation and acous-
tics ((Fant, 1960), Stevens et al. 1953, Stevens and House 1955, Heinz and Stevens
1961, Stevens and House 1961) the F1xF2 space of vowel formants (Peterson and
Barney, 1952), the understanding of the phonetic nature of stress and the use of
duration and intensity as cues (Fry, 1955), and a basic understanding of issues in
phone perception (Miller and Nicely 1955,Liberman et al. 1952). Lehiste (1967) is
a collection of classic papers on acoustic phonetics. Many of the seminal papers of
Gunnar Fant have been collected in Fant (2004).
E XERCISES 333
Exercises
14.1 Find the mistakes in the ARPAbet transcriptions of the following words:
a. “three” [dh r i] d. “study” [s t uh d i] g. “slight” [s l iy t]
b. “sing” [s ih n g] e. “though” [th ow]
c. “eyes” [ay s] f. “planning” [p pl aa n ih ng]
14.2 Ira Gershwin’s lyric for Let’s Call the Whole Thing Off talks about two pro-
nunciations (each) of the words “tomato”, “potato”, and “either”. Transcribe
into the ARPAbet both pronunciations of each of these three words.
14.3 Transcribe the following words in the ARPAbet:
1. dark
2. suit
3. greasy
4. wash
5. water
14.4 Take a wavefile of your choice. Some examples are on the textbook website.
Download the Praat software, and use it to transcribe the wavefiles at the word
level and into ARPAbet phones, using Praat to help you play pieces of each
wavefile and to look at the wavefile and the spectrogram.
14.5 Record yourself saying five of the English vowels: [aa], [eh], [ae], [iy], [uw].
Find F1 and F2 for each of your vowels.
334 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
CHAPTER
We’ll then introduce two families of methods for ASR. The first is the encoder-
decoder paradigm, and we’ll introduce the baseline attention-based encoder decoder
algorithm, sometimes called Listen Attend and Spell after an early implementation.
We’ll also introduce a more advanced encoder-decoder system, OpenAI’s Whisper
system (Radford et al., 2023) as well an open system based on the same architecture,
OWSM (the Open Whisper-style Speech Model) (Peng et al., 2023). (These mod-
els have additional capabilities including translation, as we’ll discuss later). The
second is the use of self-supervised speech models (sometimes called SSL for self-
supervised learning) like Wav2Vec2.0 or HuBERT, which are encoders that learn
abstract representations of speech that can be used for ASR by pairing them with the
CTC loss function for decoding.
We’ll conclude with the standard word error rate metric used to evaluate ASR.
half, with recordings from lower-WER speakers called “clean” and recordings from
higher-WER speakers “other”.
Switchboard The Switchboard corpus of prompted telephone conversations between strangers
was collected in the early 1990s; it contains 2430 conversations averaging 6 min-
utes each, totaling 240 hours of 8 kHz speech and about 3 million words (Godfrey
et al., 1992). Switchboard has the singular advantage of an enormous amount of
auxiliary hand-done linguistic labeling, including parses, dialogue act tags, phonetic
CALLHOME and prosodic labeling, and discourse and information structure. The CALLHOME
corpus was collected in the late 1990s and consists of 120 unscripted 30-minute
telephone conversations between native speakers of English who were usually close
friends or family (Canavan et al., 1997).
CHiME A variety of corpora try to include input that is more natural. The CHiME
Challenge is a series of difficult shared tasks with corpora that deal with robustness
in ASR. The CHiME 6 task, for example, is ASR of conversational speech in real
home environments (specifically dinner parties). The corpus contains recordings of
twenty different dinner parties in real homes, each with four participants, and in three
locations (kitchen, dining area, living room), recorded with distant microphones.
AMI The AMI Meeting Corpus contains 100 hours of recorded group meetings (some
natural meetings, some specially organized), with manual transcriptions and some
CORAAL additional hand-labels (Renals et al., 2007). CORAAL is a collection of over 150
sociolinguistic interviews with African American speakers, with the goal of studying
African American English (AAE), the many variations of language used in African
American communities and others (Kendall and Farrington, 2020). The interviews
are anonymized with transcripts aligned at the utterance level.
There are a wide variety of corpora available in other languages. In Chinese,
HKUST for example, the HKUST Mandarin Telephone Speech corpus has 1206 transcribed
ten-minute telephone conversations between speakers of Mandarin across China in-
cluding conversations between friends and between strangers (Liu et al., 2006). The
AISHELL-1 AISHELL-1 corpus contains 170 hours of Mandarin read speech of sentences taken
from various domains, read by different speakers mainly from northern China (Bu
et al., 2017).
Finally, there are many multilingual corpora. Common Voice (Ardila et al.,
2020) is a freely available crowd-sourced corpus of transcribed read speech, stored
in MPEG-3 format and designed for ASR. Crowd-working volunteers record them-
selves reading scripted speech, with scripts often extracted from from Wikipedia
articles. The recordings are then verified by other contributors. As of the writing of
this chapter, Common Voice includes 33,150 hours of speech from 133 languages.
FLEURS (Conneau et al., 2023) is a parallel speech dataset, built on the MT bench-
mark FLoRes-101 (Goyal et al., 2022), which has 3001 sentences extracted from
English Wikipedia and translated into 101 other languages by human translators.
For a subset of 2009 of the sentences in each of the 102 languages, FLEURS has
recordings of 3 different native speakers reading the sentence, in total about 12 hours
of speech per language.
Figure 15.1 shows the rough percentage of incorrect words (the word error rate,
or WER, defined on page 355) from roughly state-of-the-art systems as of the time
of this writing on some of these tasks. Note that the error rate on English read speech
(like the LibriSpeech clean audiobook corpus) is around 2% ; transcription of speech
read in English is highly accurate. By contrast, the error rate for transcribing conver-
sations between humans is higher; 5.8 to 11% for the Switchboard and CALLHOME
corpora or AMI meetings. The error rate is higher yet again for speakers of varieties
15.2 • C ONVOLUTIONAL N EURAL N ETWORKS 337
like African American English, and yet again for difficult conversational tasks like
transcription of 4-speaker dinner party speech, which can have error rates as high as
25.5%. Character error rates (CER) are also higher for Mandarin natural conversa-
tion than for Mandarin read speech. Error rates are even higher for lower resource
languages; we’ve shown a handful of examples.
input x[n]
x1 xt
kernel w[k]
w1
…
output z[n]
z1 z2 z3 … zt
Figure 15.2 A schematic view of convolution with a kernel (filter) w whose width is 1.
The kernel is walked across the input, and the output at each frame zi is the dot product of the
kernel with the input frame. With a kernel of length 1 we don’t have to worry about flipping
the kernel, and the dot product is just the scalar product. The figure shows the computation of
z3 as x3 × w1 .
Let’s now turn to longer kernels. Although we’ve described the first step of the
convolution as flipping the kernel, in fact in ASR systems (or in component libraries
like pytorch) we skip this step. Technically this means that the algorithm we are
cross-
correlation using is not in fact convolution, it’s instead cross-correlation, which is the name for
an algorithm of walking a kernel across a signal, computing its dot product frame by
frame, without flipping it first. The difference doesn’t matter, since the parameters
of the kernel will be learned during training, and so the model could easily learn a
kernel with the parameters in either order. Still, for historical reasons we still call
this process a 1d convolution rather than cross-correlation.
Let’s see a more general equation for these longer kernels. To avoid the con-
pad volution being undefined at the left and right edges of the signal, we can pad the
input by adding a small number p of zeros at the beginning and end of the signal,
so that we can start the center of the kernel at the first element x1 , and there will be
a defined value to the left of x1 . This also turns out to make it simple to have the
15.2 • C ONVOLUTIONAL N EURAL N ETWORKS 339
output length as the same as the input length. To do this, it’s convenient to define the
kernel vector as having an odd number of elements of length k = 2p + 1, thus with
the center element having p elements on either side. Each element z j of the output
vector z is then computed as the following dot product:
p
X
zj = x j+i wi+p (15.2)
i=−p
Fig. 15.3 shows the computation of the convolution x ∗ w with a kernel whose width
is 3, and with padding of 1 frame at the beginning and end of x with a value of zero.
padding padding
input x[n]
x1 xt
kernel w[k]
w1 wk
…
output z[n]
z1 z2 z3 …
Figure 15.3 A schematic view of convolution with a kernel (filter) width of 3, and with a
padding of 1, showing a zero value added at the start and end of the signal. The (already
flipped) kernel is walked across the input, and the output at each frame zi is the dot product
of the kernel with the input in the window. The figure shows the computation of z3 .
Note that the size k (the receptive field) of the kernel is designed to be small
compared to the signal. For example for the convolutional layers in Whisper, the
kernel width is 3 frames, meaning the kernel is a vector of length 3 (we say that
receptive field the kernel has a receptive field of 3). That means that the kernel is being compared
to 3 frames of speech. In Whisper there is a frame every 10 ms and each frame
represent a window of 25ms of speech information. That means each kernel is ex-
tracting information from about 40 ms of speech (10 + 10 + 12.5 + 12.5). That’s long
enough to extract various phonetic features like formant transitions or stop closures
or aspiration.
We’ve now described a simplified view of convolution in which the input is a
single vector x and the output is a single vector z, both corresponding to a signal
over time. In practice, the input to a convolutional layer is commonly the output
from a log mel spectrum, which means it has many (say 128) channels, one for each
log mel filters output. The kernel will have separate vectors for each of these input
depth channels. We say that the kernel has a depth of 128, meaning that the kernel is of
shape [128,3].
To get the output of the kernel, we sum over all the input channels. That is, we
get a single output zc for each of the input channels xc by convolving the kernel w
340 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
The output at frame j, z j , thus integrates information from all of the input channels.
Finally, the output from a convolution layer is also more complex than just a vec-
tor consisting of a single scalar value to represent each frame. Instead, the output of
the convolution layer for a given input frame needs to be an embedding, a latent rep-
resentation of that frame. As with all neural models, latent representations should
have the model dimensionality, whatever that is. For example the model dimen-
sionality of Whisper is 1280, and so the convolutional layer needs have one output
channel for each of these 1280 dimensions of the model. In order to do this, we’ll
actually learn one separate kernel for each of the model dimensions. That is, we’ll
learn 1280 separate kernels, each kernel having the depth of the number of input
channels (for example 128), and a filter-width (say of 3). That way, the embedding
representation of each frame will have 1280 independently computed features of the
input signal. We show a schematic in Fig. 15.4
dim 1024
1024
output …
dimension
dim 35
channels
…
dim 1
zi time
∑
kernel 35 128
128 …
Depth … 2
128
2 1
1 dot product
Width 3 with
kernel 35
128
128 …
log mel
input 2
channels
1
xi-1 xi xi+1 time
Figure 15.4 A schematic view of a convolutional net with 128 input channels and 1024
output channels. We see how at time point i one of the 1024 kernels (“kernel 35”, each of
depth 128 and width 3) is dot-product-ed with (each of) the 128 log mel spectrum input vec-
tors, and then summed to produce a single value for one dimension of the output embedding
at time i.
stride A 1d convolution layer can also have a stride. Stride is the amount that we move
the kernel over the input between each step. The figures above show a stride of 1,
meaning that we first position the kernel over x1 , then x2 , then x3 , and so on. For a
stride of 2, we would first position the kernel over x1 , then x3 , then x5 , and so on.
A longer stride means a shorter output sequence; a stride of two means the output
15.3 • T HE E NCODER -D ECODER A RCHITECTURE FOR ASR 341
sequence z will be half the length of the input sequence x. Convolutional layers with
strides greater than 1 are commonly used to shorten an input sequence. This is useful
partly because a shorter signal takes less memory and computational bandwidth, but
also, as we’ll see in the next section, because it helps address the mismatch between
the length of acoustic frame embeddings (10 ms) and letters or words, which cover
much more of the signal.
Finally, in practice a convolutional layer can be followed by an output nonlin-
earity, like a ReLU layer.
y1 y2 y3 y4 y5 y6 y7 y8 y9 ym
i t ‘ s t i m e …
H
DECODER
ENCODER
x1 xn
<s> i t ‘ s t i m …
Shorter sequence X …
Subsampling
80-dimensional
log Mel spectrum f1 … ft
per frame
Feature Computation
This architecture is also used in the Whisper model from OpenAI (Radford et al.,
2023). Fig. 15.6 shows a subpart of the Whisper architecture (Whisper also does
other speech tasks like speech translation and voice activity detection, which we’ll
discuss in the next chapter). Whisper models and inference code are publicly re-
leased, but the training code and training data are not. However, there are open-
source projects that use a Whisper-style architecture, like the Open Whisper-style
342 C HAPTER
ch Recognition via Large-Scale Weak15 • AUTOMATIC S PEECH R ECOGNITION
Supervision 4
Sequence-to-sequence learning
training data (680k hours) EN
TRANS-
CRIBE 0.0 The quick brown ⋯
next-token
nscription prediction
cross attention
⋮ ⋮ ⋮
pido zorro marrón salta sobre ⋯” Transformer
Encoder Blocks MLP MLP Transformer
uick brown fox jumps over ⋯ Decoder Blocks
self attention cross attention
self attention
sh transcription MLP
~
cross attention
Sinusoidal
위에 올라 내려다보면 너무나 넓고 넓은 ⋯ Positional self attention
Encoding
Learned
2 × Conv1D + GELU Positional
Encoding
ground music playing)
TRANS-
SOT EN CRIBE 0.0 The quick ⋯
Log-Mel Spectrogram Tokens in Multitask Training Format
Figure 15.6 A sketch of the Whisper architecture from Radford et al. (2023). Because
Whisper is a multitask system that also does translation and diarization (we’ll discuss these
non-ASR tasks in the following chapter), Whisper’s transcription format has a Start of Tran-
training format Language
script (SOT)X→X
token, a language tag, and then an instruction
Time-aligned token for whether to transcribe or
transcription
identification Transcription
translate.
LANGUAGE begin end begin end
time ⋯
TRANSCRIBE text tokens text tokens
previous START OF
TAG Speech Model (OWSM), time which reproduces reproduces
time Whisper-style
time training but
EOT
text tokens TRANSCRIPT offers a fully open-source toolkit and publicly available data (Peng et al., 2023).
NO NO
TRANSLATE text tokens
SPEECH TIMESTAMPS
Custom vocabulary /
prompting
15.3.1X → Input
Voice activity English and Convolutional Layers
Text-only transcription
detection Translation
text timestamp (allows dataset-specific fine-tuning)
tokens tokens The encoder-decoder architecture is particularly appropriate when input and output
(VAD)
sequences have stark length differences, as they do for speech, with long acoustic
feature sequences mapping to much shorter sequences of letters or words. For ex-
ample English,
rview of our approach. A sequence-to-sequence words aremodel
Transformer on average 5 letters
is trained on manyor 1.3 BPE tokens
different speechlong (Bostrom
processing and
tasks,
Durrett, 2020)
tilingual speech recognition, speech translation, and,language
spoken in natural conversation,
identification, andthe average
voice word
activity lasts about
detection. All of 250 mil-
these
ly represented as a sequence of tokensliseconds (Yuanby
to be predicted et the
al.,decoder,
2006), or 25 frames
allowing for aofsingle
10ms. So the
model speechmany
to replace signaldifferent
in 10ms
ditional speech processing pipeline. The frames is about
multitask 5 (25/5)
training to uses
format 19 (25/1.3)
a set of times
speciallonger
tokensthan
that the
servetext
as signal in wordsoror
task specifiers
targets, as further explained in Sectiontokens.
2.3.
Because this length difference is so extreme for speech, encoder-decoder ar-
chitectures for speech usually have a compression stage that shortens the acoustic
g Details feature sequence beforelarge the encoder
dataset stage. (We can
to encourage additionally make
generalization use of a loss
and robustness.
function that is designed to deal well with compression, like the CTC loss function
Please see Appendix F for full training hyperparameters.3
uite of models of various sizes in order we’ll to study later.)
introduce
roperties of Whisper. Please see TableThe anof theDuring
goal
1 for early is
subsampling development
to produce aand evaluation
shorter sequence weXobserved
= x1 , ..., xthat
n that
e train with data parallelism acrosswill be the input toWhisper
accelerators the transformer
modelsencoder. A very simple
had a tendency baselineplausible
to transcribe algorithmbut is a
low frame
with dynamic loss scaling methodcheck-
rateactivation
and called low
sometimesalmost frame
always rate (Pundak
incorrect guesses andfor
Sainath, 2016):offor
the names time i we
speakers.
riewank & Walther, 2000; Chen stack et al.,(concatenate)
2016). the acoustic
This happensfeature
because vector
many fi with the priorintwo
transcripts thevectors fi−1 and
pre-training
e trained with AdamW (Loshchilov & Hutter,f i−2 to make a new vector three times longer. Then we simply
dataset include the name of the person who is speaking, delete f i−1 and fi−2 .
Thus instead of (say) a 40-dimensional acoustic feature vector every 10 ms, we have
radient norm clipping (Pascanu et al., 2013) encouraging the model to try to predict them, but this infor-
a longer vector (say 120-dimensional) every 30 ms, with a shorter sequence length
learning rate decay to zero after a warmup over mation is only rarely inferable from only the most recent 30
8 updates. A batch size of 256 segments was 3
After the original release of Whisper, we trained an additional
e models are trained for 220 updates which is Large model (denoted V2) for 2.5X more epochs while adding
and three passes over the dataset. Due to only SpecAugment (Park et al., 2019), Stochastic Depth (Huang et al.,
a few epochs, over-fitting is not a large concern, 2016), and BPE Dropout (Provilkov et al., 2019) for regularization.
ot use any data augmentation or regularization Reported results have been updated to this improved model unless
15.3 • T HE E NCODER -D ECODER A RCHITECTURE FOR ASR 343
n = 3t .
But the most common way of creating a shorter input sequence is to use the
convolutional layers we introduced in the previous section. When a convolutional
layer has a stride greater than 1, the output sequence becomes shorter than the input
sequence. Let’s see this in two commonly used ASR systems.
The Whisper system (Radford et al., 2023) has an audio context window of 30
seconds. It extracts 128 channel log mel features for each frame, with a 25ms win-
dow and a stride of 10ms. These are then normalized to 0 mean and a range of -1
to 1. A stride of 10 ms (100 frames per second) means there are 3000 input frames
in a 30 second context window. These 3000 frames are passed to two convolutional
layers, each one followed by a nonlinearity (Whisper uses GELU (Gaussian Error
Linear Unit), which is a smoother version of ReLU). The first convolutional layer
has 128 input channels and uses a stride of 1, with number of output channels being
the model dimensionality, and the window length is 3000. For the second convo-
lutional layer the number of input and output channels is the model dimensionality,
and there is a stride of 2. The stride of 2 in the second convolutional layer makes the
output sequence half the length of the input sequence, bringing the output window
length down to 1500 and producing an audio token every 20 ms. Sinusoidal position
embeddings are added to these audio encodings before the output of this front end
is passed to the transformer encoder.
HuBERT (Hsu et al., 2021) uses an alternative front end architecture, in which
convolutional layers are used to completely replace the computation of the spectrum.
So the input is raw 16kHz sampled audio, and it is passed through seven 512-channel
layers with strides [5,2,2,2,2,2,2] and kernel widths [10,3,3,3,3,2,2] which learn both
to extract spectral information, and to shorten the input sequence by 320x, from
16kHz (= one representation per .0625 ms) down to a 20 ms framerate. Positional
encodings are added to the input, and then a GELU and layer norm are applied
before the output is passed to the transformer encoder.
15.3.2 Inference
After the convolutional stage, encoder-decoders for speech use the same architecture
(transformer with cross-attention) as for MT.
Let’s remind ourselves of the encoder-decoder architecture that we introduced
in Chapter 12. It uses two transformers: an encoder, which is the same as the basic
transformer from Chapter 8, and a decoder, which has one addition: a new layer
called the cross-attention layer. The encoder takes the acoustic input X = x1 , ..., xn
and maps them to an output representation Henc = h1 , ..., hn ; via a stack of encoder
blocks.
The decoder is essentially a conditional language model that attends to the en-
coder representation and generates the target text (letters or tokens) one by one, at
each timestep conditioning on the audio representations from the encoder and the
previously generated text to generate a new letter or token.
The transformer blocks in the decoder have an extra layer with a special kind
cross-attention of attention, cross-attention. Cross-attention has the same form as the multi-head
attention in a normal transformer block, except that while the queries as usual come
from the previous layer of the decoder, the keys and values come from the output of
the encoder.
That is, where in standard multi-head attention the input to each attention layer is
X, in cross attention the input is the the final output of the encoder Henc = h1 , ..., hn .
Henc is of shape [n × d], each row representing one acoustic input token. To link the
344 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
y1 y2 yi+1 ym
… Language
Modeling
Henc h1 h2 hi hn Head
… …
Unembedding Matrix
Block K Block L
… … … … … …
Block 2
Block 2
+ +
Feedforward Feedforward
Encoder
Block 1 Layer Normalize
Layer Normalize
+
+
Cross-Attention
Multi-Head Attention Decoder
Layer Normalize Block 1
Layer Normalize +
Causal (Left-to-Right)
x1 x2 … xi … xn
Multi-Head Attention
<> y1 … yi … ym
…
Decoder
Figure 15.7 The transformer block for the encoder and the decoder, showing the residual stream view. The
final output of the encoder Henc = h1 , ..., hn is the context used in the decoder. The decoder is a standard
transformer except with one extra layer, the cross-attention layer, which takes that encoder output Henc and
uses it to form its K and V inputs.
keys and values from the encoder with the query from the prior layer of the decoder,
we multiply the encoder output Henc by the cross-attention layer’s key weights WK
and value weights WV . The query comes from the output from the prior decoder
layer Hdec[`−1] , which is multiplied by the cross-attention layer’s query weights WQ :
QK|
CrossAttention(Q, K, V) = softmax √ V (15.5)
dk
The cross attention thus allows the decoder to attend to the acoustic input as pro-
jected into the entire encoder final output representations. The other attention layer
in each decoder block, the multi-head attention layer, is the same causal (left-to-
right) attention that we saw in Chapter 8. But the multi-head attention in the en-
coder, however, is allowed to look ahead at the entire source language text, so it is
not masked.
For inference, the probability of the output string y is decomposed as:
n
Y
p(y1 , . . . , yn ) = p(yi |y1 , . . . , yi−1 , X) (15.6)
i=1
15.3.3 Learning
Encoder-decoders for speech are trained with the normal cross-entropy loss gener-
ally used for conditional language models. At timestep i of decoding, the loss is the
log probability of the correct token (letter) yi :
The loss for the entire sentence is the sum of these losses:
m
X
LCE = − log p(yi |y1 , . . . , yi−1 , X) (15.10)
i=1
This loss is then backpropagated through the entire end-to-end model to train the
entire encoder-decoder.
As we described in Chapter 12, we normally use teacher forcing, in which the
decoder history is forced to be the correct gold yi rather than the predicted ŷi . It’s
also possible to use a mixture of the gold and decoder output, for example using
the gold output 90% of the time, but with probability .1 taking the decoder output
instead:
Modern data sizes are quite large. For example Whisper-v2 is trained on a corpus
of 680,000 hours of speech, mostly from English, but also including 118,000 hours
from 96 other languages. Data quality is important, so systems that scrape web
data for training implement methods to remove ASR-generated transcripts from their
training corpora, such as filtering data that is all uppercase or all lowercase. The
open OWSM system is trained on 180k hours, mainly hand-transcribed publicly
346 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
y1 y2 y3 y4 y5 … yn
softmax
projection layer A A A A A A
h1 h2 h3 h4 h5 … hn
Transformer
Stack
x1 x2
MSK x3
MSK x4
MSK x5 … xn
7 CNN Layers
Figure 15.8 Schematic architecture for the HuBERT inference pass in training. A 16kHz
wavfile is passed through a series of convolutional layers, some frames are replaced with a
MASK token, and then the sequence is passed though a transformer stack, and then a linear
layer that projects the transformer output to an output embedding. This embedding is com-
pared via cosine with the embeddings for each of the 100/500 phonetic classes to produce a
logit which is passed through softmax to get a probability distribution over the classes at each
frame.
15.4 • S ELF - SUPERVISED MODELS : H U BERT 347
exp(sim(Aht , ec )/τ)
p(c|X,t) = PC (15.12)
c0 =1 exp(sim(Aht , ec0 )/τ)
As Fig. 15.9 shows, in parallel with this forward pass, the input waveform is
passed through an MFCC to create a 39-dimensional vector which is then mapped
to one of the 100 classes by choosing the most similar centroid in the codebook. The
loss function is then the sum, over the set of masked tokens M, of the probability
that the model assigns to these correct units:
348 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
X
L= log p(zt |X) (15.13)
t∈M
Thus, as in masked language modeling, the model is being trained to predict the
units associated with the masked frames. This loss is then backpropagated through
the model
h1 h2 h3 h4 h5 … hn
z1 z2 z3 z4 z5 zn
…
Transformer map to class in
Stack
codebook …
x1 MSK MSK MSK x5 … xn
f1 f2 f3 f4 f5 fn
…
Figure 15.9 The first phase of HuBERT training. A codebook of 100 units (defined as clus-
ters of 39-dimensional MFCC vectors) is used as the targets for training. For each timestep t,
computes the probability of that class, and uses the logprob as the loss.
Once the model has been initially trained to map to MFCC vector centroids, a
second stage of training occurs, where we take the representations produced by the
model, cluster them into 500 clusters, and use those instead as the target for training.
The intuition is that the initial MFCC clusters will bias the model toward phonetic
representations, but after enough training the model will learn more accurate and
fine-grained representations. Fig. 15.10 shows the intuition.
After HuBERT has been pretrained, the projection and cosine layers are removed
and a randomly initialized linear + softmax layer is added, mapping into 29 classes
(corresponding to the 26 English letters and a few extra characters) for the ASR task.
The CNNs are frozen and the rest of the model is finetuned for ASR using the CTC
loss function to be described in Section 15.5.
codebook codebook
MFCC
Hubert forward pass
layer 6 output
Entire training data
10% of training data
Figure 15.10 Creating the targets for the two stages of HuBERT training. In the first stage,
100 acoustic units are created by computing 39-dimensional MFCC vectors for the entire
training data and then clustering them with k-means. In the second stage, 500 units are created
by passing a subsample of the training data through the HuBERT model after the first stage
training, taking the output of an intermediate transformer layer (layer 6) and clustering them
with k-means.
i i i t t … h
target letters
softmax
projection A A A A A A
h1 h2 h3 h4 h5 … hn
Transformer
Stack
x1 x2 x3 x4 x5 … xn
Figure 15.11 The HuBERT finetuning pass after pretraining. The projection layer and co-
sine steps are removed, leaving only a randomly initialized projection/softmax layer. The
CNN layers are frozen, and the rest of the model is finetuned on a dataset of audio with tran-
scripts, trained with the CTC loss (Section 15.5) to produce letters as output. The parameters
that are updated in finetuning are shown in red (the projection layer and the transformer stack).
P
where ||v|| is the L2 norm of the vector dj=1 v2j
2. Re-estimation: Re-estimate the codeword for each cluster by recomputing
the mean (centroid) of all the vectors in the cluster. If Si is the set of vectors
in cluster i, then
∀i :
1 X
µi = v (15.15)
|Si |
v∈Si
15.5 CTC
We pointed out in the previous section that speech recognition has two particular
properties that make it very appropriate for the encoder-decoder architecture, where
the encoder produces an encoding of the input that the decoder uses attention to
explore. First, in speech we have a very long acoustic input sequence X mapping to
a much shorter sequence of letters Y , and second, it’s hard to know exactly which
part of X maps to which part of Y .
In this section we briefly introduce an alternative to encoder-decoder: an algo-
CTC rithm and loss function called CTC, short for Connectionist Temporal Classifica-
tion (Graves et al., 2006), that deals with these problems in a very different way. The
intuition of CTC is to output a single character for every frame of the input, so that
the output is the same length as the input, and then to apply a collapsing function
that combines sequences of identical letters, resulting in a shorter sequence.
Let’s imagine inference on someone saying the word dinner, and let’s suppose
we had a function that chooses the most probable letter for each input spectral frame
representation xi . We’ll call the sequence of letters corresponding to each input
alignment frame an alignment, because it tells us where in the acoustic signal each letter aligns
to. Fig. 15.12 shows one such alignment, and what happens if we use a collapsing
function that just removes consecutive duplicate letters.
Well, that doesn’t work; our naive algorithm has transcribed the speech as diner,
not dinner! Collapsing doesn’t handle double letters. There’s also another problem
with our naive function; it doesn’t tell us what symbol to align with silence in the
input. We don’t want to be transcribing silence as random letters!
The CTC algorithm solves both problems by adding to the transcription alphabet
blank a special symbol for a blank, which we’ll represent as . The blank can be used in
15.5 • CTC 351
Y (output) d i n e r
A (alignment) d i i n n n n e r r r r r r
wavefile
Figure 15.12 A naive algorithm for collapsing an alignment between input and letters.
the alignment whenever we don’t want to transcribe a letter. Blank can also be used
between letters; since our collapsing function collapses only consecutive duplicate
letters, it won’t collapse across . More formally, let’s define the mapping B : a → y
between an alignment a and an output y, which collapses all repeated letters and
then removes all blanks. Fig. 15.13 sketches this collapsing function B.
Y (output) d i n n e r
remove blanks d i n n e r
merge duplicates d i ␣ n ␣ n e r ␣
A (alignment) d i ␣ n n ␣ n e r r r r ␣ ␣
X (input) x1 x2 x3 x4 x5 x6 x7 x8 x9 x10 x11 x12 x13 x14
Figure 15.13 The CTC collapsing function B, showing the space blank character ; re-
peated (consecutive) characters in an alignment A are removed to form the output Y .
d i i n ␣ n n e e e r r ␣ r
d d i n n ␣ n e r r ␣ ␣ ␣ ␣
d d d i n ␣ n n ␣ ␣ ␣ e r r
Figure 15.14 Three other legitimate alignments producing the transcript dinner.
It’s useful to think of the set of all alignments that might produce the same output
Y . We’ll use the inverse of our B function, called B−1 , and represent that set as
B−1 (Y ).
T
Y
PCTC (A|X) = p(at |X) (15.16)
t=1
Thus to find the best alignment  = {â1 , . . . , âT } we can greedily choose the charac-
ter with the max probability at each time step t:
We then pass the resulting sequence A to the CTC collapsing function B to get the
output sequence Y .
Let’s talk about how this simple inference algorithm for finding the best align-
ment A would be implemented. Because we are making a decision at each time
point, we can treat CTC as a sequence-modeling task, where we output one letter
ŷt at time t corresponding to each input token xt , eliminating the need for a full de-
coder. Fig. 15.15 sketches this architecture, where we take an encoder, produce a
hidden state ht at each timestep, and decode by taking a softmax over the character
vocabulary at each time step.
output letter y1 y2 y3 y4 y5 … yn
sequence Y
i i i t t …
Classifier …
+softmax
ENCODER
Shorter input
sequence X x1 … xn
Subsampling
Feature Computation
Figure 15.15 Inference with CTC: using an encoder-only model, with decoding done by
simple softmaxes over the hidden state ht at each output step.
Alas, there is a potential flaw with the inference algorithm sketched in (Eq. 15.17)
and Fig. 15.14. The problem is that we chose the most likely alignment A, but the
most likely alignment may not correspond to the most likely final collapsed output
string Y . That’s because there are many possible alignments that lead to the same
output string, and hence the most likely output string might not correspond to the
most probable alignment. For example, imagine the most probable alignment A for
an input X = [x1 x2 x3 ] is the string [a b ] but the next two most probable alignments
are [b b] and [ b b]. The output Y =[b b], summing over those two alignments,
might be more probable than Y =[a b].
For this reason, the most probable output sequence Y is the one that has, not
the single best CTC alignment, but the highest sum over the probability of all its
15.5 • CTC 353
possible alignments:
X
PCTC (Y |X) = P(A|X)
A∈B−1 (Y )
X T
Y
= p(at |ht )
A∈B−1 (Y ) t=1
Alas, summing over all alignments is very expensive (there are a lot of alignments),
so we approximate this sum by using a version of Viterbi beam search that cleverly
keeps in the beam the high-probability alignments that map to the same output string,
and sums those as an approximation of (Eq. 15.18). See Hannun (2017) for a clear
explanation of this extension of beam search for CTC.
Because of the strong conditional independence assumption mentioned earlier
(that the output at time t is independent of the output at time t − 1, given the input),
CTC does not implicitly learn a language model over the data (unlike the attention-
based encoder-decoder architectures). It is therefore essential when using CTC to
interpolate a language model (and some sort of length factor L(Y )) using interpola-
tion weights that are trained on a devset:
scoreCTC (Y |X) = log PCTC (Y |X) + λ1 log PLM (Y )λ2 L(Y ) (15.19)
To compute CTC loss function for a single input pair (X,Y ), we need the probability
of the output Y given the input X. As we saw in Eq. 15.18, to compute the probability
of a given output Y we need to sum over all the possible alignments that would
collapse to Y . In other words:
X T
Y
PCTC (Y |X) = p(at |ht ) (15.21)
A∈B−1 (Y ) t=1
Naively summing over all possible alignments is not feasible (there are too many
alignments). However, we can efficiently compute the sum by using dynamic pro-
gramming to merge alignments, with a version of the forward-backward algo-
rithm also used to train HMMs (Appendix A) and CRFs. The original dynamic pro-
gramming algorithms for both training and inference are laid out in (Graves et al.,
2006); see (Hannun, 2017) for a detailed explanation of both.
Fig. 15.16 shows a sketch. For training, we can simply weight the two losses with a
λ tuned on a devset:
L = −λ log Pencdec (Y |X) − (1 − λ ) log Pctc (Y |X) (15.22)
For inference, we can combine the two with the language model (or the length
penalty), again with learned weights:
Ŷ = argmax [λ log Pencdec (Y |X) − (1 − λ ) log PCTC (Y |X) + γ log PLM (Y )] (15.23)
Y
i t ’ s t i m e …
ENCODER
<s> i t ‘ s t i m …
…
x1 xn
X T
Y
= p(at |ht , y<ut )
A∈B−1 (Y ) t=1
15.6 • ASR E VALUATION : W ORD E RROR R ATE 355
SOFTMAX
zt,u
PREDICTION
NETWORK hpred
JOINT NETWORK DECODER
u
henct
yu-1
ENCODER
xt
Figure 15.17 The RNN-T model computing the output token distribution at time t by inte-
grating the output of a CTC acoustic encoder and a separate ‘predictor’ language model.
This utterance has six substitutions, three insertions, and one deletion:
6+3+1
Word Error Rate = 100 = 76.9%
13
The standard method for computing word error rates is a free script called sclite,
available from the National Institute of Standards and Technologies (NIST) (NIST,
2005). Sclite is given a series of reference (hand-transcribed, gold-standard) sen-
tences and a matching set of hypothesis sentences. Besides performing alignments,
and computing word error rate, sclite performs a number of other useful tasks. For
example, for error analysis it gives useful information such as confusion matrices
showing which words are often misrecognized for others, and summarizes statistics
of words that are often inserted or deleted. sclite also gives error rates by speaker
(if sentences are labeled for speaker ID), as well as useful statistics like the sentence
Sentence error error rate, the percentage of sentences with at least one word error.
rate
356 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
I II III IV
REF: |it was|the best|of|times it|was the worst|of times| |it was
| | | | | | | |
SYS A:|ITS |the best|of|times it|IS the worst |of times|OR|it was
| | | | | | | |
SYS B:|it was|the best| |times it|WON the TEST |of times| |it was
In region I, system A has two errors (a deletion and an insertion) and system B
has zero; in region III, system A has one error (a substitution) and system B has two.
Let’s define a sequence of variables Z representing the difference between the errors
in the two systems as follows:
NAi the number of errors made on segment i by system A
NBi the number of errors made on segment i by system B
Z NAi − NBi , i = 1, 2, · · · , n where n is the number of segments
In the example above, the sequence of Z values is {2, −1, −1, 1}. Intuitively, if
the two systems are identical, we would expect the average difference, that is, the
average of the Z values, to be zero. If we call the true average of the differences
muz , we would thus like to know whether muz = 0. Following closely the original
proposal and notation of Gillick P
and Cox (1989), we can estimate the true average
from our limited sample as µ̂z = ni=1 Zi /n. The estimate of the variance of the Zi ’s
15.7 • S UMMARY 357
is
n
1 X
σz2 = (Zi − µz )2 (15.24)
n−1
i=1
Let
µ̂z
W= √ (15.25)
σz / n
For a large enough n (> 50), W will approximately have a normal distribution with
unit variance. The null hypothesis is H0 : µz = 0, and it can thus be rejected if
2 ∗ P(Z ≥ |w|) ≤ 0.05 (two-tailed) or P(Z ≥ |w|) ≤ 0.05 (one-tailed), where Z is
standard normal and w is the realized value W ; these probabilities can be looked up
in the standard tables of the normal distribution.
McNemar’s test Earlier work sometimes used McNemar’s test for significance, but McNemar’s
is only applicable when the errors made by the system are independent, which is not
true in continuous speech recognition, where errors made on a word are extremely
dependent on errors made on neighboring words.
Could we improve on word error rate as a metric? It would be nice, for exam-
ple, to have something that didn’t give equal weight to every word, perhaps valuing
content words like Tuesday more than function words like a or of. While researchers
generally agree that this would be a good idea, it has proved difficult to agree on a
metric that works in every application of ASR.
15.7 Summary
This chapter introduced the fundamental algorithms of automatic speech recognition
(ASR).
• The task of speech recognition (or speech-to-text) is to map acoustic wave-
forms to sequences of graphemes.
• The input to a speech recognizer is a series of acoustic waves. that are sam-
pled, quantized, and converted to a spectral representation like the log mel
spectrum.
• Two common paradigms for speech recognition are the encoder-decoder with
attention model, and models based on the CTC loss function. Attention-
based models have higher accuracies, but models based on CTC more easily
adapt to streaming: outputting graphemes online instead of waiting until the
acoustic input is complete.
• ASR is evaluated using the Word Error Rate; the edit distance between the
hypothesis and the gold transcription.
Historical Notes
A number of speech recognition systems were developed by the late 1940s and early
1950s. An early Bell Labs system could recognize any of the 10 digits from a single
speaker (Davis et al., 1952). This system had 10 speaker-dependent stored patterns,
one for each digit, each of which roughly represented the first two vowel formants
358 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
in the digit. They achieved 97%–99% accuracy by choosing the pattern that had
the highest relative correlation coefficient with the input. Fry (1959) and Denes
(1959) built a phoneme recognizer at University College, London, that recognized
four vowels and nine consonants based on a similar pattern-recognition principle.
Fry and Denes’s system was the first to use phoneme transition probabilities to con-
strain the recognizer.
The late 1960s and early 1970s produced a number of important paradigm shifts.
First were a number of feature-extraction algorithms, including the efficient fast
Fourier transform (FFT) (Cooley and Tukey, 1965), the application of cepstral pro-
cessing to speech (Oppenheim et al., 1968), and the development of LPC for speech
coding (Atal and Hanauer, 1971). Second were a number of ways of handling warp-
warping ing; stretching or shrinking the input signal to handle differences in speaking rate
and segment length when matching against stored patterns. The natural algorithm for
solving this problem was dynamic programming, and, as we saw in Appendix A, the
algorithm was reinvented multiple times to address this problem. The first applica-
tion to speech processing was by Vintsyuk (1968), although his result was not picked
up by other researchers, and was reinvented by Velichko and Zagoruyko (1970) and
Sakoe and Chiba (1971) (and 1984). Soon afterward, Itakura (1975) combined this
dynamic programming idea with the LPC coefficients that had previously been used
only for speech coding. The resulting system extracted LPC features from incoming
words and used dynamic programming to match them against stored LPC templates.
The non-probabilistic use of dynamic programming to match a template against in-
dynamic time
warping coming speech is called dynamic time warping.
The third innovation of this period was the rise of the HMM. Hidden Markov
models seem to have been applied to speech independently at two laboratories around
1972. One application arose from the work of statisticians, in particular Baum and
colleagues at the Institute for Defense Analyses in Princeton who applied HMMs
to various prediction problems (Baum and Petrie 1966, Baum and Eagon 1967).
James Baker learned of this work and applied the algorithm to speech processing
(Baker, 1975a) during his graduate work at CMU. Independently, Frederick Jelinek
and collaborators (drawing from their research in information-theoretical models
influenced by the work of Shannon (1948)) applied HMMs to speech at the IBM
Thomas J. Watson Research Center (Jelinek et al., 1975). One early difference was
the decoding algorithm; Baker’s DRAGON system used Viterbi (dynamic program-
ming) decoding, while the IBM system applied Jelinek’s stack decoding algorithm
(Jelinek, 1969). Baker then joined the IBM group for a brief time before founding
the speech-recognition company Dragon Systems.
The use of the HMM, with Gaussian Mixture Models (GMMs) as the phonetic
component, slowly spread through the speech community, becoming the dominant
paradigm by the 1990s. One cause was encouragement by ARPA, the Advanced
Research Projects Agency of the U.S. Department of Defense. ARPA started a
five-year program in 1971 to build 1000-word, constrained grammar, few speaker
speech understanding (Klatt, 1977), and funded four competing systems of which
Carnegie-Mellon University’s Harpy system (Lowerre, 1976), which used a simpli-
fied version of Baker’s HMM-based DRAGON system was the best of the tested sys-
tems. ARPA (and then DARPA) funded a number of new speech research programs,
beginning with 1000-word speaker-independent read-speech tasks like “Resource
Management” (Price et al., 1988), recognition of sentences read from the Wall Street
Journal (WSJ), Broadcast News domain (LDC 1998, Graff 1997) (transcription of
actual news broadcasts, including quite difficult passages such as on-the-street inter-
H ISTORICAL N OTES 359
views) and the Switchboard, CallHome, CallFriend, and Fisher domains (Godfrey
et al. 1992, Cieri et al. 2004) (natural telephone conversations between friends or
bakeoff strangers). Each of the ARPA tasks involved an approximately annual bakeoff at
which systems were evaluated against each other. The ARPA competitions resulted
in wide-scale borrowing of techniques among labs since it was easy to see which
ideas reduced errors the previous year, and the competitions were probably an im-
portant factor in the eventual spread of the HMM paradigm.
By around 1990 neural alternatives to the HMM/GMM architecture for ASR
arose, based on a number of earlier experiments with neural networks for phoneme
recognition and other speech tasks. Architectures included the time-delay neural
network (TDNN)—the first use of convolutional networks for speech— (Waibel
hybrid et al. 1989, Lang et al. 1990), RNNs (Robinson and Fallside, 1991), and the hybrid
HMM/MLP architecture in which a feedforward neural network is trained as a pho-
netic classifier whose outputs are used as probability estimates for an HMM-based
architecture (Morgan and Bourlard 1990, Bourlard and Morgan 1994, Morgan and
Bourlard 1995).
While the hybrid systems showed performance close to the standard HMM/GMM
models, the problem was speed: large hybrid models were too slow to train on the
CPUs of that era. For example, the largest hybrid system, a feedforward network,
was limited to a hidden layer of 4000 units, producing probabilities over only a few
dozen monophones. Yet training this model still required the research group to de-
sign special hardware boards to do vector processing (Morgan and Bourlard, 1995).
A later analytic study showed the performance of such simple feedforward MLPs
for ASR increases sharply with more than 1 hidden layer, even controlling for the
total number of parameters (Maas et al., 2017). But the computational resources of
the time were insufficient for more layers.
Over the next two decades a combination of Moore’s law and the rise of GPUs
allowed deep neural networks with many layers. Performance was getting close to
traditional systems on smaller tasks like TIMIT phone recognition by 2009 (Mo-
hamed et al., 2009), and by 2012, the performance of hybrid systems had surpassed
traditional HMM/GMM systems (Jaitly et al. 2012, Dahl et al. 2012, inter alia).
Originally it seemed that unsupervised pretraining of the networks using a tech-
nique like deep belief networks was important, but by 2013, it was clear that for
hybrid HMM/GMM feedforward networks, all that mattered was to use a lot of data
and enough layers, although a few other components did improve performance: us-
ing log mel features instead of MFCCs, using dropout, and using rectified linear
units (Deng et al. 2013, Maas et al. 2013, Dahl et al. 2013).
Meanwhile early work had proposed the CTC loss function by 2006 (Graves
et al., 2006), and by 2012 the RNN-Transducer was defined and applied to phone
recognition (Graves 2012, Graves et al. 2013), and then to end-to-end speech recog-
nition rescoring (Graves and Jaitly, 2014), and then recognition (Maas et al., 2015),
with advances such as specialized beam search (Hannun et al., 2014). (Our de-
scription of CTC in the chapter draws on Hannun (2017), which we encourage the
interested reader to follow).
The encoder-decoder architecture was applied to speech at about the same time
by two different groups, in the Listen Attend and Spell system of Chan et al. (2016)
and the attention-based encoder decoder architecture of Chorowski et al. (2014)
and Bahdanau et al. (2016). By 2018 Transformers were included in this encoder-
decoder architecture. Karita et al. (2019) is a nice comparison of RNNs vs Trans-
formers in encoder-architectures for ASR, TTS, and speech-to-speech translation.
360 C HAPTER 15 • AUTOMATIC S PEECH R ECOGNITION
Kaldi Popular toolkits for speech processing include Kaldi (Povey et al., 2011) and
ESPnet ESPnet (Watanabe et al. 2018, Hayashi et al. 2020).
Exercises
CHAPTER
16 Text-to-Speech
“Words mean more than what is set down on paper. It takes the human voice to
infuse them with shades of deeper meaning.”
Maya Angelou, I Know Why the Caged Bird Sings
The task of mapping from text to speech is a task with an even longer history than
speech to text. In Vienna in 1769, Wolfgang von Kempelen built for the Empress
Maria Theresa the famous Mechanical Turk, a chess-playing automaton consisting
of a wooden box filled with gears, behind which sat a robot mannequin who played
chess by moving pieces with his mechanical arm. The Turk toured Europe and the
Americas for decades, defeating Napoleon Bonaparte and even playing Charles Bab-
bage. The Mechanical Turk might have been one of the early successes of artificial
intelligence were it not for the fact that it was, alas, a hoax, powered by a human
chess player hidden inside the box.
What is less well known is that von Kempelen, an extraordinarily
prolific inventor, also built between
1769 and 1790 what was definitely
not a hoax: the first full-sentence
speech synthesizer, shown partially to
the right. His device consisted of a
bellows to simulate the lungs, a rub-
ber mouthpiece and a nose aperture, a
reed to simulate the vocal folds, var-
ious whistles for the fricatives, and a
small auxiliary bellows to provide the puff of air for plosives. By moving levers
with both hands to open and close apertures, and adjusting the flexible leather “vo-
cal tract”, an operator could produce different consonants and vowels.
More than two centuries later, we no longer build our synthesizers out of wood
text-to-speech and leather, nor do we need human operators. The modern task of text-to-speech or
TTS TTS, also called speech synthesis, is exactly the reverse of ASR; to map text:
speech
synthesis It’s time for lunch!
to an acoustic waveform:
o Speech Synthesizers
model whose vocabulary includes both speech tokens and text tokens.
We train this language model to take as input two sequences, a text transcript
and a small sample of speech from the desired talker, to tokenize both the text and
the speech into discrete tokens, and then to conditionally generate discrete samples
Yu Wu , Ziqiang Zhang of the , Long
speechZhou, Shujie
corresponding Liu
to the , Member,
text string, voice. Zhuo Chen,
IEEE,
in the desired
Wang, Jinyu Li , Fellow, IEEE, Lei He , Sheng Zhao, and Furu Wei text string and
At inference time we prompt this language model with a tokenized
a sample of the desired voice (tokenized by the codec into discrete audio tokens) and
conditionally generate to produce the desired audio tokens. Then these tokens can
be converted into a waveform.
scribe in the next section. Codecs have three parts: an encoder (that turns
speech into embedding vectors), a quantizer (that turns the embeddings into
discrete tokens) and decoders (that turns the discrete tokens back into speech).
2. The 2-stage conditional language model that can generate audio tokens cor-
responding to the desired text. We’ll sketch this in Section 16.3.
En r
co de
der co
De
Quantization
x zt qt,1 qt,2 … qt,N zqt x^
c
Fig. 16.2 adapted from Mousavi et al. (2025). shows the standard architecture
of an audio tokenizer. Audio tokenizers take as input an audio waveform, and are
364 C HAPTER 16 • T EXT- TO -S PEECH
trained to recreate the same audio waveform out, via an intermediate representation
consisting of discrete tokens created by vector quantization.
Audio tokenizers have three stages:
1. an encoder maps the acoustic waveform, a series of T values x = x1 , x2 , ..., xT ,
to a sequence of τ embeddings z = z1 , z2 , ..., zτ . τ is typically 100-1000 times
smaller than T .
2. a vector quantizer that takes each embedding zt corresponding to part of the
waveform, and represents it by a sequence of discrete tokens each taken from
one of the Nc codebooks, qt = qt,1 , qt,2 , ..., qt,Nc . The vector quantizer also
sums the vector codewords from each codebook to create a quantizer output
vector zq t .
3. a decoder that generates a lossy reconstructed waveform span x̂ from the
quantizer output vector zq t .
Audio tokenizers are generally learned end-to-end, using loss functions that re-
ward a tokenization that allows the system to reconstruct the input waveform.
In the following subsections we’ll go through the components of one particular
tokenizer, the E N C ODEC tokenizer of Défossez et al. (2023).
Decoder
Encoder
(N=16C, S=8)
Conv1D (N=16C, S=8)
(K=2S, N=C, Stride=S)
EncoderBlock Conv1D Conv1DT (K=2S, N=C)
(N=8C, S=5) (K=3, N=C)
DecoderBlock
(N=8C, S=5)
Residual Unit Residual Unit
EncoderBlock Conv1D
(K=3, N=C) DecoderBlock
(N=4C, S=4) (N=4C, S=4)
EncoderBlock DecoderBlock
(N=2C, S=2) (N=2C, S=2)
Figure 16.3 The encoder and decoder stages of the E N C ODEC model. The goal of the encoder is to down-
sample an input waveform by encoded it as a series of embeddings zt at 75Hz, i.e. 75 embeddings a second.
Because the original signal was represented at 24kHz, this is a downsampling of 24000
75 = 320 times. Between
the encoder and decoder is a quantization step producing a lossy embedding zq t . The goal of the decoder is to
take the lossy embedding zq t and upsample it, converting it back to a waveform.
The encoder and decoder of the E N C ODEC model (Défossez et al., 2023) are
sketched in Fig. 16.3. The goal of the encoder is to downsample a span of waveform
at time t, which is at 24kHz—one second of speech has 24,000 real values—to an
embedding representation zt at 75Hz—one second of audio is represented by 75
16.2 • U SING A CODEC TO LEARN DISCRETE AUDIO TOKENS 365
vectors, each of dimensionality D. For the purposes of this explanation, we’ll use
D = 256.
This downsampling is accomplished by having a series of encoder blocks that are
made up of convolutional layers with strides larger than 1 that iteratively downsam-
ple the audio, as we discussed at the end of Section 15.2. The convolution blocks are
sketched in Fig. 16.3, and include a long series of convolutions as well as residual
units that add a convolution to the prior input.
The output of the encoder is an embedding zt at time t, 75 of which are produced
per second. This embedding is then quantized (as discussed in the next section),
turning each embedding zt into a series of Nc discrete symbols qt = qt,1 , qt,2 , ..., qt,Nc ,
and also turning the series of symbols into a new quantizer output vector zq t . Fi-
nally, the decoder takes the output embedding from the quantizer zq t and generates
a waveform via a symmetric set of convnets that upsample the audio.
In summary, a 24kHz waveform comes through, we encode/downsample it into
a vector zt of dimensionality D = 256, quantize it into discrete symbols qt , turn it
back into a vector zq t of dimensionality D = 256, and then decode/upsample that
vector back into a waveform at 24kHz.
256-d vector
Output:
to be quantized
Vector Quantizer discrete symbol
0 1.2
0.9
Encoder 0.1 Similarity 3
…
-55
0.2
7
1 2 3 4 5 … 1024 Cluster #
255 -9
… 256-d codewords
(vectors)
Codebook
Figure 16.4 The basic VQ algorithm at inference time, after the codebook has been learned.
The input is a span of speech encoded by the encoder into a vector of dimensionality D =
256. This vector is compared with each codeword (cluster centroid) in the codebook. The
codeword for cluster 3 is most similar, so the VQ outputs 3 as the discrete representation of
this vector.
estimation step, the codeword for each cluster is recomputed by recalculating a new
mean vector. The result is that the clusters and their centroids slowly adjust to the
training space. We iterate back and forth between these two steps until the algorithm
converges.
VQ can also be used as part of end-to-end training, as we will discuss below,
in which case instead of iterative k-means, we instead recompute the means during
minibatch training via online algorithms like exponential moving averages.
At the end of clustering, the cluster index can be used as a discrete symbol. Each
codeword cluster is also associated with a codeword, the vector which is the centroid of all
the vectors in the cluster. We call the list of cluster ids (tokens) together with their
codebook codeword the codebook, and we often call the cluster id the code.
code In inference, when a new vector comes in, we compare it to each vector in the
codebook. Whichever codeword is closest, we assign it to that codeword’s associated
cluster. Fig. 16.4 shows an intuition of this inference step in the context of speech
encoding:
1. an input speech waveform is encoded into a vector v,
2. this input vector v is compared to each of the 1024 possible codewords in the
codebook,
3. v is found to be most similar to codeword 3,
4. and so the output of VQ is the discrete symbol 3 as a representation of v.
As we will see below, for training the E N C ODEC model end-to-end we will need
a way to turn this discrete symbol back into a waveform. For simple VQ we do
that by directly using the codeword for that cluster, passing that codeword to the
decoder for it to reconstruct the waveform. Of course the codeword vector won’t
exactly match the original vector encoding of the input speech span, especially with
only 1024 possible codewords, but the hope is that it’s at least close if our codebook
is good, and the decoder will still produce reasonable speech. Nonetheless, more
powerful methods are usually used, as we’ll see in the next section.
16.2 • U SING A CODEC TO LEARN DISCRETE AUDIO TOKENS 367
Figure 16.5 Residual VQ (figure from Chen et al. (2025)). We run VQ on the encoder out-
Figure
put 2: The neural
embedding audio codec
to produce modelsymbol
a discrete revisit. Because
and the RVQ is employed,
corresponding the first quantizer
codeword. We thenplays
look
the most important role in reconstruction, and the impact from others gradually decreases.
at the residual, the difference between the encoder output embedding zt and the codeword
chosen by VQ. We then take a second codebook and run VQ on this residual. We repeat the
we can explicitly
process until we control
have 8 the content in speech synthesis. Another direction is to apply pre-training
tokens.
to the neural TTS. Chung et al. [2018] pre-trains speech decoder in TTS through autoregressive
mel-spectrogram prediction. In Ao et al. [2022], the authors propose a unified-modal encoder-decoder
The idea
framework is verywhich
SpeechT5, simple. We rununlabeled
can leverage standardspeech
VQ with a codebook
and text just as
data to pre-train all in Fig. 16.4
components
in the prior
of TTS [Link].
Tjandra etThen for an
al. [2019] input unlabeled
quantizes embedding zt we
speech into take thetokens
discrete codeword vector
by a VQVAE
model
that is [van den Oord
produced, et al.,
let’s call2017],
it zq1and
fortrain
the azmodel
as with the token-to-speech
quantified by codebook sequence.
1, and They
take the
demonstrate that the pre-trained modelt only requires t a small amount of real data for fine-tuning. Bai
difference between the two:
et al. [2022] proposes mask and reconstruction on mel spectrogram and showing better performance
on speech editing and synthesis. Previous TTS pre-training work leverages less than 1K hours of
(1)
data, whereas VALL-E is pre-trainedresidual
with 60K= hours
zt −ofzdata.
q t . Furthermore, VALL-E is the first (16.1)
to
use audio codec codes as intermediate representations, and emerge in-context learning capability in
residual zero-shot
This TTS. is the error in the VQ; the part of the original vector that the VQ
residual
didn’t capture. The residual is kind of a rounding error; it’s as if in VQ we ‘round’
3 vector
the Background: Speech
to the nearest Quantization
codeword, and that creates some error. So we then take that
residual vector and pass it through another
Since audio is typically stored as a sequence of 16-bit vector
integerquantizer! That gives
values, a generative model usisarequired
second
codeword 16that represents the residual part of the vector. We then take the residual
to output 2 = 65, 536 probabilities per timestep to synthesize the raw audio. In addition, the audio
samplethe
from ratesecond
exceeding ten thousand
codeword, leads
and dotothis
an extraordinarily
again. The long totalsequence length,
result is making it more
8 codewords (the
intractable for raw audio synthesis. To this end, speech quantization is required to compress integer
original codeword and the 7 residuals).
values and sequence length. µ-law transformation can quantize each timestep to 256 values and
That means
reconstruct for RVQ
high-quality we represent
raw audio. It is widelytheusedoriginal
in speechspeech span
generative by asuch
models, sequence of 8
as WaveNet
[van den Oord
discrete et al.,(instead
symbols 2016], butofthe1 inference
discrete speed
symbolis still
in slow
basicsince
VQ).the Fig.
sequence
16.5length
shows is not
the
reduced. Recently, vector quantization is widely applied in self-supervised speech models for feature
intuition.
extraction, such as vq-wav2vec [Baevski et al., 2020a] and HuBERT [Hsu et al., 2021]. The following
workWhat do we
[Lakhotia do2021,
et al., when Duwe want
et al., 2022]toshows
reconstruct
the codesthe speech?
from The method
self-supervised models canusedalsoin
Ereconstruct
N C ODEC content,
RVQand the inference
is again simple: speedweis faster than 8WaveNet.
take the codewordsHowever,
andthe speaker
add themidentity has
together!
been discarded and the reconstruction quality is low [Borsos et al., 2022]. AudioLM [Borsos et al.,
The resulting vector
2022] trains speech-to-speech z is then passed through the decoder to generate a
q t language models on both k-means tokens from a self-supervised model waveform.
and .acoustic tokens from a neural codec model, leading to high-quality speech-to-speech generation.
In this paper, we follow AudioLM [Borsos et al., 2022] to leverage neural codec models to represent
16.2.4 Training
speech in discrete tokens. the
To compress audio formodel
E N C ODEC network of audio tokens
transmission, codec models are able to
encode waveform into discrete acoustic codes and reconstruct high-quality waveform even if the
speaker
The E NisCunseen
ODEC in training.
model Compared
(like similartoaudio
traditional audio codec
tokenizer approaches,
models) the neural-based
is trained end to end.
codec is significantly better at low bitrates, and we believe the quantized tokens contain sufficient
The input isabout
information a waveform,
the speakeraand
span of speech
recording of perhaps
conditions. 1 or to
Compared 10other
seconds extracted
quantization from
methods,
athelonger original
audio codec waveform.
shows The
the following desired output
advantages: is theabundant
1) It contains same waveform span, since
speaker information and
acoustic information, which could maintain speaker identity in reconstruction compared to HuBERT
codes [Hsu et al., 2021]. 2) There is an off-the-shelf codec decoder to convert discrete tokens into a
waveform, without the additional efforts on vocoder training like VQ-based methods that operated on
spectrum [Du et al., 2022]. 3) It could reduce the length of time steps for efficiency to address the
problem in µ-law transformation [van den Oord et al., 2016].
4
368 C HAPTER 16 • T EXT- TO -S PEECH
the model is a kind of autoencoder that learns to map to itself. The model is trained
to do this reconstruction on large speech datasets like Common Voice (Ardila et al.,
2020) (over 30,000 hours of speech in 133 languages) as well as other audio data
like Audio Set (Gemmeke et al., 2017) (1.7 million 10 sec excerpts from YouTube
videos labeled from a large ontology including natural, animal, and machine sounds,
music, and so on).
En
cod 𝓛 GAN de
r
er co
De
Quantization
x zt qt,1 qt,2 … qt,N zqt x^
𝓛 VQ
c
𝓛 reconstruction
Figure 16.6 Architecture of audio tokenizer training, figure adapted from Mousavi et al.
(2025). The audio tokenizer is trained with a weighted combination of various loss functions,
summarized in the figure and described below.
The E N C ODEC model, like most audio tokenizers, is trained with a number of
reconstruction loss functions, as suggested in Fig. 16.6. The reconstruction loss Lreconstruction mea-
loss
sures how similar the output waveform is to the input waveform, for example by the
sum-squared difference between the original and reconstructed audio:
T
X
Lreconstruction (x, x̂) = ||xt − x̂t ||2 (16.2)
t=1
output vector zt and the reconstructed vector after the quantization zq t , i.e. the
codeword, summed over all the Nc codebooks and residuals.
Nc
T X
X (c)
LVQ (x, x̂) = ||z(c) t − zq t || (16.3)
t=1 c=1
The total loss function can then just be a weighted sum of these losses:
L(x, x̂) = λ1 Lreconstruction (x, x̂) + λ2 LGAN (x, x̂) + λ3 LVQ (x, x̂) (16.4)
Non-Autoregressive
CT’+1,1
CT’+2,1 CT,1
x C
…
CT’+1,2 CT’+2,2 CT,2
Non-Autoregressive
x C CT’+1,1 CT’+2,1 … CT,1
x C
Text Audio Prompt
Figure 16.7 The 2-stage language modelling approach for VALL-E, showing the inference
stage for the autoregressive transformer and the first 3 of the 7 non-autoregressive transform-
ers. The output sequence of discrete audio codes is generated in two stages. First the au-
toregressive LM generates all the codes for the first quantizer from left to right. Then the
non-autoregressive model is called 7 times to generate the remaining codes conditioned on all
the codes from the preceding quantizer, including conditioning on the codes to the right.
the language model generates the acoustic codes in two stages. First, an autoregres-
sive LM generates the first-quantizer codes for the entire output sequence, given the
input text and enrolled audio. Then given those codes, a non-autoregressive LM is
run 7 times, each time taking as input the output of the initial autoregressive codes
and the prior non-autoregressive quantizer and thus generating the codes from the
remaining quantizers one by one. Fig. 16.7 shows the intuition for the inference
step.
Now let’s see the architecture in a bit more detail. For training, we are given
an audio sample y and its tokenized text transcription x = [x0 , x1 , . . . , xL ]. We use a
pretrained E N C ODEC to convert y into a code matrix C. Let T be the number of
downsampled vectors output by E N C ODEC, with 8 codes per vector. Then we can
represent the encoder output as
CT ×8 = E N C ODEC(y) (16.5)
Here C is a two-dimensional acoustic code matrix that has T × 8 entries, where the
columns represent time and the rows represent different quantizers. That is, the row
vector ct,: of the matrix contains the 8 codes for the t-th frame, and the column vector
c:, j contains the code sequence from the j-th vector quantizer where j ∈ [1, ..., 8].
Given the text x and audio C, we train the TTS as a conditional code language
model to maximize the likelihood of C conditioned on x:
L = − log p(C|x)
T
Y
= − log p(c<t,: , x) (16.6)
CHEN et al.: NEURAL CODEC LANGUAGE MODELS ARE ZERO-SHOT TEXT TO SPEECH
t=0 SYNTHESIZERS 709
Figure
Fig. 3. Training overview 16.8 We
of VALL-E. Training
regard TTSprocedure for VALL-E.
as a conditional Given
codec language the text
modeling task. prompt, theVALL-E
We structure autoregressive
as two conditional codec language
models in a hierarchicaltransformer
structure. The is
ARfirst trained
model is usedtotogenerate eachcode
generate each code of the
of the first first-quantizer
code sequence incode sequence, manner,
an autoregressive autore-while the NAR model is
used to generate each remaining code sequence based on the previous code sequences in a non-autoregressive manner.
gressively The the non-autoregressive transformer generates the rest of the codes. Figure from
Chen et al. (2025).
[Link]
B. Hierarchical Structure: 16.8andshows the intuition. On thesimultaneously,
NAR Model left, we have thus
an audio sample
reducing andcomplexity
the time its from O(T )
transcription, and both are tokenized. Then
As introduced in Section III, the codec codes derived from the to we append
O(1). an [ EOS ] and [ BOS ] token
to x and
neural audio codec model withanRVQ
[ EOS ] token
exhibit twotokeythe end of C and train the autoregressive transformer
properties:
(1) A single speechtosample
predictis the acoustic
encoded tokens, starting
into multiple code se- withC. cTraining:
0,1 , untilConditional
[ EOS ], andCodec
thenLanguage
the non- Modeling
quences with multiple autoregressive
quantizers intransformers
the audio codecto model.
fill in the
(2)otherAstokens.
depicted in Fig. 3, VALL-E is trained using the condi-
is presentinference,
A hierarchical structure During we are
where the code given afrom
sequence text sequence to belanguage
tional codec spoken as y0 , an en-
well as method.
modeling It is noteworthy that
rolledmost
the first quantizer covers speech
of thesample from
acoustic some unseen
information, whilespeaker, for which
the training we have requires
of VALL-E the transcription
only simple utterance-wise
subsequent code sequences contain the residual acoustic infor- audio-transcription pair data, and no complex data such as
mation from their predecessors, serving to refine and augment force-alignment information or additional audio clips of the
the acoustic details. same speaker for reference. This greatly simplifies the process of
Inspired by these properties, we design VALL-E as two collecting and processing training data, facilitating scalability.
conditional codec language models in a hierarchical structure: Specifically, for each audio and corresponding transcription
an Autoregressive (AR) codec language model and a Non- in the training dataset, we initially utilize the audio codec
16.4 • TTS E VALUATION 371
transcript(y0 ). We first run the codec to get an acoustic code matrix for y0 , which will
be CP = C:T 0 ,: = [c0,: , c1,: , . . . cT 0 ,: :]. Next we concatenate the transcription of y0 to
the text sequence to be spoken to create the total input text x, which we pass through
a text tokenizer. At this stage we thus have a tokenized text x and a tokenized audio
prompt CP .
Then we generate CT = C>T 0 ,: = [cT 0 +1,: , . . . cT,: ] conditioned on the text se-
quence x and the prompt CP :
CT = argmax p(CT |CP , x)
CT
T
Y
= argmax p(ct,: |c<t,: , x) (16.7)
CT t=T 0 +1
IEEE TRANSACTIONS ON AUDIO,
Then the generated SPEECH
tokens ANDbeLANGUAGE
CT can converted byPROCESSING,
the E N C ODECVOL. 33, 2025
decoder into
a waveform. Fig. 16.9 shows the intuition.
θAR ) (10)
remaining code sequence See Chen et al. (2025) for more details on the transformer components and other
ence x and the preceding Overall,
details the NAR model is optimized by minimizing the
of training.
toregressive manner, where negative log likelihood of each j-th target code sequence c>T ′ ,j
conditioned on the text sequence x, all the code sequences of
16.4 TTS Evaluation
e sequences of the prompt the acoustic condition C:T ′ ,: and the preceding j − 1 target code
e speaker information of the sequences c>T ′ ,<j .
TTS systems are evaluated by humans, by playing an utterance to listeners and ask-
itly split the code matrix MOSC ingLthem
NARto= give mean
−alog p(Copinion score
>T ′ ,>1 |x,(MOS),
C<T ′a,:rating
, c>Tof′ ,1how (15)
good)the synthesized
; θNAR
d target code matrix C>T ′ ,: utterances are, usually on a scale from 1–5. We can then compare systems by com-
The model is then optimized paring their MOS$ scores
8 on the same sentences (using, e.g., paired t-tests to test for
significant=
differences).
− log p(c>T ′ ,j |x, C<T ′ ,: , C>T ′ ,<j ; θNAR ). (16)
e c>T ′ ,j conditioned on the
j=2
es in the acoustic condition
nces in the target code matrix In practice, to optimize computational efficiency during training,
ner. we do not calculate the training loss by iterating over all values of
t of Fig. 3, we first obtain the j and aggregating the corresponding losses. Instead, during each
372 C HAPTER 16 • T EXT- TO -S PEECH
16.7 Summary
This chapter introduced the fundamental algorithms of text-to-speech (TTS).
• A common modern algorithm for TTS is to use conditional generation with a
language model over audio tokens learned by a codec model.
• A neural audio codec, short for coder/decoder, is a system that encodes ana-
log speech signals into a digitized, discrete compressed representation for
compression.
• The discrete symbols that a codec produces as its compressed representation
can be used as discrete codes for language modeling.
• A codec includes an encoder that uses convnets to downsample speech into
a downsampled embedding, a quantizer that converts the embedding into a
series of discrete tokens, and a decoder that uses convnets to upsample the
tokens/embedding back into a lossy reconstructed waveform.
• Vector Quantization (VQ) is a method for turning a series of vectors into a
series of discrete symbols. This can be done by using k-means clustering, and
then creating a codebook in which each code is represented by a vector at the
centroid of each cluster, called a codeword. Input vector can be assigned the
nearest codeword cluster.
• Residual Vector Quantization (RVQ) is a hierarchical version of vector
quantization that produces multiple codes for an input vector by first quantiz-
ing a vector into a codebook, and then quantizing the residual (the difference
between the codeword and the input vector) and then iterating.
• TTS systems like VALL-E take a text to be synthesized and a sample of the
voice to be used, tokenize with BPE (text) and an audio codec (speech) and
then use an LM to conditionally generate discrete audio tokens corresponding
to the text prompt, in the voice of the speech sample.
• TTS is evaluated by playing a sentence to human listeners and having them
give a mean opinion score (MOS).
Historical Notes
As we noted at the beginning of the chapter, speech synthesis is one of the earliest
fields of speech and language processing. The 18th century saw a number of physical
models of the articulation process, including the von Kempelen model mentioned
above, as well as the 1773 vowel model of Kratzenstein in Copenhagen using organ
pipes.
The early 1950s saw the development of three early paradigms of waveform
synthesis: formant synthesis, articulatory synthesis, and concatenative synthesis.
Formant synthesizers originally were inspired by attempts to mimic human
speech by generating artificial spectrograms. The Haskins Laboratories Pattern
Playback Machine generated a sound wave by painting spectrogram patterns on a
moving transparent belt and using reflectance to filter the harmonics of a wave-
form (Cooper et al., 1951); other very early formant synthesizers include those of
Lawrence (1953) and Fant (1951). Perhaps the most well-known of the formant
synthesizers were the Klatt formant synthesizer and its successor systems, includ-
ing the MITalk system (Allen et al., 1987) and the Klattalk software used in Digital
Equipment Corporation’s DECtalk (Klatt, 1982). See Klatt (1975) for details.
374 C HAPTER 16 • T EXT- TO -S PEECH
A second early paradigm, concatenative synthesis, seems to have been first pro-
posed by Harris (1953) at Bell Laboratories; he literally spliced together pieces of
magnetic tape corresponding to phones. Soon afterwards, Peterson et al. (1958) pro-
posed a theoretical model based on diphones, including a database with multiple
copies of each diphone with differing prosody, each labeled with prosodic features
including F0, stress, and duration, and the use of join costs based on F0 and formant
distance between neighboring units. But such diphone synthesis models were not
actually implemented until decades later (Dixon and Maxey 1968, Olive 1977). The
1980s and 1990s saw the invention of unit selection synthesis, based on larger units
of non-uniform length and the use of a target cost, (Sagisaka 1988, Sagisaka et al.
1992, Hunt and Black 1996, Black and Taylor 1994, Syrdal et al. 2000).
A third paradigm, articulatory synthesizers attempt to synthesize speech by
modeling the physics of the vocal tract as an open tube. Representative models
include Stevens et al. (1953), Flanagan et al. (1975), and Fant (1986). See Klatt
(1975) and Flanagan (1972) for more details.
Most early TTS systems used phonemes as input; development of the text anal-
ysis components of TTS came somewhat later, drawing on NLP. Indeed the first
true text-to-speech system seems to have been the system of Umeda and Teranishi
(Umeda et al. 1968, Teranishi and Umeda 1968, Umeda 1976), which included a
parser that assigned prosodic boundaries, as well as accent and stress.
History of codecs and modern history of neural TTS TBD.
Exercises
Volume II
ANNOTATING LINGUISTIC
STRUCTURE
In the second volume of the book we discuss the task of detecting linguistic
structure. In the early history of NLP these structures were an intermediate step to-
ward deeper language processing. In modern NLP, we don’t generally make explicit
use of parse or other structures inside the large language models we introduced in
Part I.
Instead linguistic structure plays a number of new roles. One important role is for
interpretability: to provide a useful interpretive lens on neural networks. Knowing
that a particular layer or neuron may be computing something related to a particular
kind of structure can help us break open the ‘black box’ and understand what the
components of our language models are doing.
A second important role for linguistic structure is as a practical tool for social
scientific studies of text: knowing which adjective modifies which noun, or whether
a particular implicit metaphor is being used, can be important for measuring attitudes
toward groups or individuals. Detailed semantic structure can be helpful, for exam-
ple in finding particular clauses that have particular meanings in legal contracts.
Word sense labels can help keep any corpus study from measuring facts about the
wrong word sense. Relation structures can be used to help build knowledge bases
from text.
Finally, computation of linguistic structure is an important tool for answering
questions about language itself, a research area called computational linguistics
that is sometimes distinguished from natural language processing. To answer lin-
guistic questions about how language changes over time or across individuals we’ll
need to be able, for example, to parse entire documents from different time periods.
To understand how certain linguistic structures are learned or processed by people,
it’s necessary to be able to automatically label structures for arbitrary text.
In our study of linguistic structure, we begin with one of the oldest tasks in
computational linguistics: the extraction of syntactic structure, and give two sets of
algorithms for parsing: extracting syntactic structure, including constituency pars-
ing and dependency parsing. We then introduce a variety of structures related to
meaning, including semantic roles, word senses, entity relations, and events. We
376
conclude with linguistic structures that tend to be related to discourse and meaning
over larger texts, including coreference and discourse coherence. In each case we’ll
give algorithms for automatically annotating the relevant structure.
378 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
CHAPTER
Dionysius Thrax of Alexandria (c. 100 B . C .), or perhaps someone else (it was a long
time ago), wrote a grammatical sketch of Greek (a “technē”) that summarized the
linguistic knowledge of his day. This work is the source of an astonishing proportion
of modern linguistic vocabulary, including the words syntax, diphthong, clitic, and
parts of speech analogy. Also included are a description of eight parts of speech: noun, verb,
pronoun, preposition, adverb, conjunction, participle, and article. Although earlier
scholars (including Aristotle as well as the Stoics) had their own lists of parts of
speech, it was Thrax’s set of eight that became the basis for descriptions of European
languages for the next 2000 years. (All the way to the Schoolhouse Rock educational
television shows of our childhood, which had songs about 8 parts of speech, like the
late great Bob Dorough’s Conjunction Junction.) The durability of parts of speech
through two millennia speaks to their centrality in models of human language.
Proper names are another important and anciently studied linguistic category.
While parts of speech are generally assigned to individual words or morphemes, a
proper name is often an entire multiword phrase, like the name “Marie Curie”, the
location “New York City”, or the organization “Stanford University”. We’ll use the
named entity term named entity for, roughly speaking, anything that can be referred to with a
proper name: a person, a location, an organization, although as we’ll see the term is
commonly extended to include things that aren’t entities per se.
POS Parts of speech (also known as POS) and named entities are useful clues to
sentence structure and meaning. Knowing whether a word is a noun or a verb tells us
about likely neighboring words (nouns in English are preceded by determiners and
adjectives, verbs by nouns) and syntactic structure (verbs have dependency links to
nouns), making part-of-speech tagging a key aspect of parsing. Knowing if a named
entity like Washington is a name of a person, a place, or a university is important to
many natural language processing tasks like question answering, stance detection,
or information extraction.
In this chapter we’ll introduce the task of part-of-speech tagging, taking a se-
quence of words and assigning each word a part of speech like NOUN or VERB, and
the task of named entity recognition (NER), assigning words or phrases tags like
PERSON , LOCATION , or ORGANIZATION .
Such tasks in which we assign, to each word xi in an input word sequence, a
label yi , so that the output sequence Y has the same length as the input sequence X
sequence
labeling are called sequence labeling tasks. We’ll introduce classic sequence labeling algo-
rithms, one generative— the Hidden Markov Model (HMM)—and one discriminative—
the Conditional Random Field (CRF). In following chapters we’ll introduce modern
sequence labelers based on RNNs and Transformers.
17.1 • (M OSTLY ) E NGLISH W ORD C LASSES 379
Tag
Description Example
ADJ
Adjective: noun modifiers describing properties red, young, awesome
ADV
Adverb: verb modifiers of time, place, manner very, slowly, home, yesterday
Open Class
NOUN
words for persons, places, things, etc. algorithm, cat, mango, beauty
VERB
words for actions and processes draw, provide, go
PROPN
Proper noun: name of a person, organization, place, etc.. Regina, IBM, Colorado
INTJ
Interjection: exclamation, greeting, yes/no response, etc. oh, um, yes, hello
ADP
Adposition (Preposition/Postposition): marks a noun’s in, on, by, under
spacial, temporal, or other relation
Closed Class Words
AUX Auxiliary: helping verb marking tense, aspect, mood, etc., can, may, should, are
CCONJ Coordinating Conjunction: joins two phrases/clauses and, or, but
DET Determiner: marks noun phrase properties a, an, the, this
NUM Numeral one, two, 2026, 11:00, hundred
PART Particle: a function word that must be associated with an- ’s, not, (infinitive) to
other word
PRON Pronoun: a shorthand for referring to an entity or event she, who, I, others
SCONJ Subordinating Conjunction: joins a main clause with a whether, because
subordinate clause such as a sentential complement
PUNCT Punctuation ,̇ , ()
Other
closed class Parts of speech fall into two broad categories: closed class and open class.
open class Closed classes are those with relatively fixed membership, such as prepositions—
new prepositions are rarely coined. By contrast, nouns and verbs are open classes—
new nouns and verbs like iPhone or to fax are continually being created or borrowed.
function word Closed class words are generally function words like of, it, and, or you, which tend
to be very short, occur frequently, and often have structuring uses in grammar.
Four major open classes occur in the languages of the world: nouns (including
proper nouns), verbs, adjectives, and adverbs, as well as the smaller open class of
interjections. English has all five, although not every language does.
noun Nouns are words for people, places, or things, but include others as well. Com-
common noun mon nouns include concrete terms like cat and mango, abstractions like algorithm
and beauty, and verb-like terms like pacing as in His pacing to and fro became quite
annoying. Nouns in English can occur with determiners (a goat, this bandwidth)
take possessives (IBM’s annual revenue), and may occur in the plural (goats, abaci).
count noun Many languages, including English, divide common nouns into count nouns and
mass noun mass nouns. Count nouns can occur in the singular and plural (goat/goats, rela-
tionship/relationships) and can be counted (one goat, two goats). Mass nouns are
used when something is conceptualized as a homogeneous group. So snow, salt, and
proper noun communism are not counted (i.e., *two snows or *two communisms). Proper nouns,
like Regina, Colorado, and IBM, are names of specific persons or entities.
380 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
verb Verbs refer to actions and processes, including main verbs like draw, provide,
and go. English verbs have inflections (non-third-person-singular (eat), third-person-
singular (eats), progressive (eating), past participle (eaten)). While many scholars
believe that all human languages have the categories of noun and verb, others have
argued that some languages, such as Riau Indonesian and Tongan, don’t even make
this distinction (Broschart 1997; Evans 2000; Gil 2000) .
adjective Adjectives often describe properties or qualities of nouns, like color (white,
black), age (old, young), and value (good, bad), but there are languages without
adjectives. In Korean, for example, the words corresponding to English adjectives
act as a subclass of verbs, so what is in English an adjective “beautiful” acts in
Korean like a verb meaning “to be beautiful”.
adverb Adverbs are a hodge-podge. All the italicized words in this example are adverbs:
Actually, I ran home extremely quickly yesterday
Adverbs generally modify something (often verbs, hence the name “adverb”, but
locative also other adverbs and entire verb phrases). Directional adverbs or locative ad-
degree verbs (home, here, downhill) specify the direction or location of some action; degree
adverbs (extremely, very, somewhat) specify the extent of some action, process, or
manner property; manner adverbs (slowly, slinkily, delicately) describe the manner of some
temporal action or process; and temporal adverbs describe the time that some action or event
took place (yesterday, Monday).
interjection Interjections (oh, hey, alas, uh, um) are a smaller open class that also includes
greetings (hello, goodbye) and question responses (yes, no, uh-huh).
preposition English adpositions occur before nouns, hence are called prepositions. They can
indicate spatial or temporal relations, whether literal (on it, before then, by the house)
or metaphorical (on time, with gusto, beside herself), and relations like marking the
agent in Hamlet was written by Shakespeare.
particle A particle resembles a preposition or an adverb and is used in combination with
a verb. Particles often have extended meanings that aren’t quite the same as the
prepositions they resemble, as in the particle over in she turned the paper over. A
phrasal verb verb and a particle acting as a single unit is called a phrasal verb. The meaning
of phrasal verbs is often non-compositional—not predictable from the individual
meanings of the verb and the particle. Thus, turn down means ‘reject’, rule out
‘eliminate’, and go on ‘continue’.
determiner Determiners like this and that (this chapter, that page) can mark the start of an
article English noun phrase. Articles like a, an, and the, are a type of determiner that mark
discourse properties of the noun and are quite frequent; the is the most common
word in written English, with a and an right behind.
conjunction Conjunctions join two phrases, clauses, or sentences. Coordinating conjunc-
tions like and, or, and but join two elements of equal status. Subordinating conjunc-
tions are used when one of the elements has some embedded status. For example,
the subordinating conjunction that in “I thought that you might like some milk” links
the main clause I thought with the subordinate clause you might like some milk. This
clause is called subordinate because this entire clause is the “content” of the main
verb thought. Subordinating conjunctions like that which link a verb to its argument
complementizer in this way are also called complementizers.
pronoun Pronouns act as a shorthand for referring to an entity or event. Personal pro-
nouns refer to persons or entities (you, she, I, it, me, etc.). Possessive pronouns are
forms of personal pronouns that indicate either actual possession or more often just
an abstract relation between the person and some object (my, your, his, her, its, one’s,
wh our, their). Wh-pronouns (what, who, whom, whoever) are used in certain question
17.2 • PART- OF -S PEECH TAGGING 381
Below we show some examples with each word tagged according to both the UD
(in blue) and Penn (in red) tagsets. Notice that the Penn tagset distinguishes tense
and participles on verbs, and has a special tag for the existential there construction in
English. Note that since London Journal of Medicine is a proper noun, both tagsets
mark its component nouns as PROPN/NNP, including journal and medicine, which
might otherwise be labeled as common nouns (NOUN/NN).
(17.1) There/PRON/EX are/VERB/VBP 70/NUM/CD children/NOUN/NNS
there/ADV/RB ./PUNC/.
(17.2) Preliminary/ADJ/JJ findings/NOUN/NNS were/AUX/VBD
reported/VERB/VBN in/ADP/IN today/NOUN/NN ’s/PART/POS
London/PROPN/NNP Journal/PROPN/NNP of/ADP/IN Medicine/PROPN/NNP
y1 y2 y3 y4 y5
Figure 17.3 The task of part-of-speech tagging: mapping from input words x1 , x2 , ..., xn to
output POS tags y1 , y2 , ..., yn .
ambiguity thought that your flight was earlier). The goal of POS-tagging is to resolve these
resolution
ambiguities, choosing the proper tag for the context.
accuracy The accuracy of part-of-speech tagging algorithms (the percentage of test set
tags that match human gold labels) is extremely high. One study found accuracies
over 97% across 15 languages from the Universal Dependency (UD) treebank (Wu
and Dredze, 2019). Accuracies on various English treebanks are also 97% (no matter
the algorithm; HMMs, CRFs, BERT perform similarly). This 97% number is also
about the human performance on this task, at least for English (Manning, 2011).
We’ll introduce algorithms for the task in the next few sections, but first let’s
explore the task. Exactly how hard is it? Fig. 17.4 shows that most word types
(85-86%) are unambiguous (Janet is always NNP, hesitantly is always RB). But the
ambiguous words, though accounting for only 14-15% of the vocabulary, are very
common, and 55-67% of word tokens in running text are ambiguous. Particularly
ambiguous common words include that, back, down, put and set; here are some
examples of the 6 different parts of speech for the word back:
earnings growth took a back/JJ seat
a small building in the back/NN
a clear majority of senators back/VBP the bill
Dave began to back/VB toward the door
enable the country to buy back/RP debt
I was twenty-one back/RB then
Nonetheless, many words are easy to disambiguate, because their different tags
aren’t equally likely. For example, a can be a determiner or the letter a, but the
determiner sense is much more likely.
This idea suggests a useful baseline: given an ambiguous word, choose the tag
which is most frequent in the training corpus. This is a key concept:
Most Frequent Class Baseline: Always compare a classifier against a baseline at
least as good as the most frequent class baseline (assigning each token to the class
it occurred in most often in the training set).
17.3 • NAMED E NTITIES AND NAMED E NTITY TAGGING 383
Named entity tagging is a useful first step in lots of natural language processing
tasks. In sentiment analysis we might want to know a consumer’s sentiment toward a
particular entity. Entities are a useful first stage in question answering, or for linking
text to information in structured knowledge sources like Wikipedia. And named
entity tagging is also central to tasks involving building semantic representations,
like extracting events and the relationship between participants.
1 In English, on the WSJ corpus, tested on sections 22-24.
384 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
[PER Washington] was born into slavery on the farm of James Burroughs.
[ORG Washington] went up 2 games to 1 in the four-game series.
Blair arrived in [LOC Washington] for what may well be his last state visit.
In June, [GPE Washington] passed a primary seatbelt law.
Figure 17.6 Examples of type ambiguities in the use of the name Washington.
We’ve also shown two variant tagging schemes: IO tagging, which loses some
information by eliminating the B tag, and BIOES tagging, which adds an end tag
E for the end of a span, and a span tag S for a span consisting of only one word.
A sequence labeler (HMM, CRF, RNN, Transformer, etc.) is trained to label each
token in a text with tags that indicate the presence (or absence) of particular kinds
of named entities.
17.4 • HMM PART- OF -S PEECH TAGGING 385
.8
are .2
.1 COLD2 .1 .4 .5
.1 .5
.1
.3 uniformly charming
HOT1 WARM3 .5
.6 .3 .6 .1 .2
.6
(a) (b)
Figure 17.8 A Markov chain for weather (a) and one for words (b), showing states and
transitions. A start distribution π is required; setting π = [0.1, 0.7, 0.2] for (a) would mean a
probability 0.7 of starting in state 2 (cold), probability 0.1 of starting in state 1 (hot), etc.
C(ti−1 ,ti )
P(ti |ti−1 ) = (17.8)
C(ti−1 )
In the WSJ corpus, for example, MD occurs 13124 times of which it is followed
by VB 10471, for an MLE estimate of
C(MD,V B) 10471
P(V B|MD) = = = .80 (17.9)
C(MD) 13124
The B emission probabilities, P(wi |ti ), represent the probability, given a tag (say
MD), that it will be associated with a given word (say will). The MLE of the emis-
sion probability is
C(ti , wi )
P(wi |ti ) = (17.10)
C(ti )
Of the 13124 occurrences of MD in the WSJ corpus, it is associated with will 4046
times:
C(MD, will) 4046
P(will|MD) = = = .31 (17.11)
C(MD) 13124
We saw this kind of Bayesian modeling in Appendix K; recall that this likelihood
term is not asking “which is the most likely tag for the word will?” That would be
the posterior P(MD|will). Instead, P(will|MD) answers the slightly counterintuitive
question “If we were going to generate a MD, how likely is it that this modal would
be will?”
B2 a22
P("aardvark" | MD)
...
P(“will” | MD)
...
P("the" | MD)
...
MD2 B3
P(“back” | MD)
... a12 a32 P("aardvark" | NN)
P("zebra" | MD) ...
a11 a21 a33 P(“will” | NN)
a23 ...
P("the" | NN)
B1 a13 ...
P(“back” | NN)
P("aardvark" | VB)
...
VB1 a31
NN3 ...
P("zebra" | NN)
P(“will” | VB)
...
P("the" | VB)
...
P(“back” | VB)
...
P("zebra" | VB)
Figure 17.9 An illustration of the two parts of an HMM representation: the A transition
probabilities used to compute the prior probability, and the B observation likelihoods that are
associated with each state, one likelihood for each possible observation word.
388 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
For part-of-speech tagging, the goal of HMM decoding is to choose the tag
sequence t1 . . .tn that is most probable given the observation sequence of n words
w1 . . . wn :
tˆ1:n = argmax P(t1 . . .tn |w1 . . . wn ) (17.12)
t1 ... tn
The way we’ll do this in the HMM is to use Bayes’ rule to instead compute:
HMM taggers make two further simplifying assumptions. The first (output in-
dependence, from Eq. 17.7) is that the probability of a word appearing depends only
on its own tag and is independent of neighboring words and tags:
n
Y
P(w1 . . . wn |t1 . . .tn ) ≈ P(wi |ti ) (17.15)
i=1
The second assumption (the Markov assumption, Eq. 17.6) is that the probability of
a tag is dependent only on the previous tag, rather than the entire tag sequence;
n
Y
P(t1 . . .tn ) ≈ P(ti |ti−1 ) (17.16)
i=1
Plugging the simplifying assumptions from Eq. 17.15 and Eq. 17.16 into Eq. 17.14
results in the following equation for the most probable tag sequence from a bigram
tagger:
emission transition
n z }| { z }| {
Y
tˆ1:n = argmax P(t1 . . .tn |w1 . . . wn ) ≈ argmax P(wi |ti ) P(ti |ti−1 ) (17.17)
t1 ... tn t1 ... tn
i=1
The two parts of Eq. 17.17 correspond neatly to the B emission probability and A
transition probability that we just defined above!
17.4 • HMM PART- OF -S PEECH TAGGING 389
Figure 17.10 Viterbi algorithm for finding the optimal sequence of tags. Given an observation sequence and
an HMM λ = (A, B), the algorithm returns the state path through the HMM that assigns maximum likelihood
to the observation sequence.
We represent the most probable path by taking the maximum over all possible
previous state sequences max . Like other dynamic programming algorithms,
q1 ,...,qt−1
Viterbi fills each cell recursively. Given that we had already computed the probabil-
ity of being in every state at time t − 1, we compute the Viterbi probability by taking
the most probable of the extensions of the paths that lead to the current cell. For a
given state q j at time t, the value vt ( j) is computed as
N
vt ( j) = max vt−1 (i) ai j b j (ot ) (17.19)
i=1
The three factors that are multiplied in Eq. 17.19 for extending the previous paths to
compute the Viterbi probability at time t are
390 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
DT DT DT DT DT
RB RB RB RB RB
NN NN NN NN NN
JJ JJ JJ JJ JJ
VB VB VB VB VB
MD MD MD MD MD
vt−1 (i) the previous Viterbi path probability from the previous time step
ai j the transition probability from previous state qi to current state q j
b j (ot ) the state observation likelihood of the observation symbol ot given
the current state j
NNP MD VB JJ NN RB DT
<s > 0.2767 0.0006 0.0031 0.0453 0.0449 0.0510 0.2026
NNP 0.3777 0.0110 0.0009 0.0084 0.0584 0.0090 0.0025
MD 0.0008 0.0002 0.7968 0.0005 0.0008 0.1698 0.0041
VB 0.0322 0.0005 0.0050 0.0837 0.0615 0.0514 0.2231
JJ 0.0366 0.0004 0.0001 0.0733 0.4509 0.0036 0.0036
NN 0.0096 0.0176 0.0014 0.0086 0.1216 0.0177 0.0068
RB 0.0068 0.0102 0.1011 0.1012 0.0120 0.0728 0.0479
DT 0.1147 0.0021 0.0002 0.2157 0.4744 0.0102 0.0017
Figure 17.12 The A transition probabilities P(ti |ti−1 ) computed from the WSJ corpus with-
out smoothing. Rows are labeled with the conditioning event; thus P(V B|MD) is 0.7968.
<s > is the start token.
Let the HMM be defined by the two tables in Fig. 17.12 and Fig. 17.13. Fig-
ure 17.12 lists the ai j probabilities for transitioning between the hidden states (part-
of-speech tags). Figure 17.13 expresses the bi (ot ) probabilities, the observation
likelihoods of words given tags. This table is (slightly simplified) from counts in the
WSJ corpus. So the word Janet only appears as an NNP, back has 4 possible parts
17.4 • HMM PART- OF -S PEECH TAGGING 391
of speech, and the word the can appear as a determiner or as an NNP (in titles like
“Somewhere Over the Rainbow” all words are tagged as NNP).
v1(7) v2(7)
q7 DT
art)
D
q3 VB B|st
|J
=. =0 = 2.5e-13
* P
(MD
= 0 |VB)
v2(2) =
tart) v1(2)=
q2 MD D|s
P(M 0006 .0006 x 0 = * P(MD|M max * .308 =
= . D) 2.772e-8
0 =0
8 1 =)
.9 9*.0 NP
v1(1) =
00 D|N
v2(1)
.0 P(M
= .000009
*
= .28
backtrace
start start start start
start
π backtrace
Janet will
t back the bill
o1 o2 o3 o4 o5
Figure 17.14 The first few entries in the individual state columns for the Viterbi algorithm. Each cell keeps
the probability of the best path so far and a pointer to the previous cell along that path. We have only filled out
columns 1 and 2; to avoid clutter most cells with value 0 are left empty. The rest is left as an exercise for the
reader. After the cells are filled in, backtracing from the end state, we should be able to reconstruct the correct
state sequence NNP MD VB DT NN.
Figure 17.14 shows a fleshed-out version of the sketch we saw in Fig. 17.11,
the Viterbi lattice for computing the best hidden state sequence for the observation
sequence Janet will back the bill.
There are N = 5 state columns. We begin in column 1 (for the word Janet) by
setting the Viterbi value in each cell to the product of the π transition probability (the
start probability for that state i, which we get from the <s> entry of Fig. 17.12), and
392 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
the observation likelihood of the word Janet given the tag for that cell. Most of the
cells in the column are zero since the word Janet cannot be any of those tags. The
reader should find this in Fig. 17.14.
Next, each cell in the will column gets updated. For each state, we compute the
value viterbi[s,t] by taking the maximum over the extensions of all the paths from
the previous column that lead to the current cell according to Eq. 17.19. We have
shown the values for the MD, VB, and NN cells. Each cell gets the max of the 7 val-
ues from the previous column, multiplied by the appropriate transition probability;
as it happens in this case, most of them are zero from the previous column. The re-
maining value is multiplied by the relevant observation probability, and the (trivial)
max is taken. In this case the final value, 2.772e-8, comes from the NNP state at the
previous column. The reader should fill in the rest of the lattice in Fig. 17.14 and
backtrace to see whether or not the Viterbi algorithm returns the gold state sequence
NNP MD VB DT NN.
In a CRF, by contrast, we compute the posterior p(Y |X) directly, training the CRF
17.5 • C ONDITIONAL R ANDOM F IELDS (CRF S ) 393
However, the CRF does not compute a probability for each tag at each time step. In-
stead, at each time step the CRF computes log-linear functions over a set of relevant
features, and these local features are aggregated and normalized to produce a global
probability for the whole sequence.
Let’s introduce the CRF more formally, again using X and Y as the input and
output sequences. A CRF is a log-linear model that assigns a probability to an
entire output (tag) sequence Y , out of all possible sequences Y, given the entire input
(word) sequence X. We can think of a CRF as like a giant sequential version of
the multinomial logistic regression algorithm we saw for text categorization. Recall
that we introduced the feature function f in regular multinomial logistic regression
for text categorization as a function of a tuple: the input text x and a single class y
(page 71). In a CRF, we’re dealing with a sequence, so the function F maps an entire
input sequence X and an entire output sequence Y to a feature vector. Let’s assume
we have K features, with a weight wk for each feature Fk :
K
!
X
exp wk Fk (X,Y )
k=1
p(Y |X) = K
! (17.23)
X X
0
exp wk Fk (X,Y )
Y 0 ∈Y k=1
It’s common to also describe the same equation by pulling out the denominator into
a function Z(X):
K
!
1 X
p(Y |X) = exp wk Fk (X,Y ) (17.24)
Z(X)
k=1
K
!
X X
0
Z(X) = exp wk Fk (X,Y ) (17.25)
Y 0 ∈Y k=1
We’ll call these K functions Fk (X,Y ) global features, since each one is a property
of the entire input sequence X and output sequence Y . We compute them by decom-
posing into a sum of local features for each position i in Y :
n
X
Fk (X,Y ) = fk (yi−1 , yi , X, i) (17.26)
i=1
Each of these local features fk in a linear-chain CRF is allowed to make use of the
current output token yi , the previous output token yi−1 , the entire input string X (or
any subpart of it), and the current position i. This constraint to only depend on
the current and previous output tokens yi and yi−1 are what characterizes a linear
linear chain chain CRF. As we will see, this limitation makes it possible to use versions of the
CRF
efficient Viterbi and Forward-Backwards algorithms from the HMM. A general CRF,
by contrast, allows a feature to make use of any output token, and are thus necessary
for tasks in which the decision depend on distant output tokens, like yi−4 . General
CRFs require more complex inference, and are less commonly used for language
processing.
394 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
These templates automatically populate the set of features from every instance in
the training and test set. Thus for our example Janet/NNP will/MD back/VB the/DT
bill/NN, when xi is the word back, the following features would be generated and
have the value 1 (we’ve assigned them arbitrary feature numbers):
f3743 : yi = VB and xi = back
f156 : yi = VB and yi−1 = MD
f99732 : yi = VB and xi−1 = will and xi+2 = bill
It’s also important to have features that help with unknown words. One of the
word shape most important is word shape features, which represent the abstract letter pattern
of the word by mapping lower-case letters to ‘x’, upper-case to ‘X’, numbers to
’d’, and retaining punctuation. Thus for example I.M.F. would map to X.X.X. and
DC10-30 would map to XXdd-dd. A second class of shorter word shape features is
also used. In these features consecutive character types are removed, so words in all
caps map to X, words with initial-caps map to Xx, DC10-30 would be mapped to
Xd-d but I.M.F would still map to X.X.X. Prefix and suffix features are also useful.
In summary, here are some sample feature templates that help with unknown words:
For example the word well-dressed might generate the following non-zero val-
ued feature values:
2 Because in HMMs all computation is based on the two probabilities P(tag|tag) and P(word|tag), if
we want to include some source of knowledge into the tagging process, we must find a way to encode
the knowledge into one of these two probabilities. Each time we add a feature we have to do a lot of
complicated conditioning which gets harder and harder as we have more and more such features.
17.5 • C ONDITIONAL R ANDOM F IELDS (CRF S ) 395
prefix(xi ) = w
prefix(xi ) = we
suffix(xi ) = ed
suffix(xi ) = d
word-shape(xi ) = xxxx-xxxxxxx
short-word-shape(xi ) = x-x
The known-word templates are computed for every word seen in the training
set; the unknown word features can also be computed for all words in training, or
only on training words whose frequency is below some threshold. The result of the
known-word templates and word-signature features is a very large set of features.
Generally a feature cutoff is used in which features are thrown out if they have count
< 5 in the training set.
Remember that in a CRF we don’t learn weights for each of these local features
fk . Instead, we first sum the values of each local feature (for example feature f3743 )
over the entire sentence, to create each global feature (for example F3743 ). It is those
global features that will then be multiplied by weight w3743 . Thus for training and
inference there is always a fixed set of K features with K weights, even though the
length of each sentence is different.
gazetteer One feature that is especially useful for locations is a gazetteer, a list of place
names, often providing millions of entries for locations with detailed geographical
and political information.3 This can be implemented as a binary feature indicating a
phrase appears in the list. Other related resources like name-lists, for example from
the United States Census Bureau4 , can be used, as can other entity dictionaries like
lists of corporations or products, although they may not be as helpful as a gazetteer
(Mikheev et al., 1999).
The sample named entity token L’Occitane would generate the following non-
zero valued feature values (assuming that L’Occitane is neither in the gazetteer nor
the census).
3 [Link]
4 [Link]
396 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
We can ignore the exp function and the denominator Z(X), as we do above, because
exp doesn’t change the argmax, and the denominator Z(X) is constant for a given
observation sequence X.
How should we decode to find this optimal tag sequence ŷ? Just as with HMMs,
we’ll turn to the Viterbi algorithm, which works because, like the HMM, the linear-
chain CRF depends at each timestep on only one previous output token yi−1 .
Concretely, this involves filling an N ×T array with the appropriate values, main-
taining backpointers as we proceed. As with HMM Viterbi, when the table is filled,
we simply follow pointers back from the maximum value in the final column to
retrieve the desired set of labels.
17.6 • E VALUATION OF NAMED E NTITY R ECOGNITION 397
The requisite changes from HMM Viterbi have to do only with how we fill each
cell. Recall from Eq. 17.19 that the recursive step of the Viterbi equation computes
the Viterbi value of time t for state j as
N
vt ( j) = max vt−1 (i) ai j b j (ot ); 1 ≤ j ≤ N, 1 < t ≤ T (17.31)
i=1
The CRF requires only a slight change to this latter formula, replacing the a and b
prior and likelihood probabilities with the CRF features:
" K
#
N X
vt ( j) = max vt−1 (i) + wk fk (yt−1 , yt , X,t) 1 ≤ j ≤ N, 1 < t ≤ T (17.33)
i=1
k=1
presented are supervised, having labeled data is essential for training and testing. A
wide variety of datasets exist for part-of-speech tagging and/or NER. The Universal
Dependencies (UD) dataset (de Marneffe et al., 2021) has POS tagged corpora in
over a hundred languages, as do the Penn Treebanks in English, Chinese, and Arabic.
OntoNotes has corpora labeled for named entities in English, Chinese, and Arabic
(Hovy et al., 2006). Named entity tagged corpora are also available in particular
domains, such as for biomedical (Bada et al., 2012) and literary text (Bamman et al.,
2019).
guages need to label words with case and gender information. Tagsets for morpho-
logically rich languages are therefore sequences of morphological tags rather than a
single primitive tag. Here’s a Turkish example, in which the word izin has three pos-
sible morphological/part-of-speech tags and meanings (Hakkani-Tür et al., 2002):
1. Yerdeki izin temizlenmesi gerek. iz + Noun+A3sg+Pnon+Gen
The trace on the floor should be cleaned.
17.8 Summary
This chapter introduced parts of speech and named entities, and the tasks of part-
of-speech tagging and named entity recognition:
• Languages generally have a small set of closed class words that are highly
frequent, ambiguous, and act as function words, and open-class words like
nouns, verbs, adjectives. Various part-of-speech tagsets exist, of between 40
and 200 tags.
• Part-of-speech tagging is the process of assigning a part-of-speech label to
each of a sequence of words.
• Named entities are words for proper nouns referring mainly to people, places,
and organizations, but extended to many other types that aren’t strictly entities
or even proper nouns.
• Two common approaches to sequence modeling are a generative approach,
HMM tagging, and a discriminative approach, CRF tagging. We will see a
neural approach in following chapters.
• The probabilities in HMM taggers are estimated by maximum likelihood es-
timation on tag-labeled training corpora. The Viterbi algorithm is used for
decoding, finding the most likely tag sequence
• Conditional Random Fields or CRF taggers train a log-linear model that can
choose the best tag sequence given an observation sequence, based on features
that condition on the output tag, the prior output tag, the entire input sequence,
and the current timestep. They use the Viterbi algorithm for inference, to
choose the best sequence of tags, and a version of the Forward-Backward
algorithm (see Appendix A) for training,
400 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
Historical Notes
What is probably the earliest part-of-speech tagger was part of the parser in Zellig
Harris’s Transformations and Discourse Analysis Project (TDAP), implemented be-
tween June 1958 and July 1959 at the University of Pennsylvania (Harris, 1962),
although earlier systems had used part-of-speech dictionaries. TDAP used 14 hand-
written rules for part-of-speech disambiguation; the use of part-of-speech tag se-
quences and the relative frequency of tags for a word prefigures modern algorithms.
The parser was implemented essentially as a cascade of finite-state transducers; see
Joshi and Hopely (1999) and Karttunen (1999) for a reimplementation.
The Computational Grammar Coder (CGC) of Klein and Simmons (1963) had
three components: a lexicon, a morphological analyzer, and a context disambigua-
tor. The small 1500-word lexicon listed only function words and other irregular
words. The morphological analyzer used inflectional and derivational suffixes to as-
sign part-of-speech classes. These were run over words to produce candidate parts
of speech which were then disambiguated by a set of 500 context rules by relying on
surrounding islands of unambiguous words. For example, one rule said that between
an ARTICLE and a VERB, the only allowable sequences were ADJ-NOUN, NOUN-
ADVERB, or NOUN-NOUN. The TAGGIT tagger (Greene and Rubin, 1971) used
the same architecture as Klein and Simmons (1963), with a bigger dictionary and
more tags (87). TAGGIT was applied to the Brown corpus and, according to Francis
and Kučera (1982, p. 9), accurately tagged 77% of the corpus; the remainder of the
Brown corpus was then tagged by hand. All these early algorithms were based on
a two-stage architecture in which a dictionary was first used to assign each word a
set of potential parts of speech, and then lists of handwritten disambiguation rules
winnowed the set down to a single part of speech per word.
Probabilities were used in tagging by Stolz et al. (1965) and a complete proba-
bilistic tagger with Viterbi decoding was sketched by Bahl and Mercer (1976). The
Lancaster-Oslo/Bergen (LOB) corpus, a British English equivalent of the Brown cor-
pus, was tagged in the early 1980’s with the CLAWS tagger (Marshall 1983; Mar-
shall 1987; Garside 1987), a probabilistic algorithm that approximated a simplified
HMM tagger. The algorithm used tag bigram probabilities, but instead of storing the
word likelihood of each tag, the algorithm marked tags either as rare (P(tag|word) <
.01) infrequent (P(tag|word) < .10) or normally frequent (P(tag|word) > .10).
DeRose (1988) developed a quasi-HMM algorithm, including the use of dy-
namic programming, although computing P(t|w)P(w) instead of P(w|t)P(w). The
same year, the probabilistic PARTS tagger of Church 1988, 1989 was probably the
first implemented HMM tagger, described correctly in Church (1989), although
Church (1988) also described the computation incorrectly as P(t|w)P(w) instead
of P(w|t)P(w). Church (p.c.) explained that he had simplified for pedagogical pur-
poses because using the probability P(t|w) made the idea seem more understandable
as “storing a lexicon in an almost standard form”.
Later taggers explicitly introduced the use of the hidden Markov model (Kupiec
1992; Weischedel et al. 1993; Schütze and Singer 1994). Merialdo (1994) showed
that fully unsupervised EM didn’t work well for the tagging task and that reliance
on hand-labeled data was important. Charniak et al. (1993) showed the importance
of the most frequent tag baseline; the 92.3% number we give above was from Abney
et al. (1999). See Brants (2000) for HMM tagger implementation details, includ-
ing the extension to trigram contexts, and the use of sophisticated unknown word
features; its performance is still close to state of the art taggers.
E XERCISES 401
Exercises
17.1 Find one tagging error in each of the following sentences that are tagged with
the Penn Treebank tagset:
1. I/PRP need/VBP a/DT flight/NN from/IN Atlanta/NN
2. Does/VBZ this/DT flight/NN serve/VB dinner/NNS
3. I/PRP have/VB a/DT friend/NN living/VBG in/IN Denver/NNP
4. Can/VBP you/PRP list/VB the/DT nonstop/JJ afternoon/NN flights/NNS
17.2 Use the Penn Treebank tagset to tag each word in the following sentences
from Damon Runyon’s short stories. You may ignore punctuation. Some of
these are quite difficult; do your best.
1. It is a nice night.
2. This crap game is over a garage in Fifty-second Street. . .
3. . . . Nobody ever takes the newspapers she sells . . .
4. He is a tall, skinny guy with a long, sad, mean-looking kisser, and a
mournful voice.
402 C HAPTER 17 • S EQUENCE L ABELING FOR PARTS OF S PEECH AND NAMED E NTITIES
The study of grammar has an ancient pedigree. The grammar of Sanskrit was
described by the Indian grammarian Pā[Link] sometime between the 7th and 4th cen-
syntax turies BCE, in his famous treatise the As.t.ādhyāyı̄ (‘8 books’). And our word syntax
comes from the Greek sýntaxis, meaning “setting out together or arrangement”, and
refers to the way words are arranged together. We have seen syntactic notions in pre-
vious chapters like the use of part-of-speech categories (Chapter 17). In this chapter
and the next one we introduce formal models for capturing more sophisticated no-
tions of grammatical structure and algorithms for parsing these structures.
Our focus in this chapter is context-free grammars and the CKY algorithm
for parsing them. Context-free grammars are the backbone of many formal mod-
els of the syntax of natural language (and, for that matter, of computer languages).
Syntactic parsing is the task of assigning a syntactic structure to a sentence. Parse
trees (whether for context-free grammars or for the dependency or CCG formalisms
we introduce in following chapters) can be used in applications such as grammar
checking: sentence that cannot be parsed may have grammatical errors (or at least
be hard to read). Parse trees can be an intermediate stage of representation for for-
mal semantic analysis. And parsers and the grammatical structure they assign a
sentence are a useful text analysis tool for text data science applications that require
modeling the relationship of elements in sentences.
In this chapter we introduce context-free grammars, give a small sample gram-
mar of English, introduce more formal definitions of context-free grammars and
grammar normal form, and talk about treebanks: corpora that have been anno-
tated with syntactic structure. We then discuss parse ambiguity and the problems
it presents, and turn to parsing itself, giving the famous Cocke-Kasami-Younger
(CKY) algorithm (Kasami 1965, Younger 1967), the standard dynamic program-
ming approach to syntactic parsing. The CKY algorithm returns an efficient repre-
sentation of the set of parse trees for a sentence, but doesn’t tell us which parse tree
is the right one. For that, we need to augment CKY with scores for each possible
constituent. We’ll see how to do this with neural span-based parsers. Finally, we’ll
introduce the standard set of metrics for evaluating parser accuracy.
404 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
18.1 Constituency
Syntactic constituency is the idea that groups of words can behave as single units,
or constituents. Part of developing a grammar involves building an inventory of the
constituents in the language. How do words group together in English? Consider
noun phrase the noun phrase, a sequence of words surrounding at least one noun. Here are some
examples of noun phrases (thanks to Damon Runyon):
What evidence do we have that these words group together (or “form constituents”)?
One piece of evidence is that they can all appear in similar syntactic environments,
for example, before a verb.
But while the whole noun phrase can occur before a verb, this is not true of each
of the individual words that make up a noun phrase. The following are not grammat-
ical sentences of English (recall that we use an asterisk (*) to mark fragments that
are not grammatical English sentences):
Thus, to correctly describe facts about the ordering of these words in English, we
must be able to say things like “Noun Phrases can occur before verbs”. Let’s now
see how to do this in a more formal way!
more Nouns.1
NP → Det Nominal
NP → ProperNoun
Nominal → Noun | Nominal Noun
Context-free rules can be hierarchically embedded, so we can combine the previous
rules with others, like the following, that express facts about the lexicon:
Det → a
Det → the
Noun → flight
The symbols that are used in a CFG are divided into two classes. The symbols
terminal that correspond to words in the language (“the”, “nightclub”) are called terminal
symbols; the lexicon is the set of rules that introduce these terminal symbols. The
non-terminal symbols that express abstractions over these terminals are called non-terminals. In
each context-free rule, the item to the right of the arrow (→) is an ordered list of one
or more terminals and non-terminals; to the left of the arrow is a single non-terminal
symbol expressing some cluster or generalization. The non-terminal associated with
each word in the lexicon is its lexical category, or part of speech.
A CFG can be thought of in two ways: as a device for generating sentences
and as a device for assigning a structure to a given sentence. Viewing a CFG as a
generator, we can read the → arrow as “rewrite the symbol on the left with the string
of symbols on the right”.
So starting from the symbol: NP
we can use our first rule to rewrite NP as: Det Nominal
and then rewrite Nominal as: Noun
and finally rewrite these parts-of-speech as: a flight
We say the string a flight can be derived from the non-terminal NP. Thus, a CFG
can be used to generate a set of strings. This sequence of rule expansions is called a
derivation derivation of the string of words. It is common to represent a derivation by a parse
parse tree tree (commonly shown inverted with the root at the top). Figure 18.1 shows the tree
representation of this derivation.
NP
Det Nom
a Noun
flight
dominates In the parse tree shown in Fig. 18.1, we can say that the node NP dominates
all the nodes in the tree (Det, Nom, Noun, a, flight). We can say further that it
immediately dominates the nodes Det and Nom.
The formal language defined by a CFG is the set of strings that are derivable
start symbol from the designated start symbol. Each grammar must have one designated start
1 When talking about these rules we can pronounce the rightarrow → as “goes to”, and so we might
read the first rule above as “NP goes to Det Nominal”.
406 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
symbol, which is often called S. Since context-free grammars are often used to define
sentences, S is usually interpreted as the “sentence” node, and the set of strings that
are derivable from S is the set of sentences in some simplified version of English.
Let’s add a few additional rules to our inventory. The following rule expresses
verb phrase the fact that a sentence can consist of a noun phrase followed by a verb phrase:
Or the verb phrase may have a verb followed by a prepositional phrase alone:
The NP inside a PP need not be a location; PPs are often used with times and
dates, and with other nouns as well; they can be arbitrarily complex. Here are ten
examples from the ATIS corpus:
to Seattle on these flights
in Minneapolis about the ground transportation in Chicago
on Wednesday of the round trip flight on United Airlines
in the evening of the AP fifty seven flight
on the ninth of July with a stopover in Nashville
Figure 18.2 gives a sample lexicon, and Fig. 18.3 summarizes the grammar rules
we’ve seen so far, which we’ll call L0 . Note that we can use the or-symbol | to
indicate that a non-terminal has alternate possible expansions.
NP → Pronoun I
| Proper-Noun Los Angeles
| Det Nominal a + flight
Nominal → Nominal Noun morning + flight
| Noun flights
VP → Verb do
| Verb NP want + a flight
| Verb NP PP leave + Boston + in the morning
| Verb PP leaving + on Thursday
NP VP
Pro Verb NP
a Nom Noun
Noun flight
morning
Figure 18.4 The parse tree for “I prefer a morning flight” according to grammar L0 .
I), and a random expansion of VP (let’s say, to Verb NP), and so on until we generate
the string I prefer a morning flight. Figure 18.4 shows a parse tree that represents a
complete derivation of I prefer a morning flight.
We can also represent a parse tree in a more compact format called bracketed
bracketed notation; here is the bracketed representation of the parse tree of Fig. 18.4:
notation
(18.1) [S [NP [Pro I]] [VP [V prefer] [NP [Det a] [Nom [N morning] [Nom [N flight]]]]]]
A CFG like that of L0 defines a formal language. Sentences (strings of words)
that can be derived by a grammar are in the formal language defined by that gram-
grammatical mar, and are called grammatical sentences. Sentences that cannot be derived by
a given formal grammar are not in the language defined by that grammar and are
ungrammatical referred to as ungrammatical. This hard line between “in” and “out” characterizes
all formal languages but is only a very simplified model of how natural languages
really work. This is because determining whether a given sentence is part of a given
natural language (say, English) often depends on the context. In linguistics, the use
generative
grammar of formal languages to model natural languages is called generative grammar since
the language is defined by the set of possible sentences “generated” by the grammar.
(Note that this is a different sense of the word ‘generate’ than when we talk about
408 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
For the remainder of the book we adhere to the following conventions when dis-
cussing the formal properties of context-free grammars (as opposed to explaining
particular facts about English or other languages).
Capital letters like A, B, and S Non-terminals
S The start symbol
Lower-case Greek letters like α, β , and γ Strings drawn from (Σ ∪ N)∗
Lower-case Roman letters like u, v, and w Strings of terminals
A language is defined through the concept of derivation. One string derives an-
other one if it can be rewritten as the second one by some series of rule applications.
More formally, following Hopcroft and Ullman (1979),
if A → β is a production of R and α and γ are any strings in the set
directly derives (Σ ∪ N)∗ , then we say that αAγ directly derives αβ γ, or αAγ ⇒ αβ γ.
Derivation is then a generalization of direct derivation:
Let α1 , α2 , . . . , αm be strings in (Σ ∪ N)∗ , m ≥ 1, such that
α1 ⇒ α2 , α2 ⇒ α3 , . . . , αm−1 ⇒ αm
∗
derives We say that α1 derives αm , or α1 ⇒ αm .
We can then formally define the language LG generated by a grammar G as the
set of strings composed of terminal symbols that can be derived from the designated
start symbol S.
∗
LG = {w|w is in Σ∗ and S ⇒ w}
The problem of mapping from a string of words to its parse tree is called syn-
syntactic
parsing tactic parsing, as we’ll see in Section 18.6.
18.3 Treebanks
treebank A corpus in which every sentence is annotated with a parse tree is called a treebank.
18.3 • T REEBANKS 409
((S
(NP-SBJ (DT That) ((S
(JJ cold) (, ,) (NP-SBJ The/DT flight/NN )
(JJ empty) (NN sky) ) (VP should/MD
(VP (VBD was) (VP arrive/VB
(ADJP-PRD (JJ full) (PP-TMP at/IN
(PP (IN of) (NP eleven/CD a.m/RB ))
(NP (NN fire) (NP-TMP tomorrow/NN )))))
(CC and)
(NN light) ))))
(. .) ))
(a) (b)
Figure 18.5 Parses from the LDC Treebank3 for (a) Brown and (b) ATIS sentences.
NP-SBJ VP .
DT JJ , JJ NN VBD ADJP-PRD .
full IN NP
of NN CC NN
Grammar Lexicon
S → NP VP . DT → the | that
S → NP VP JJ → cold | empty | full
NP → CD RB NN → sky | fire | light | flight | tomorrow
NP → DT NN CC → and
NP → NN CC NN IN → of | at
NP → DT JJ , JJ NN CD → eleven
NP → NN RB → a.m.
VP → MD VP VB → arrive
VP → VBD ADJP VBD → was | said
VP → MD VP MD → should | would
VP → VB PP NP
ADJP → JJ PP
PP → IN NP
Figure 18.7 CFG grammar rules and lexicon from the treebank sentences in Fig. 18.5.
among the approximately 4,500 different rules for expanding VPs are separate rules
for PP sequences of any length and every possible arrangement of verb arguments:
VP → VBD PP
VP → VBD PP PP
VP → VBD PP PP PP
VP → VBD PP PP PP PP
VP → VB ADVP PP
VP → VB PP ADVP
VP → ADVP VB PP
A → B C D
can be converted into the following two CNF rules (Exercise 18.1 asks the reader to
18.5 • A MBIGUITY 411
Grammar Lexicon
S → NP VP Det → that | this | the | a
S → Aux NP VP Noun → book | flight | meal | money
S → VP Verb → book | include | prefer
NP → Pronoun Pronoun → I | she | me
NP → Proper-Noun Proper-Noun → Houston | United
NP → Det Nominal Aux → does
Nominal → Noun Preposition → from | to | on | near | through
Nominal → Nominal Noun
Nominal → Nominal PP
VP → Verb
VP → Verb NP
VP → Verb NP PP
VP → Verb PP
VP → VP PP
PP → Preposition NP
Figure 18.8 The L1 miniature English grammar and lexicon.
A → B X
X → C D
Sometimes using binary branching can actually produce smaller grammars. For
example, the sentences that might be characterized as
VP -> VBD NP PP*
are represented in the Penn Treebank by this series of rules:
VP → VBD NP PP
VP → VBD NP PP PP
VP → VBD NP PP PP PP
VP → VBD NP PP PP PP PP
...
but could also be generated by the following two-rule grammar:
VP → VBD NP PP
VP → VP PP
The generation of a symbol A with a potentially infinite sequence of symbols B with
Chomsky-
adjunction a rule of the form A → A B is known as Chomsky-adjunction.
18.5 Ambiguity
Ambiguity is the most serious problem faced by syntactic parsers. Chapter 17 intro-
duced the notions of part-of-speech ambiguity and part-of-speech disambigua-
structural
ambiguity tion. Here, we introduce a new kind of ambiguity, called structural ambiguity,
illustrated with a new toy grammar L1 , shown in Figure 18.8, which adds a few
rules to the L0 grammar.
Structural ambiguity occurs when the grammar can assign more than one parse
to a sentence. Groucho Marx’s well-known line as Captain Spaulding in Animal
412 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
S S
NP VP NP VP
elephant elephant
Figure 18.9 Two parse trees for an ambiguous sentence. The parse on the left corresponds to the humorous
reading in which the elephant is in the pajamas, the parse on the right corresponds to the reading in which
Captain Spaulding did the shooting in his pajamas.
will deliver tomorrow night to the American people could be an adjunct modifying
the verb pushed. A PP like over nationwide television and radio could be attached
to any of the higher VPs or NPs (e.g., it could modify people or night).
The fact that there are many grammatically correct but semantically unreason-
able parses for naturally occurring sentences is an irksome problem that affects all
parsers. Fortunately, the CKY algorithm below is designed to efficiently handle
structural ambiguities. And as we’ll see in the following section, we can augment
CKY with neural methods to choose a single correct parse by syntactic disambigua-
syntactic
disambiguation tion.
is a non-unit production in our grammar, then we add A → γ for each such rule in
the grammar and discard all the intervening unit productions. As we demonstrate
with our toy grammar, this can lead to a substantial flattening of the grammar and a
consequent promotion of terminals to fairly high levels in the resulting trees.
Rules with right-hand sides longer than 2 are normalized through the introduc-
tion of new non-terminals that spread the longer sequences over several new rules.
Formally, if we have a rule like
A → BCγ
we replace the leftmost pair of non-terminals with a new non-terminal and introduce
a new production, resulting in the following new rules:
A → X1 γ
X1 → B C
In the case of longer right-hand sides, we simply iterate this process until the of-
fending rule has been replaced by rules of length 2. The choice of replacing the
leftmost pair of non-terminals is purely arbitrary; any systematic scheme that results
in binary rules would suffice.
In our current grammar, the rule S → Aux NP VP would be replaced by the two
rules S → X1 VP and X1 → Aux NP.
The entire conversion process can be summarized as follows:
1. Copy all conforming rules to the new grammar unchanged.
2. Convert terminals within rules to dummy non-terminals.
3. Convert unit productions.
4. Make all rules binary and add them to new grammar.
Figure 18.10 shows the results of applying this entire conversion procedure to
the L1 grammar introduced earlier on page 411. Note that this figure doesn’t show
the original lexical rules; since these original lexical rules are already in CNF, they
all carry over unchanged to the new grammar. Figure 18.10 does, however, show
the various places where the process of eliminating unit productions has, in effect,
created new lexical rules. For example, all the original verbs have been promoted to
both VPs and to Ss in the converted grammar.
L1 Grammar L1 in CNF
S → NP VP S → NP VP
S → Aux NP VP S → X1 VP
X1 → Aux NP
S → VP S → book | include | prefer
S → Verb NP
S → X2 PP
S → Verb PP
S → VP PP
NP → Pronoun NP → I | she | me
NP → Proper-Noun NP → United | Houston
NP → Det Nominal NP → Det Nominal
Nominal → Noun Nominal → book | flight | meal | money
Nominal → Nominal Noun Nominal → Nominal Noun
Nominal → Nominal PP Nominal → Nominal PP
VP → Verb VP → book | include | prefer
VP → Verb NP VP → Verb NP
VP → Verb NP PP VP → X2 PP
X2 → Verb NP
VP → Verb PP VP → Verb PP
VP → VP PP VP → VP PP
PP → Preposition NP PP → Preposition NP
Figure 18.10 L1 Grammar and its conversion to CNF. Note that although they aren’t shown
here, all the original lexical entries from L1 carry over unchanged as well.
a position k, the first constituent [i, k] must lie to the left of entry [i, j] somewhere
along row i, and the second entry [k, j] must lie beneath it, along column j.
To make this more concrete, consider the following example with its completed
parse matrix, shown in Fig. 18.11.
(18.4) Book the flight through Houston.
The superdiagonal row in the matrix contains the parts of speech for each word in
the input. The subsequent diagonals above that superdiagonal contain constituents
that cover all the spans of increasing length in the input.
Given this setup, CKY recognition consists of filling the parse table in the right
way. To do this, we’ll proceed in a bottom-up fashion so that at the point where we
are filling any cell [i, j], the cells containing the parts that could contribute to this
entry (i.e., the cells to the left and the cells below) have already been filled. The
algorithm given in Fig. 18.12 fills the upper-triangular matrix a column at a time
working from left to right, with each column filled from bottom to top, as the right
side of Fig. 18.11 illustrates. This scheme guarantees that at each point in time we
have all the information we need (to the left, since all the columns to the left have
already been filled, and below since we’re filling bottom to top). It also mirrors on-
line processing, since filling the columns from left to right corresponds to processing
each word one at a time.
The outermost loop of the algorithm given in Fig. 18.12 iterates over the columns,
and the second loop iterates over the rows, from the bottom up. The purpose of the
innermost loop is to range over all the places where a substring spanning i to j in
the input might be split in two. As k ranges over the places where the string can be
split, the pairs of cells we consider move, in lockstep, to the right along row i and
down along column j. Figure 18.13 illustrates the general case of filling cell [i, j].
416 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
[3,4] [3,5]
NP,
Proper-
Noun
[4,5]
Figure 18.11 Completed parse table for Book the flight through Houston.
At each such split, the algorithm considers whether the contents of the two cells can
be combined in a way that is sanctioned by a rule in the grammar. If such a rule
exists, the non-terminal on its left-hand side is entered into the table.
Figure 18.14 shows how the five cells of column 5 of the table are filled after the
word Houston is read. The arrows point out the two spans that are being used to add
an entry to the table. Note that the action in cell [0, 5] indicates the presence of three
alternative parses for this input, one where the PP modifies the flight, one where
it modifies the booking, and one that captures the second argument in the original
VP → Verb NP PP rule, now captured indirectly with the VP → X2 PP rule.
[0,1] [0,n]
...
[i,j]
[i,i+1] [i,i+2]
... [i,j-2] [i,j-1]
[i+1,j]
[i+2,j]
[j-2,j]
[j-1,j]
...
[n-1, n]
Figure 18.13 All the ways to fill the [i, j]th cell in the CKY table.
parse consists of choosing an S from cell [0, n] and then recursively retrieving its
component constituents from the table. Of course, instead of returning every parse
for a sentence, we usually want just the best parse; we’ll see how to do that in the
next section.
Book the flight through Houston Book the flight through Houston
Book the flight through Houston Book the flight through Houston
[3,4] [3,5]
NP,
Proper-
Noun
[4,5]
Figure 18.14 Filling the cells of column 5 after reading the word Houston.
18.7 • S PAN -BASED N EURAL C ONSTITUENCY PARSING 419
0 1 2 3 4 5
postprocessing layers
map back to words
ENCODER
map to subwords
Fig. 18.15 sketches the architecture. The input word tokens are embedded by
420 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
passing them through a pretrained language model like BERT. Because BERT oper-
ates on the level of subword (wordpiece) tokens rather than words, we’ll first need to
convert the BERT outputs to word representations. One standard way of doing this
is to simply use the first subword unit as the representation for the entire word; us-
ing the last subword unit, or the sum of all the subword units are also common. The
embeddings can then be passed through some postprocessing layers; Kitaev et al.
(2019), for example, use 8 Transformer layers.
The resulting word encoder outputs yt are then used to compute a span score.
First, we must map the word encodings (indexed by word positions) to span encod-
ings (indexed by fenceposts). We do this by representing each fencepost with two
separate values; the intuition is that a span endpoint to the right of a word represents
different information than a span endpoint to the left of a word. We convert each
word output yt into a (leftward-pointing) value for spans ending at this fencepost,
←−
y t , and a (rightward-pointing) value → −y t for spans beginning at this fencepost, by
splitting yt into two halves. Each span then stretches from one double-vector fence-
post to another, as in the following representation of the flight, which is span(1, 3):
span(1,3)
v(i, j) = [→
−
yj −→
−
yi ; ← −− − ←
y j+1 −−]
yi+1 (18.5)
The span vector v is then passed through an MLP span classifier, with two fully-
connected layers and one ReLU activation function, whose output dimensionality is
the number of possible non-terminal labels:
T = {(it , jt , lt ) : t = 1, . . . , |T |} (18.7)
Thus once we have a score for each span, the parser can compute a score for the
whole tree s(T ) simply by summing over the scores of its constituent spans:
X
s(T ) = s(i, j, l) (18.8)
(i, j,l)∈T
18.8 • E VALUATING PARSERS 421
And we can choose the final parse tree as the tree with the maximum score:
The simplest method to produce the most likely parse is to greedily choose the
highest scoring label for each span. This greedy method is not guaranteed to produce
a tree, since the best label for a span might not fit into a complete tree. In practice,
however, the greedy method tends to find trees; in their experiments Gaddy et al.
(2018) finds that 95% of predicted bracketings form valid trees.
Nonetheless it is more common to use a variant of the CKY algorithm to find the
full parse. The variant defined in Gaddy et al. (2018) works as follows. Let’s define
sbest (i, j) as the score of the best subtree spanning (i, j). For spans of length one, we
choose the best label:
Note that the parser is using the max label for span (i, j) + the max labels for spans
(i, k) and (k, j) without worrying about whether those decisions make sense given a
grammar. The role of the grammar in classical parsing is to help constrain possible
combinations of constituents (NPs like to be followed by VPs). By contrast, the
neural model seems to learn these kinds of contextual constraints during its mapping
from spans to non-terminals.
For more details on span-based parsing, including the margin-based training al-
gorithm, see Stern et al. (2017), Gaddy et al. (2018), Kitaev and Klein (2018), and
Kitaev et al. (2019).
S(dumped)
NP(workers) VP(dumped)
a bin
Figure 18.16 A lexicalized tree from Collins (1999).
cross-brackets: the number of constituents for which the reference parse has a
bracketing such as ((A B) C) but the hypothesis parse has a bracketing such
as (A (B C)).
For comparing parsers that use different grammars, the PARSEVAL metric in-
cludes a canonicalization algorithm for removing information likely to be grammar-
specific (auxiliaries, pre-infinitival “to”, etc.) and for computing a simplified score
(Black et al., 1991). The canonical implementation of the PARSEVAL metrics is
evalb called evalb (Sekine and Collins, 1997).
most phrases. (Should the complementizer to or the verb be the head of an infinite
verb phrase?) Modern linguistic theories of syntax generally include a component
that defines heads (see, e.g., (Pollard and Sag, 1994)).
An alternative approach to finding a head is used in most practical computational
systems. Instead of specifying head rules in the grammar itself, heads are identified
dynamically in the context of trees for specific sentences. In other words, once
a sentence is parsed, the resulting tree is walked to decorate each node with the
appropriate head. Most current systems rely on a simple set of handwritten rules,
such as a practical one for Penn Treebank grammars given in Collins (1999) but
developed originally by Magerman (1995). For example, the rule for finding the
head of an NP is as follows (Collins, 1999, p. 238):
Selected other rules from this set are shown in Fig. 18.17. For example, for VP
rules of the form VP → Y1 · · · Yn , the algorithm would start from the left of Y1 · · ·
Yn looking for the first Yi of type TO; if no TOs are found, it would search for the
first Yi of type VBD; if no VBDs are found, it would search for a VBN, and so on.
See Collins (1999) for more details.
18.10 Summary
This chapter introduced constituency parsing. Here’s a summary of the main points:
• In many languages, groups of consecutive words act as a group or a con-
stituent, which can be modeled by context-free grammars (which are also
known as phrase-structure grammars).
• A context-free grammar consists of a set of rules or productions, expressed
over a set of non-terminal symbols and a set of terminal symbols. Formally,
a particular context-free language is the set of strings that can be derived
from a particular context-free grammar.
424 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
Historical Notes
According to Percival (1976), the idea of breaking up a sentence into a hierarchy of
constituents appeared in the Völkerpsychologie of the groundbreaking psychologist
Wilhelm Wundt (Wundt, 1900):
...den sprachlichen Ausdruck für die willkürliche Gliederung einer Ge-
sammtvorstellung in ihre in logische Beziehung zueinander gesetzten
Bestandteile
[the linguistic expression for the arbitrary division of a total idea
into its constituent parts placed in logical relations to one another]
Wundt’s idea of constituency was taken up into linguistics by Leonard Bloom-
field in his early book An Introduction to the Study of Language (Bloomfield, 1914).
By the time of his later book, Language (Bloomfield, 1933), what was then called
“immediate-constituent analysis” was a well-established method of syntactic study
in the United States. By contrast, traditional European grammar, dating from the
Classical period, defined relations between words rather than constituents, and Eu-
ropean syntacticians retained this emphasis on such dependency grammars, the sub-
ject of Chapter 19. (And indeed, both dependency and constituency grammars have
been in vogue in computational linguistics at different times).
American Structuralism saw a number of specific definitions of the immediate
constituent, couched in terms of their search for a “discovery procedure”: a method-
ological algorithm for describing the syntax of a language. In general, these attempt
to capture the intuition that “The primary criterion of the immediate constituent
is the degree in which combinations behave as simple units” (Bazell, 1952/1966, p.
284). The most well known of the specific definitions is Harris’ idea of distributional
similarity to individual units, with the substitutability test. Essentially, the method
proceeded by breaking up a construction into constituents by attempting to substitute
simple structures for possible constituents—if a substitution of a simple form, say,
E XERCISES 425
man, was substitutable in a construction for a more complex set (like intense young
man), then the form intense young man was probably a constituent. Harris’s test was
the beginning of the intuition that a constituent is a kind of equivalence class.
The context-free grammar was a formalization of this idea of hierarchical
constituency defined in Chomsky (1956) and further expanded upon (and argued
against) in Chomsky (1957) and Chomsky (1956/1975). Shortly after Chomsky’s
initial work, the context-free grammar was reinvented by Backus (1959) and inde-
pendently by Naur et al. (1960) in their descriptions of the ALGOL programming
language; Backus (1996) noted that he was influenced by the productions of Emil
Post and that Naur’s work was independent of his (Backus’) own. After this early
work, a great number of computational models of natural language processing were
based on context-free grammars because of the early development of efficient pars-
ing algorithms.
Dynamic programming parsing has a history of independent discovery. Ac-
cording to the late Martin Kay (personal communication), a dynamic programming
parser containing the roots of the CKY algorithm was first implemented by John
Cocke in 1960. Later work extended and formalized the algorithm, as well as prov-
ing its time complexity (Kay 1967, Younger 1967, Kasami 1965). The related well-
WFST formed substring table (WFST) seems to have been independently proposed by
Kuno (1965) as a data structure that stores the results of all previous computations
in the course of the parse. Based on a generalization of Cocke’s work, a similar
data structure had been independently described in Kay (1967) (and Kay 1973). The
top-down application of dynamic programming to parsing was described in Earley’s
Ph.D. dissertation (Earley 1968, Earley 1970). Sheil (1976) showed the equivalence
of the WFST and the Earley algorithm. Norvig (1991) shows that the efficiency of-
fered by dynamic programming can be captured in any language with a memoization
function (such as in LISP) simply by wrapping the memoization operation around a
simple top-down parser.
probabilistic
The earliest disambiguation algorithms for parsing were based on probabilistic
context-free context-free grammars, first worked out by Booth (1969) and Salomaa (1969); see
grammars
Appendix C for more history. Neural methods were first applied to parsing at around
the same time as statistical parsing methods were developed (Henderson, 1994). In
the earliest work neural networks were used to estimate some of the probabilities for
statistical constituency parsers (Henderson, 2003, 2004; Emami and Jelinek, 2005)
. The next decades saw a wide variety of neural parsing algorithms, including re-
cursive neural architectures (Socher et al., 2011, 2013), encoder-decoder models
(Vinyals et al., 2015; Choe and Charniak, 2016), and the idea of focusing on spans
(Cross and Huang, 2016). For more on the span-based self-attention approach we
describe in this chapter see Stern et al. (2017), Gaddy et al. (2018), Kitaev and Klein
(2018), and Kitaev et al. (2019). See Chapter 19 for the parallel history of neural
dependency parsing.
The classic reference for parsing algorithms is Aho and Ullman (1972); although
the focus of that book is on computer languages, most of the algorithms have been
applied to natural language.
Exercises
18.1 Implement the algorithm to convert arbitrary context-free grammars to CNF.
426 C HAPTER 18 • C ONTEXT-F REE G RAMMARS AND C ONSTITUENCY PARSING
19 Dependency Parsing
Tout mot qui fait partie d’une phrase... Entre lui et ses voisins, l’esprit aperçoit
des connexions, dont l’ensemble forme la charpente de la phrase.
[Between each word in a sentence and its neighbors, the mind perceives con-
nections. These connections together form the scaffolding of the sentence.]
Lucien Tesnière. 1959. Éléments de syntaxe structurale, A.1.§4
The focus of the last chapter was on context-free grammars and constituent-
based representations. Here we present another important family of grammar for-
dependency
grammars malisms called dependency grammars. In dependency formalisms, phrasal con-
stituents and phrase-structure rules do not play a direct role. Instead, the syntactic
structure of a sentence is described solely in terms of directed binary grammatical
relations between the words, as in the following dependency parse:
root
obj
det nmod
(19.1)
nsubj compound case
Relations among the words are illustrated above the sentence with directed, labeled
typed
dependency arcs from heads to dependents. We call this a typed dependency structure because
the labels are drawn from a fixed inventory of grammatical relations. A root node
explicitly marks the root of the tree, the head of the entire structure.
Figure 19.1 on the next page shows the dependency analysis from Eq. 19.1 but
visualized as a tree, alongside its corresponding phrase-structure analysis of the kind
given in the prior chapter. Note the absence of nodes corresponding to phrasal con-
stituents or lexical categories in the dependency parse; the internal structure of the
dependency parse consists solely of directed relations between words. These head-
dependent relationships directly encode important information that is often buried in
the more complex phrase-structure parses. For example, the arguments to the verb
prefer are directly linked to it in the dependency structure, while their connection
to the main verb is more distant in the phrase-structure tree. Similarly, morning
and Denver, modifiers of flight, are linked to it directly in the dependency structure.
This fact that the head-dependent relations are a good proxy for the semantic rela-
tionship between predicates and their arguments is an important reason why depen-
dency grammars are currently more common than constituency grammars in natural
language processing.
Another major advantage of dependency grammars is their ability to deal with
free word order languages that have a relatively free word order. For example, word order in Czech
can be much more flexible than in English; a grammatical object might occur before
or after a location adverbial. A phrase-structure grammar would need a separate rule
428 C HAPTER 19 • D EPENDENCY PARSING
prefer S
I flight NP VP
Nom Noun P NP
morning Denver
Figure 19.1 Dependency and constituent analyses for I prefer the morning flight through Denver.
for each possible place in the parse tree where such an adverbial phrase could occur.
A dependency-based approach can have just one link type representing this particu-
lar adverbial relation; dependency grammar approaches can thus abstract away a bit
more from word order information.
In the following sections, we’ll give an inventory of relations used in dependency
parsing, discuss two families of parsing algorithms (transition-based, and graph-
based), and discuss evaluation.
Here the clausal relations NSUBJ and OBJ identify the subject and direct object of
the predicate cancel, while the NMOD, DET, and CASE relations denote modifiers of
the nouns flights and Houston.
19.1.2 Projectivity
The notion of projectivity imposes an additional constraint that is derived from the
order of the words in the input. An arc from a head to a dependent is said to be
projective projective if there is a path from the head to every word that lies between the head
and the dependent in the sentence. A dependency tree is then said to be projective if
all the arcs that make it up are projective. All the dependency trees we’ve seen thus
far have been projective. There are, however, many valid constructions which lead
to non-projective trees, particularly in languages with relatively flexible word order.
Consider the following example.
acl:relcl
root obl
obj cop
(19.3)
nsubj det det nsubj adv
JetBlue canceled our flight this morning which was already late
In this example, the arc from flight to its modifier late is non-projective since there
is no path from flight to the intervening words this and morning. As we can see from
this diagram, projectivity (and non-projectivity) can be detected in the way we’ve
been drawing our trees. A dependency tree is projective if it can be drawn with
no crossing edges. Here there is no way to link flight to its dependent late without
crossing the arc that links morning to its head.
19.1 • D EPENDENCY R ELATIONS 431
Our concern with projectivity arises from two related issues. First, the most
widely used English dependency treebanks were automatically derived from phrase-
structure treebanks through the use of head-finding rules. The trees generated in such
a fashion will always be projective, and hence will be incorrect when non-projective
examples like this one are encountered.
Second, there are computational limitations to the most widely used families of
parsing algorithms. The transition-based approaches discussed in Section 19.2 can
only produce projective trees, hence any sentences with non-projective structures
will necessarily contain some errors. This limitation is one of the motivations for
the more flexible graph-based parsing approach described in Section 19.3.
punct
obl:tmod
obl
case case
det det
nsubj punct
obj aux
adv
nsubj
obj:tmod obj
advmod compound:vv
transition-based Our first approach to dependency parsing is called transition-based parsing. This
architecture draws on shift-reduce parsing, a paradigm originally developed for
analyzing programming languages (Aho and Ullman, 1972). In transition-based
parsing we’ll have a stack on which we build the parse, a buffer of tokens to be
parsed, and a parser which takes actions on the parse via a predictor called an oracle,
as illustrated in Fig. 19.4.
Input buffer
w1 w2 wn
s1 Dependency
s2
Parser LEFTARC Relations
Action
Stack ... Oracle RIGHTARC
w3 w2
SHIFT
sn
Figure 19.4 Basic transition-based parser. The parser examines the top two elements of the
stack and selects an action by consulting an oracle that examines the current configuration.
The parser walks through the sentence left-to-right, successively shifting items
from the buffer onto the stack. At each time point we examine the top two elements
on the stack, and the oracle makes a decision about what transition to apply to build
the parse. The possible transitions correspond to the intuitive actions one might take
in creating a dependency tree by examining the words in a single pass over the input
from left to right (Covington, 2001):
• Assign the current word as the head of some previously seen word,
• Assign some previously seen word as the head of the current word,
• Postpone dealing with the current word, storing it for later processing.
We’ll formalize this intuition with the following three transition operators that
will operate on the top two elements of the stack:
• LEFTA RC: Assert a head-dependent relation between the word at the top of
the stack and the second word; remove the second word from the stack.
• RIGHTA RC: Assert a head-dependent relation between the second word on
the stack and the word at the top; remove the top word from the stack;
19.2 • T RANSITION -BASED D EPENDENCY PARSING 433
• SHIFT: Remove the word from the front of the input buffer and push it onto
the stack.
We’ll sometimes call operations like LEFTA RC and RIGHTA RC reduce operations,
based on a metaphor from shift-reduce parsing, in which reducing means combin-
ing elements on the stack. There are some preconditions for using operators. The
LEFTA RC operator cannot be applied when ROOT is the second element of the stack
(since by definition the ROOT node cannot have any incoming arcs). And both the
LEFTA RC and RIGHTA RC operators require two elements to be on the stack to be
applied.
arc standard This particular set of operators implements what is known as the arc standard
approach to transition-based parsing (Covington 2001, Nivre 2003). In arc standard
parsing the transition operators only assert relations between elements at the top of
the stack, and once an element has been assigned its head it is removed from the
stack and is not available for further processing. As we’ll see, there are alterna-
tive transition systems which demonstrate different parsing behaviors, but the arc
standard approach is quite effective and is simple to implement.
The specification of a transition-based parser is quite simple, based on repre-
configuration senting the current state of the parse as a configuration: the stack, an input buffer
of words or tokens, and a set of relations representing a dependency tree. Parsing
means making a sequence of transitions through the space of possible configura-
tions. We start with an initial configuration in which the stack contains the ROOT
node, the buffer has the tokens in the sentence, and an empty set of relations repre-
sents the parse. In the final goal state, the stack and the word list should be empty,
and the set of relations will represent the final parse. Fig. 19.5 gives the algorithm.
At each step, the parser consults an oracle (we’ll come back to this shortly) that
provides the correct transition operator to use given the current configuration. It then
applies that operator to the current configuration, producing a new configuration.
The process ends when all the words in the sentence have been consumed and the
ROOT node is the only element remaining on the stack.
The efficiency of transition-based parsers should be apparent from the algorithm.
The complexity is linear in the length of the sentence since it is based on a single
left to right pass through the words in the sentence. (Each word must first be shifted
onto the stack and then later reduced.)
Note that unlike the dynamic programming and search-based approaches dis-
cussed in Chapter 18, this approach is a straightforward greedy algorithm—the or-
acle provides a single choice at each step and the parser proceeds with that choice,
no other options are explored, no backtracking is employed, and a single parse is
returned in the end.
Figure 19.6 illustrates the operation of the parser with the sequence of transitions
434 C HAPTER 19 • D EPENDENCY PARSING
root
obj
det
(19.7)
iobj compound
Let’s consider the state of the configuration at Step 2, after the word me has been
pushed onto the stack.
The correct operator to apply here is RIGHTA RC which assigns book as the head of
me and pops me from the stack resulting in the following configuration.
Here, all the remaining words have been passed onto the stack and all that is left
to do is to apply the appropriate reduce operators. In the current configuration, we
employ the LEFTA RC operator resulting in the following state.
At this point, the parse for this sentence consists of the following structure.
iobj compound
(19.8)
Book me the morning flight
There are several important things to note when examining sequences such as
the one in Figure 19.6. First, the sequence given is not the only one that might lead
to a reasonable parse. In general, there may be more than one path that leads to the
same result, and due to ambiguity, there may be other transition sequences that lead
to different equally valid parses.
Second, we are assuming that the oracle always provides the correct operator
at each point in the parse—an assumption that is unlikely to be true in practice.
As a result, given the greedy nature of this algorithm, incorrect choices will lead to
incorrect parses since the parser has no opportunity to go back and pursue alternative
choices. Section 19.2.4 will introduce several techniques that allow transition-based
approaches to explore the search space more fully.
19.2 • T RANSITION -BASED D EPENDENCY PARSING 435
Finally, for simplicity, we have illustrated this example without the labels on
the dependency relations. To produce labeled trees, we can parameterize the LEFT-
A RC and RIGHTA RC operators with dependency labels, as in LEFTA RC ( NSUBJ ) or
RIGHTA RC ( OBJ ). This is equivalent to expanding the set of transition operators from
our original set of three to a set that includes LEFTA RC and RIGHTA RC operators for
each relation in the set of dependency relations being used, plus an additional one
for the SHIFT operator. This, of course, makes the job of the oracle more difficult
since it now has a much larger set of operators from which to choose.
Let’s walk through the processing of the following example as shown in Fig. 19.7.
root
obj nmod
(19.9)
det case
possible action. The same conditions hold in the next two steps. In step 3, LEFTA RC
is selected to link the to its head.
Now consider the situation in Step 4.
Here, we might be tempted to add a dependency relation between book and flight,
which is present in the reference parse. But doing so now would prevent the later
attachment of Houston since flight would have been removed from the stack. For-
tunately, the precondition on choosing RIGHTA RC prevents this choice and we’re
again left with SHIFT as the only viable option. The remaining choices complete the
set of operators needed for this example.
To recap, we derive appropriate training instances consisting of configuration-
transition pairs from a treebank by simulating the operation of a parser in the con-
text of a reference dependency tree. We can deterministically record correct parser
actions at each step as we progress through each training example, thereby creating
the training set we require.
hs1 .w, opi, hs2 .w, opihs1 .t, opi, hs2 .t, opi
hb1 .w, opi, hb1 .t, opihs1 .wt, opi (19.10)
The correct transition here is SHIFT (you should convince yourself of this before
proceeding). The application of our set of feature templates to this configuration
would result in the following set of instantiated features.
Given that the left and right arc transitions operate on the top two elements of the
stack, features that combine properties from these positions are even more useful.
For example, a feature like s1 .t ◦ s2 .t concatenates the part of speech tag of the word
at the top of the stack with the tag of the word beneath it.
Given the training data and features, any classifier, like multinomial logistic re-
gression or support vector machines, can be used.
Input buffer
Parser Oracle
w …
w e(w) Dependency
Action
Relations
Softmax
...
ENCODER
w1 w2 w3 w4 w5 w6
Figure 19.8 Neural classifier for the oracle for the transition-based parser. The parser takes
the top 2 words on the stack and the first word of the buffer, represents them by their encodings
(from running the whole sentence through the encoder), concatenates the embeddings and
passes through a softmax to choose a parser action (transition).
19.2 • T RANSITION -BASED D EPENDENCY PARSING 439
obj nmod
det case
underlying parsing algorithm. This flexibility has led to the development of a di-
verse set of transition systems that address different aspects of syntax and semantics
including: assigning part of speech tags (Choi and Palmer, 2011a), allowing the
generation of non-projective dependency structures (Nivre, 2009), assigning seman-
tic roles (Choi and Palmer, 2011b), and parsing texts containing multiple languages
(Bhat et al., 2017).
Beam Search
The computational efficiency of the transition-based approach discussed earlier de-
rives from the fact that it makes a single pass through the sentence, greedily making
decisions without considering alternatives. Of course, this is also a weakness – once
a decision has been made it can not be undone, even in the face of overwhelming
beam search evidence arriving later in a sentence. We can use beam search to explore alterna-
tive decision sequences. Recall from Chapter 8 that beam search uses a breadth-first
search strategy with a heuristic filter that prunes the search frontier to stay within a
beam width fixed-size beam width.
In applying beam search to transition-based parsing, we’ll elaborate on the al-
gorithm given in Fig. 19.5. Instead of choosing the single best transition operator
at each iteration, we’ll apply all applicable operators to each state on an agenda and
then score the resulting configurations. We then add each of these new configura-
tions to the frontier, subject to the constraint that there has to be room within the
beam. As long as the size of the agenda is within the specified beam width, we can
add new configurations to the agenda. Once the agenda reaches the limit, we only
add new configurations that are better than the worst configuration on the agenda
(removing the worst element so that we stay within the limit). Finally, to insure that
we retrieve the best possible state on the agenda, the while loop continues as long as
there are non-final states on the agenda.
The beam search approach requires a more elaborate notion of scoring than we
used with the greedy algorithm. There, we assumed that the oracle would be a
supervised classifier that chose the best transition operator based on features of the
current configuration. This choice can be viewed as assigning a score to all the
possible transitions and picking the best one.
With beam search we are now searching through the space of decision sequences,
so it makes sense to base the score for a configuration on its entire history. So we
can define the score for a new configuration as the score of its predecessor plus the
19.3 • G RAPH -BASED D EPENDENCY PARSING 441
edge-factored We’ll make the simplifying assumption that this score can be edge-factored,
meaning that the overall score for a tree is the sum of the scores of each of the scores
of the edges that comprise the tree.
X
Score(t, S) = Score(e)
e∈t
4
4
12
5 8
root Book that flight
6 7
Figure 19.11 Initial rooted, directed graph for Book that flight.
Before describing the algorithm it’s useful to consider two intuitions about di-
rected graphs and their spanning trees. The first intuition begins with the fact that
every vertex in a spanning tree has exactly one incoming edge. It follows from this
that every connected component of a spanning tree (i.e., every set of vertices that
are linked to each other by paths over edges) will also have one incoming edge.
The second intuition is that the absolute values of the edge scores are not critical
to determining its maximum spanning tree. Instead, it is the relative weights of the
edges entering each vertex that matters. If we were to subtract a constant amount
from each edge entering a given vertex it would have no impact on the choice of
19.3 • G RAPH -BASED D EPENDENCY PARSING 443
the maximum spanning tree since every possible spanning tree would decrease by
exactly the same amount.
The first step of the algorithm itself is quite straightforward. For each vertex
in the graph, an incoming edge (representing a possible head assignment) with the
highest score is chosen. If the resulting set of edges produces a spanning tree then
we’re done. More formally, given the original fully-connected graph G = (V, E), a
subgraph T = (V, F) is a spanning tree if it has no cycles and each vertex (other than
the root) has exactly one edge entering it. If the greedy selection process produces
such a tree then it is the best possible one.
Unfortunately, this approach doesn’t always lead to a tree since the set of edges
selected may contain cycles. Fortunately, in yet another case of multiple discovery,
there is a straightforward way to eliminate cycles generated during the greedy se-
lection phase. Chu and Liu (1965) and Edmonds (1967) independently developed
an approach that begins with greedy selection and follows with an elegant recursive
cleanup phase that eliminates cycles.
The cleanup phase begins by adjusting all the weights in the graph by subtracting
the score of the maximum edge entering each vertex from the score of all the edges
entering that vertex. This is where the intuitions mentioned earlier come into play.
We have scaled the values of the edges so that the weights of the edges in the cycle
have no bearing on the weight of any of the possible spanning trees. Subtracting the
value of the edge with maximum weight from each edge entering a vertex results
in a weight of zero for all of the edges selected during the greedy selection phase,
including all of the edges involved in the cycle.
Having adjusted the weights, the algorithm creates a new graph by selecting a
cycle and collapsing it into a single new node. Edges that enter or leave the cycle
are altered so that they now enter or leave the newly collapsed node. Edges that do
not touch the cycle are included and edges within the cycle are dropped.
Now, if we knew the maximum spanning tree of this new graph, we would have
what we need to eliminate the cycle. The edge of the maximum spanning tree di-
rected towards the vertex representing the collapsed cycle tells us which edge to
delete in order to eliminate the cycle. How do we find the maximum spanning tree
of this new graph? We recursively apply the algorithm to the new graph. This will
either result in a spanning tree or a graph with a cycle. The recursions can continue
as long as cycles are encountered. When each recursion completes we expand the
collapsed vertex, restoring all the vertices and edges from the cycle with the excep-
tion of the single edge to be deleted.
Putting all this together, the maximum spanning tree algorithm consists of greedy
edge selection, re-scoring of edge costs and a recursive cleanup phase when needed.
The full algorithm is shown in Fig. 19.12.
Fig. 19.13 steps through the algorithm with our Book that flight example. The
first row of the figure illustrates greedy edge selection with the edges chosen shown
in blue (corresponding to the set F in the algorithm). This results in a cycle between
that and flight. The scaled weights using the maximum value entering each node are
shown in the graph to the right.
Collapsing the cycle between that and flight to a single node (labelled tf) and
recursing with the newly scaled costs is shown in the second row. The greedy selec-
tion step in this recursion yields a spanning tree that links root to book, as well as an
edge that links book to the contracted node. Expanding the contracted node, we can
see that this edge corresponds to the edge from book to flight in the original graph.
This in turn tells us which edge to drop to eliminate the cycle.
444 C HAPTER 19 • D EPENDENCY PARSING
F ← []
T’ ← []
score’ ← []
for each v ∈ V do
bestInEdge ← argmaxe=(u,v)∈ E score[e]
F ← F ∪ bestInEdge
for each e=(u,v) ∈ E do
score’[e] ← score[e] − score[bestInEdge]
Figure 19.12 The Chu-Liu Edmonds algorithm for finding a maximum spanning tree in a
weighted directed graph.
On arbitrary directed graphs, this version of the CLE algorithm runs in O(mn)
time, where m is the number of edges and n is the number of nodes. Since this par-
ticular application of the algorithm begins by constructing a fully connected graph
m = n2 yielding a running time of O(n3 ). Gabow et al. (1986) present a more effi-
cient implementation with a running time of O(m + nlogn).
Or more succinctly.
score(S, e) = w · f
Given this formulation, we need to identify relevant features and train the weights.
The features (and feature combinations) used to train edge-factored models mir-
ror those used in training transition-based parsers, such as
19.3 • G RAPH -BASED D EPENDENCY PARSING 445
-4
4
4 -3
12 0
5 8 -2 0
Book that flight Book that flight
root root
12 7 8 12 -6 7 8
6 7 0
7 -1
5 -7
-4 -4
-3 -3
0 0
-2 -2
Book tf
root Book -6 tf root -6
0 -1
-1 -1
-7 -7
• Wordforms, lemmas, and parts of speech of the headword and its dependent.
• Corresponding features from the contexts before, after and between the words.
• Word embeddings.
• The dependency relation itself.
• The direction of the relation (to the right or left).
• The distance from the head to the dependent.
Given a set of features, our next problem is to learn a set of weights correspond-
ing to each. Unlike many of the learning problems discussed in earlier chapters,
here we are not training a model to associate training items with class labels, or
parser actions. Instead, we seek to train a model that assigns higher scores to cor-
rect trees than to incorrect ones. An effective framework for problems like this is to
inference-based
learning use inference-based learning combined with the perceptron learning rule. In this
framework, we parse a sentence (i.e, perform inference) from the training set using
some initially random set of initial weights. If the resulting parse matches the cor-
responding tree in the training data, we do nothing to the weights. Otherwise, we
find those features in the incorrect parse that are not present in the reference parse
and we lower their weights by a small amount based on the learning rate. We do this
incrementally for each sentence in our training data until the weights converge.
446 C HAPTER 19 • D EPENDENCY PARSING
score(h1head, h3dep)
∑
Biaffine
U W b
+
h1 head h1 dep h2 head h2 dep h3 head h3 dep
r1 r2 r3
ENCODER
Here we’ll sketch the biaffine algorithm of Dozat and Manning (2017) and Dozat
et al. (2017) shown in Fig. 19.14, drawing on the work of Grünewald et al. (2021)
who tested many versions of the algorithm via their STEPS system. The algorithm
first runs the sentence X = x1 , ..., xn through an encoder to produce a contextual
embedding representation for each token R = r1 , ..., rn . The embedding for each
token is now passed through two separate feedforward networks, one to produce a
representation of this token as a head, and one to produce a representation of this
token as a dependent:
hhead
i = FFNhead (ri ) (19.13)
hdep
i = FFN dep
(ri ) (19.14)
Now to assign a score to the directed edge i → j, (wi is the head and w j is the depen-
dent), we feed the head representation of i, hhead
i , and the dependent representation
of j, hdep
j , into a biaffine scoring function:
Score(i → j) = Biaff(hhead
i , hdep
j ) (19.15)
|
Biaff(x, y) = x Uy + W(x ⊕ y) + b (19.16)
19.4 • E VALUATION 447
where U, W, and b are weights learned by the model. The idea of using a biaffine
function is to allow the system to learn multiplicative interactions between the vec-
tors x and y.
If we pass Score(i → j) through a softmax, we end up with a probability distri-
bution, for each token j, over potential heads i (all other tokens in the sentence):
This probability can then be passed to the maximum spanning tree algorithm of
Section 19.3.1 to find the best tree.
This p(i → j) classifier is trained by optimizing the cross-entropy loss.
Note that the algorithm as we’ve described it is unlabeled. To make this into
a labeled algorithm, the Dozat and Manning (2017) algorithm actually trains two
classifiers. The first classifier, the edge-scorer, the one we described above, assigns
a probability p(i → j) to each word wi and w j . Then the Maximum Spanning Tree
algorithm is run to get a single best dependency parse tree for the second. We then
apply a second classifier, the label-scorer, whose job is to find the maximum prob-
ability label for each edge in this parse. This second classifier has the same form
as (19.15-19.17), but instead of being trained to predict with binary softmax the
probability of an edge existing between two words, it is trained with a softmax over
dependency labels to predict the dependency label between the words.
19.4 Evaluation
As with phrase structure-based parsing, the evaluation of dependency parsers pro-
ceeds by measuring how well they work on a test set. An obvious metric would be
exact match (EM)—how many sentences are parsed correctly. This metric is quite
pessimistic, with most sentences being marked wrong. Such measures are not fine-
grained enough to guide the development process. Our metrics need to be sensitive
enough to tell if actual improvements are being made.
For these reasons, the most common method for evaluating dependency parsers
are labeled and unlabeled attachment accuracy. Labeled attachment refers to the
proper assignment of a word to its head along with the correct dependency relation.
Unlabeled attachment simply looks at the correctness of the assigned head, ignor-
ing the dependency relation. Given a system output and a corresponding reference
parse, accuracy is simply the percentage of words in an input that are assigned the
correct head with the correct relation. These metrics are usually referred to as the
labeled attachment score (LAS) and unlabeled attachment score (UAS). Finally, we
can make use of a label accuracy score (LS), the percentage of tokens with correct
labels, ignoring where the relations are coming from.
As an example, consider the reference parse and system parse for the following
example shown in Fig. 19.15.
(19.18) Book me the flight through Houston.
The system correctly finds 4 of the 6 dependency relations present in the reference
parse and receives an LAS of 2/3. However, one of the 2 incorrect relations found
by the system holds between book and flight, which are in a head-dependent relation
in the reference parse; the system therefore achieves a UAS of 5/6.
Beyond attachment scores, we may also be interested in how well a system is
performing on a particular kind of dependency relation, for example NSUBJ, across
448 C HAPTER 19 • D EPENDENCY PARSING
root root
obj xcomp
nmod nsubj nmod
iobj det case det case
Book me the flight through Houston Book me the flight through Houston
(a) Reference (b) System
Figure 19.15 Reference and system parses for Book me the flight through Houston, resulting in an LAS of
2/3 and an UAS of 5/6.
a development corpus. Here we can make use of the notions of precision and recall
introduced in Chapter 17, measuring the percentage of relations labeled NSUBJ by
the system that were correct (precision), and the percentage of the NSUBJ relations
present in the development set that were in fact discovered by the system (recall).
We can employ a confusion matrix to keep track of how often each dependency type
was confused for another.
19.5 Summary
This chapter has introduced the concept of dependency grammars and dependency
parsing. Here’s a summary of the main points that we covered:
Historical Notes
The dependency-based approach to grammar is much older than the relatively recent
phrase-structure or constituency grammars, which date only to the 20th century. De-
pendency grammar dates back to the Indian grammarian Pā[Link] sometime between
the 7th and 4th centuries BCE, as well as the ancient Greek linguistic traditions.
Contemporary theories of dependency grammar all draw heavily on the 20th cen-
tury work of Tesnière (1959).
Automatic parsing using dependency grammars was first introduced into compu-
tational linguistics by early work on machine translation at the RAND Corporation
led by David Hays. This work on dependency parsing closely paralleled work on
constituent parsing and made explicit use of grammars to guide the parsing process.
After this early period, computational work on dependency parsing remained inter-
mittent over the following decades. Notable implementations of dependency parsers
for English during this period include Link Grammar (Sleator and Temperley, 1993),
Constraint Grammar (Karlsson et al., 1995), and MINIPAR (Lin, 2003).
Dependency parsing saw a major resurgence in the late 1990’s with the appear-
ance of large dependency-based treebanks and the associated advent of data driven
approaches described in this chapter. Eisner (1996) developed an efficient dynamic
programming approach to dependency parsing based on bilexical grammars derived
from the Penn Treebank. Covington (2001) introduced the deterministic word by
word approach underlying current transition-based approaches. Yamada and Mat-
sumoto (2003) and Kudo and Matsumoto (2002) introduced both the shift-reduce
paradigm and the use of supervised machine learning in the form of support vector
machines to dependency parsing.
Transition-based parsing is based on the shift-reduce parsing algorithm orig-
inally developed for analyzing programming languages (Aho and Ullman, 1972).
Shift-reduce parsing also makes use of a context-free grammar. Input tokens are
successively shifted onto the stack and the top two elements of the stack are matched
against the right-hand side of the rules in the grammar; when a match is found the
matched elements are replaced on the stack (reduced) by the non-terminal from the
left-hand side of the rule being matched. In transition-based dependency parsing
we skip the grammar, and alter the reduce operation to add a dependency relation
between a word and its head.
Nivre (2003) defined the modern, deterministic, transition-based approach to
dependency parsing. Subsequent work by Nivre and his colleagues formalized and
analyzed the performance of numerous transition systems, training methods, and
methods for dealing with non-projective language (Nivre and Scholz 2004, Nivre
2006, Nivre and Nilsson 2005, Nivre et al. 2007b, Nivre 2007). The neural ap-
proach was pioneered by Chen and Manning (2014) and extended by Kiperwasser
and Goldberg (2016); Kulmizev et al. (2019).
The graph-based maximum spanning tree approach to dependency parsing was
introduced by McDonald et al. 2005a, McDonald et al. 2005b. The neural classifier
was introduced by (Kiperwasser and Goldberg, 2016).
The long-running Prague Dependency Treebank project (Hajič, 1998) is the most
significant effort to directly annotate a corpus with multiple layers of morphological,
syntactic and semantic information. PDT 3.0 contains over 1.5 M tokens (Bejček
et al., 2013).
Universal Dependencies (UD) (de Marneffe et al., 2021) is an open community
450 C HAPTER 19 • D EPENDENCY PARSING
project to create a framework for dependency treebank annotation, with nearly 200
treebanks in over 100 languages. The UD annotation scheme evolved out of several
distinct efforts including Stanford dependencies (de Marneffe et al. 2006, de Marn-
effe and Manning 2008, de Marneffe et al. 2014), Google’s universal part-of-speech
tags (Petrov et al., 2012), and the Interset interlingua for morphosyntactic tagsets
(Zeman, 2008).
The Conference on Natural Language Learning (CoNLL) has conducted an in-
fluential series of shared tasks related to dependency parsing over the years (Buch-
holz and Marsi 2006, Nivre et al. 2007a, Surdeanu et al. 2008, Hajič et al. 2009).
More recent evaluations have focused on parser robustness with respect to morpho-
logically rich languages (Seddah et al., 2013), and non-canonical language forms
such as social media, texts, and spoken language (Petrov and McDonald, 2012).
Choi et al. (2015) presents a performance analysis of 10 dependency parsers across
a range of metrics, as well as DEPENDA BLE, a robust parser evaluation tool.
Exercises
CHAPTER
20 Information Extraction:
Relations, Events, and Time
Time will explain.
Jane Austen, Persuasion
Imagine that you are an analyst with an investment firm that tracks airline stocks.
You’re given the task of determining the relationship (if any) between airline an-
nouncements of fare increases and the behavior of their stocks the next day. His-
torical data about stock prices is easy to come by, but what about the airline an-
nouncements? You will need to know at least the name of the airline, the nature of
the proposed fare hike, the dates of the announcement, and possibly the response of
other airlines. Fortunately, these can be all found in news articles like this one:
Citing high fuel prices, United Airlines said Friday it has increased fares
by $6 per round trip on flights to some cities also served by lower-
cost carriers. American Airlines, a unit of AMR Corp., immediately
matched the move, spokesman Tim Wagner said. United, a unit of UAL
Corp., said the increase took effect Thursday and applies to most routes
where it competes against discount carriers, such as Chicago to Dallas
and Denver to San Francisco.
This chapter presents techniques for extracting limited kinds of semantic con-
information tent from text. This process of information extraction (IE) turns the unstructured
extraction
information embedded in texts into structured data, for example for populating a
relational database to enable further processing.
relation We begin with the task of relation extraction: finding and classifying semantic
extraction
relations among entities mentioned in a text, like child-of (X is the child-of Y), or
part-whole or geospatial relations. Relation extraction has close links to populat-
knowledge
graphs ing a relational database, and knowledge graphs, datasets of structured relational
knowledge, are a useful way for search engines to present information to users.
event Next, we discuss event extraction, the task of finding events in which these en-
extraction
tities participate, like, in our sample text, the fare increases by United and American
and the reporting events said and cite. Events are also situated in time, occurring at
a particular date or time, and events can be related temporally, happening before or
after or simultaneously with each other. We’ll need to recognize temporal expres-
sions like Friday, Thursday or two days from now and times such as 3:30 P.M., and
normalize them onto specific calendar dates or times. We’ll need to link Friday to
the time of United’s announcement, Thursday to the previous day’s fare increase,
and we’ll need to produce a timeline in which United’s announcement follows the
fare increase and American’s announcement follows both of those events.
template filling The related task of template filling is to find recurring stereotypical events or
situations in documents and fill in the template slots. These slot-fillers may consist
of text segments extracted directly from the text, or concepts like times, amounts, or
ontology entities that have been inferred through additional processing. Our airline
452 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
Lasting Subsidiary
Family Near Citizen-
Personal Resident- Geographical
Located Ethnicity- Org-Location-
Business
Religion Origin
ORG
AFFILIATION ARTIFACT
Investor
Founder
Student-Alum
Ownership User-Owner-Inventor-
Employment Manufacturer
Membership
Sports-Affiliation
Figure 20.1 The 17 relations used in the ACE relation extraction task.
text presents such a stereotypical situation since airlines often raise fares and then
wait to see if competitors follow along. Here we can identify United as a lead air-
line that initially raised its fares, $6 as the amount, Thursday as the increase date,
and American as an airline that followed along, leading to a filled template like the
following:
FARE -R AISE ATTEMPT: L EAD A IRLINE : U NITED A IRLINES
A MOUNT: $6
E FFECTIVE DATE : 2006-10-26
F OLLOWER : A MERICAN A IRLINES
Sets of relations have been defined for many other domains as well. For example
UMLS, the Unified Medical Language System from the US National Library of
Medicine has a network that defines 134 broad subject categories, entity types, and
54 relations between the entities, such as the following:
Entity Relation Entity
Injury disrupts Physiological Function
Bodily Location location-of Biologic Function
Anatomical Structure part-of Organism
Pharmacologic Substance causes Pathological Function
Pharmacologic Substance treats Pathologic Function
Given a medical sentence like this one:
(20.1) Doppler echocardiography can be used to diagnose left anterior descending
artery stenosis in patients with type 2 diabetes
We could thus extract the UMLS relation:
Echocardiography, Doppler Diagnoses Acquired stenosis
infoboxes Wikipedia also offers a large supply of relations, drawn from infoboxes, struc-
tured tables associated with certain Wikipedia articles. For example, the Wikipedia
infobox for Stanford includes structured facts like state = "California" or
president = "Marc Tessier-Lavigne". These facts can be turned into rela-
RDF tions like president-of or located-in. or into relations in a metalanguage called RDF
RDF triple (Resource Description Framework). An RDF triple is a tuple of entity-relation-
entity, called a subject-predicate-object expression. Here’s a sample RDF triple:
subject predicate object
Golden Gate Park location San Francisco
For example the crowdsourced DBpedia (Bizer et al., 2009) is an ontology de-
rived from Wikipedia containing over 2 billion RDF triples. Another dataset from
Freebase Wikipedia infoboxes, Freebase (Bollacker et al., 2008), now part of Wikidata (Vrandečić
and Krötzsch, 2014), has relations between people and their nationality, or locations,
and other locations they are contained in.
WordNet or other ontologies offer useful ontological relations that express hier-
is-a archical relations between words or concepts. For example WordNet has the is-a or
hypernym hypernym relation between classes,
Giraffe is-a ruminant is-a ungulate is-a mammal is-a vertebrate ...
WordNet also has Instance-of relation between individuals and classes, so that for
example San Francisco is in the Instance-of relation with city. Extracting these
relations is an important step in extending or building ontologies.
Finally, there are large datasets that contain sentences hand-labeled with their
relations, designed for training and testing relation extractors. The TACRED dataset
(Zhang et al., 2017) contains 106,264 examples of relation triples about particular
people or organizations, labeled in sentences from news and web text drawn from the
454 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
annual TAC Knowledge Base Population (TAC KBP) challenges. TACRED contains
41 relation types (like per:city of birth, org:subsidiaries, org:member of, per:spouse),
plus a no relation tag; examples are shown in Fig. 20.3. About 80% of all examples
are annotated as no relation; having sufficient negative data is important for training
supervised classifiers.
A standard dataset was also produced for the SemEval 2010 Task 8, detecting
relations between nominals (Hendrickx et al., 2009). The dataset has 10,717 exam-
ples, each with a pair of nominals (untyped) hand-labeled with one of 9 directed
relations like product-producer ( a factory manufactures suits) or component-whole
(my apartment has a large kitchen).
allowing us to infer
hyponym(Gelidium, red algae) (20.4)
20.2 • R ELATION E XTRACTION A LGORITHMS 455
NP {, NP}* {,} (and|or) other NPH temples, treasuries, and other important civic buildings
NPH such as {NP,}* {(or|and)} NP red algae such as Gelidium
such NPH as {NP,}* {(or|and)} NP such authors as Herrick, Goldsmith, and Shakespeare
NPH {,} including {NP,}* {(or|and)} NP common-law countries, including Canada and England
NPH {,} especially {NP}* {(or|and)} NP European countries, especially France, England, and Spain
Figure 20.4 Hand-built lexico-syntactic patterns for finding hypernyms, using {} to mark optionality (Hearst
1992a, Hearst 1998).
Figure 20.4 shows five patterns Hearst (1992a, 1998) suggested for inferring
the hyponym relation; we’ve shown NPH as the parent/hyponym. Modern versions
of the pattern-based approach extend it by adding named entity constraints. For
example if our goal is to answer questions about “Who holds what office in which
organization?”, we can use patterns like the following:
PER, POSITION of ORG:
George Marshall, Secretary of State of the United States
relations ← nil
entities ← F IND E NTITIES(words)
forall entity pairs he1, e2i in entities do
if R ELATED ?(e1, e2)
relations ← relations+C LASSIFY R ELATION(e1, e2)
Figure 20.5 Finding and classifying the relations among entities in a text.
p(relation|SUBJ,OBJ)
Linear
Classifier
ENCODER
[CLS] [SUBJ_PERSON] was born in [OBJ_LOC] , Michigan
Figure 20.6 Relation extraction as a linear layer on top of an encoder (in this case BERT),
with the subject and object entities replaced in the input by their NER tags (Zhang et al. 2017,
Joshi et al. 2020).
curacies. But labeling a large training set is extremely expensive and supervised
models are brittle: they don’t generalize well to different text genres. For this rea-
son, much research in relation extraction has focused on the semi-supervised and
unsupervised approaches we turn to next.
Suppose, for example, that we need to create a list of airline/hub pairs, and we
know only that Ryanair has a hub at Charleroi. We can use this seed fact to discover
new patterns by finding other mentions of this relation in our corpus. We search
for the terms Ryanair, Charleroi and hub in some proximity. Perhaps we find the
following set of sentences:
(20.6) Budget airline Ryanair, which uses Charleroi as a hub, scrapped all
weekend flights out of the airport.
(20.7) All flights in and out of Ryanair’s hub at Charleroi airport were grounded on
Friday...
(20.8) A spokesman at Charleroi, a main hub for Ryanair, estimated that 8000
passengers had already been affected.
458 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
From these results, we can use the context of words between the entity mentions,
the words before mention one, the word after mention two, and the named entity
types of the two mentions, and perhaps other features, to extract general patterns
such as the following:
/ [ORG], which uses [LOC] as a hub /
/ [ORG]’s hub at [LOC] /
/ [LOC], a main hub for [ORG] /
These new patterns can then be used to search for additional tuples.
confidence Bootstrapping systems also assign confidence values to new tuples to avoid se-
values
semantic drift mantic drift. In semantic drift, an erroneous pattern leads to the introduction of
erroneous tuples, which, in turn, lead to the creation of problematic patterns and the
meaning of the extracted relations ‘drifts’. Consider the following example:
(20.9) Sydney has a ferry hub at Circular Quay.
If accepted as a positive example, this expression could lead to the incorrect in-
troduction of the tuple hSydney,CircularQuayi. Patterns based on this tuple could
propagate further errors into the database.
Confidence values for patterns are based on balancing two factors: the pattern’s
performance with respect to the current set of tuples and the pattern’s productivity
in terms of the number of matches it produces in the document collection. More
formally, given a document collection D, a current set of tuples T , and a proposed
pattern p, we need to track two factors:
• hits(p): the set of tuples in T that p matches while looking in D
• finds(p): The total set of tuples that p finds in D
The following equation balances these considerations (Riloff and Jones, 1999).
|hits(p)|
Conf RlogF (p) = log(|finds(p)|) (20.10)
|finds(p)|
relations, the classifier will also need to be able to label an example as no-relation.
This label is trained by randomly selecting entity pairs that do not appear in any
Freebase relation, extracting features for them, and building a feature vector for
each such tuple. The final algorithm is sketched in Fig. 20.8.
foreach relation R
foreach tuple (e1,e2) of entities with relation R in D
sentences ← Sentences in T that contain e1 and e2
f ← Frequent features in sentences
observations ← observations + new training tuple (e1, e2, f, R)
C ← Train supervised classifier on observations
return C
Figure 20.8 The distant supervision algorithm for relation extraction. A neural classifier
would skip the feature set f .
Distant supervision shares advantages with each of the methods we’ve exam-
ined. Like supervised classification, distant supervision uses a classifier with lots
of features, and supervised by detailed hand-created knowledge. Like pattern-based
classifiers, it can make use of high-precision evidence for the relation between en-
tities. Indeed, distance supervision systems learn patterns just like the hand-built
patterns of early relation extractors. For example the is-a or hypernym extraction
system of Snow et al. (2005) used hypernym/hyponym NP pairs from WordNet as
distant supervision, and then learned new patterns from large amounts of text. Their
system induced exactly the original 5 template patterns of Hearst (1992a), but also
70,000 additional patterns including these four:
NPH like NP Many hormones like leptin...
NPH called NP ...using a markup language called XHTML
NP is a NPH Ruby is a programming language...
NP, a NPH IBM, a company with a long...
This ability to use a large number of features simultaneously means that, un-
like the iterative expansion of patterns in seed-based systems, there’s no semantic
drift. Like unsupervised classification, it doesn’t use a labeled training corpus of
texts, so it isn’t sensitive to genre issues in the training corpus, and relies on very
large amounts of unlabeled data. Distant supervision also has the advantage that it
can create training tuples to be used with neural classifiers, where features are not
required.
The main problem with distant supervision is that it tends to produce low-precision
results, and so current research focuses on ways to improve precision. Furthermore,
distant supervision can only help in extracting relations for which a large enough
database already exists. To extract new relations without datasets, or relations for
new domains, purely unsupervised methods must be used.
has the relation phrases has a hub in and is the headquarters of (it also has has and
is, but longer phrases are preferred). Step 3 finds United to the left and Chicago to
the right of has a hub in, and skips over which to find Chicago to the left of is the
headquarters of. The final output is:
r1: <United, has a hub in, Chicago>
r2: <Chicago, is the headquarters of, United Continental Holdings>
The great advantage of unsupervised relation extraction is its ability to handle
a huge number of relations without having to specify them in advance. The dis-
advantage is the need to map all the strings into some canonical form for adding
to databases or knowledge graphs. Current methods focus heavily on relations ex-
pressed with verbs, and so will miss many relations that are expressed nominally.
[EVENT Citing] high fuel prices, United Airlines [EVENT said] Fri-
day it has [EVENT increased] fares by $6 per round trip on flights to
some cities also served by lower-cost carriers. American Airlines, a unit
of AMR Corp., immediately [EVENT matched] [EVENT the move],
spokesman Tim Wagner [EVENT said]. United, a unit of UAL Corp.,
[EVENT said] [EVENT the increase] took effect Thursday and [EVENT
applies] to most routes where it [EVENT competes] against discount
carriers, such as Chicago to Dallas and Denver to San Francisco.
In English, most event mentions correspond to verbs, and most verbs introduce
events. However, as we can see from our example, this is not always the case. Events
can be introduced by noun phrases, as in the move and the increase, and some verbs
fail to introduce events, as in the phrasal verb took effect, which refers to when the
light verbs event began rather than to the event itself. Similarly, light verbs such as make, take,
and have often fail to denote events. A light verb is a verb that has very little meaning
itself, and the associated event is instead expressed by its direct object noun. In light
verb examples like took a flight, it’s the word flight that defines the event; these light
verbs just provide a syntactic structure for the noun’s arguments.
Various versions of the event extraction task exist, depending on the goal. For
example in the TempEval shared tasks (Verhagen et al. 2009) the goal is to extract
events and aspects like their aspectual and temporal properties. Events are to be
reporting classified as actions, states, reporting events (say, report, tell, explain), perception
events
events, and so on. The aspect, tense, and modality of each event also needs to be
extracted. Thus for example the various said events in the sample text would be
annotated as (class=REPORTING, tense=PAST, aspect=PERFECTIVE).
Event extraction is generally modeled via supervised learning, detecting events
via IOB sequence models and assigning event classes and attributes with multi-class
classifiers. The input can be neural models starting from encoders; or classic feature-
based models using features like those in Fig. 20.10.
Feature Explanation
Character affixes Character-level prefixes and suffixes of target word
Nominalization suffix Character-level suffixes for nominalizations (e.g., -tion)
Part of speech Part of speech of the target word
Light verb Binary feature indicating that the target is governed by a light verb
Subject syntactic category Syntactic category of the subject of the sentence
Morphological stem Stemmed version of the target word
Verb root Root form of the verb basis for a nominalization
WordNet hypernyms Hypernym set for the target
Figure 20.10 Features commonly used in classic feature-based approaches to event detection.
to the second. Accompanying these notions in most theories is the idea of the cur-
rent moment in time. Combining this notion with the idea of a temporal ordering
relationship yields the familiar notions of past, present, and future.
Various kinds of temporal representation systems can be used to talk about tem-
poral ordering relationship. One of the most commonly used in computational mod-
interval algebra eling is the interval algebra of Allen (1984). Allen models all events and time
expressions as intervals there is no representation for points (although intervals can
be very short). In order to deal with intervals without points, he identifies 13 primi-
tive relations that can hold between these temporal intervals. Fig. 20.11 shows these
Allen relations 13 Allen relations.
A A
A before B B
A overlaps B B
B after A B overlaps' A
A
A
A equals B
B (B equals A)
A meets B
B
B meets' A
A A
A starts B A finishes B
B starts' A B finishes' A
B B
A during B A
B during' A
Time
E R U R,E U E R,U
U,R,E U,R E U E R
Figure 20.12 Reichenbach’s approach applied to various English tenses. In these diagrams,
time flows from left to right, E denotes the time of the event, R denotes the reference time,
and U denotes the time of the utterance.
Languages have many other ways to convey temporal information besides tense.
Most useful for our purposes will be temporal expressions like in the morning or
6:45 or afterwards.
(20.20) I’d like to go at 6:45 in the morning.
(20.21) Somewhere around noon, please.
(20.22) I want to take the train back afterwards.
Incidentally, temporal expressions display a fascinating metaphorical conceptual
organization. Temporal expressions in English are frequently expressed in spatial
terms, as is illustrated by the various uses of at, in, somewhere, and near in these
examples (Lakoff and Johnson 1980, Jackendoff 1983). Metaphorical organizations
such as these, in which one domain is systematically expressed in terms of another,
are very common in languages of the world.
466 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
Delta Air Lines earnings <EVENT eid="e1" class="OCCURRENCE"> soared </EVENT> 33% to a
record in <TIMEX3 tid="t58" type="DATE" value="1989-Q1" anchorTimeID="t57"> the
fiscal first quarter </TIMEX3>, <EVENT eid="e3" class="OCCURRENCE">bucking</EVENT>
the industry trend toward <EVENT eid="e4" class="OCCURRENCE">declining</EVENT>
profits.
BEFORE ENDS
11 9 10
CULMINATES
Figure 20.14 A graph of the text in Eq. 20.35, adapted from (Ocal et al., 2022). TL INKS
are shown in blue, AL INKS in red, and SL INKS in green.
be nouns, proper nouns, adjectives, and adverbs; full temporal expressions consist
of their phrasal projections: noun phrases, adjective phrases, and adverbial phrases
(Figure 20.16).
Category Examples
Noun morning, noon, night, winter, dusk, dawn
Proper Noun January, Monday, Ides, Easter, Rosh Hashana, Ramadan, Tet
Adjective recent, past, annual, former
Adverb hourly, daily, monthly, yearly
Figure 20.16 Examples of temporal lexical triggers.
The task is to detect temporal expressions in running text, like this examples,
shown with TIMEX3 tags (Pustejovsky et al. 2005, Ferro et al. 2005).
A fare increase initiated <TIMEX3>last week</TIMEX3> by UAL
Corp’s United Airlines was matched by competitors over <TIMEX3>the
weekend</TIMEX3>, marking the second successful fare increase in
<TIMEX3>two weeks</TIMEX3>.
Rule-based approaches use cascades of regular expressions to recognize larger
and larger chunks from previous stages, based on patterns containing parts of speech,
trigger words (e.g., February) or classes (e.g., MONTH) (Chang and Manning, 2012;
Strötgen and Gertz, 2013; Chambers, 2013). Here’s a rule from SUTime (Chang and
Manning, 2012) for detecting expressions like 3 years old:
/(\d+)[-\s]($TEUnits)(s)?([-\s]old)?/
Sequence-labeling approaches use the standard IOB scheme, marking words
that are either (I)nside, (O)utside or at the (B)eginning of a temporal expression:
A fare increase initiated last week by UAL Corp’s...
OO O O B I O O O
A statistical sequence labeler is trained, using either embeddings or a fine-tuned
encoder, or classic features extracted from the token and context including words,
lexical triggers, and POS.
Temporal expression recognizers are evaluated with the usual recall, precision,
and F-measures. A major difficulty for all of these very lexicalized approaches is
avoiding expressions that trigger false positives:
(20.36) 1984 tells the story of Winston Smith...
(20.37) ...U2’s classic Sunday Bloody Sunday
are represented via the ISO 8601 standard for encoding temporal values (ISO8601,
2004). Fig. 20.17 reproduces our earlier example with these value attributes.
The dateline, or document date, for this text was July 2, 2007. The ISO repre-
sentation for this kind of expression is YYYY-MM-DD, or in this case, 2007-07-02.
The encodings for the temporal expressions in our sample text all follow from this
date, and are shown here as values for the VALUE attribute.
The first temporal expression in the text proper refers to a particular week of the
year. In the ISO standard, weeks are numbered from 01 to 53, with the first week
of the year being the one that has the first Thursday of the year. These weeks are
represented with the template YYYY-Wnn. The ISO week for our document date is
week 27; thus the value for last week is represented as “2007-W26”.
The next temporal expression is the weekend. ISO weeks begin on Monday;
thus, weekends occur at the end of a week and are fully contained within a single
week. Weekends are treated as durations, so the value of the VALUE attribute has
to be a length. Durations are represented according to the pattern Pnx, where n is
an integer denoting the length and x represents the unit, as in P3Y for three years
or P2D for two days. In this example, one weekend is captured as P1WE. In this
case, there is also sufficient information to anchor this particular weekend as part of
a particular week. Such information is encoded in the ANCHORT IME ID attribute.
Finally, the phrase two weeks also denotes a duration captured as P2W. Figure 20.18
give some more examples, but there is a lot more to the various temporal annotation
standards; consult ISO8601 (2004), Ferro et al. (2005), and Pustejovsky et al. (2005)
for more details.
Unit Pattern Sample Value
Fully specified dates YYYY-MM-DD 1991-09-28
Weeks YYYY-Wnn 2007-W27
Weekends PnWE P1WE
24-hour clock times HH:MM:SS [Link]
Dates and times YYYY-MM-DDTHH:MM:SS 1991-09-28T[Link]
Financial quarters Qn 1999-Q3
Figure 20.18 Sample ISO patterns for representing various times and durations.
temporal to as the document’s temporal anchor. The values of temporal expressions such
anchor
as today, yesterday, or tomorrow can all be computed with respect to this temporal
anchor. The semantic procedure for today simply assigns the anchor, and the attach-
ments for tomorrow and yesterday add a day and subtract a day from the anchor,
respectively. Of course, given the cyclic nature of our representations for months,
weeks, days, and times of day, our temporal arithmetic procedures must use modulo
arithmetic appropriate to the time unit being used.
Unfortunately, even simple expressions such as the weekend or Wednesday in-
troduce a fair amount of complexity. In our current example, the weekend clearly
refers to the weekend of the week that immediately precedes the document date. But
this won’t always be the case, as is illustrated in the following example.
(20.38) Random security checks that began yesterday at Sky Harbor will continue
at least through the weekend.
In this case, the expression the weekend refers to the weekend of the week that the
anchoring date is part of (i.e., the coming weekend). The information that signals
this meaning comes from the tense of continue, the verb governing the weekend.
Relative temporal expressions are handled with temporal arithmetic similar to
that used for today and yesterday. The document date indicates that our example
article is ISO week 27, so the expression last week normalizes to the current week
minus 1. To resolve ambiguous next and last expressions we consider the distance
from the anchoring date to the nearest unit. Next Friday can refer either to the
immediately next Friday or to the Friday following that, but the closer the document
date is to a Friday, the more likely it is that the phrase will skip the nearest one. Such
ambiguities are handled by encoding language and domain-specific heuristics into
the temporal attachments.
supervised by the gold labels in the TimeBank corpus with features like words/em-
beddings, parse paths, tense and aspect The sieve-based architecture using precision-
ranked sets of classifiers, which we’ll introduce in Chapter 23, is also commonly
used.
Systems that perform all 4 tasks (time extraction creation and normalization,
event extraction, and time/event linking) include TARSQI (Verhagen et al., 2005)
C LEARTK (Bethard, 2013), CAEVO (Chambers et al., 2014), and CATENA (Mirza
and Tonelli, 2016).
This template has four slots (LEAD AIRLINE, AMOUNT, EFFECTIVE DATE, FOL -
LOWER ). The next section describes a standard sequence-labeling approach to filling
slots. Section 20.8.2 then describes an older system based on the use of cascades of
finite-state transducers and designed to address a more complex template-filling task
that current learning-based systems don’t yet address.
set of features can be used: tokens, embeddings, word shapes, part-of-speech tags,
syntactic chunk tags, and named entity tags.
role-filler The second system has the job of role-filler extraction. A separate classifier is
extraction
trained to detect each role (LEAD - AIRLINE, AMOUNT, and so on). This can be a
binary classifier that is run on every noun-phrase in the parsed input sentence, or a
sequence model run over sequences of words. Each role classifier is trained on the
labeled data in the training set. Again, the usual set of features can be used, but now
trained only on an individual noun phrase or the fillers of a single slot.
Multiple non-identical text segments might be labeled with the same slot la-
bel. For example in our sample text, the strings United or United Airlines might be
labeled as the L EAD A IRLINE. These are not incompatible choices and the corefer-
ence resolution techniques introduced in Chapter 23 can provide a path to a solution.
A variety of annotated collections have been used to evaluate this style of ap-
proach to template filling, including sets of job announcements, conference calls for
papers, restaurant guides, and biological texts. A key open question is extracting
templates in cases where there is no training data or even predefined templates, by
inducing templates as sets of linked events (Chambers and Jurafsky, 2011).
Tie-up-1 Activity-1:
R ELATIONSHIP tie-up C OMPANY Bridgestone Sports Taiwan Co.
E NTITIES Bridgestone Sports Co. P RODUCT iron and “metal wood” clubs
a local concern S TART DATE DURING: January 1990
a Japanese trading house
J OINT V ENTURE Bridgestone Sports Taiwan Co.
ACTIVITY Activity-1
A MOUNT NT$20000000
Figure 20.19 The templates produced by FASTUS given the input text on page 473.
Early systems for dealing with these complex templates were based on cascades
of transducers based on handwritten rules, as sketched in Fig. 20.20.
The first four stages use handwritten regular expression and grammar rules to
do basic tokenization, chunking, and parsing. Stage 5 then recognizes entities and
events with a recognizer based on finite-state transducers (FSTs), and inserts the rec-
ognized objects into the appropriate slots in templates. This FST recognizer is based
474 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
on hand-built regular expressions like the following (NG indicates Noun-Group and
VG Verb-Group), which matches the first sentence of the news story above.
NG(Company/ies) VG(Set-up) NG(Joint-Venture) with NG(Company/ies)
VG(Produce) NG(Product)
The result of processing these two sentences is the five draft templates (Fig. 20.21)
that must then be merged into the single hierarchical structure shown in Fig. 20.19.
The merging algorithm, after performing coreference resolution, merges two activi-
ties that are likely to be describing the same events.
# Template/Slot Value
1 R ELATIONSHIP : TIE - UP
E NTITIES : Bridgestone Co., a local concern, a Japanese trading house
2 ACTIVITY: PRODUCTION
P RODUCT: “golf clubs”
3 R ELATIONSHIP : TIE - UP
J OINT V ENTURE : “Bridgestone Sports Taiwan Co.”
A MOUNT: NT$20000000
4 ACTIVITY: PRODUCTION
C OMPANY: “Bridgestone Sports Taiwan Co.”
S TART DATE : DURING : January 1990
5 ACTIVITY: PRODUCTION
P RODUCT: “iron and “metal wood” clubs”
Figure 20.21 The five partial templates produced by stage 5 of FASTUS. These templates
are merged in stage 6 to produce the final template shown in Fig. 20.19 on page 473.
20.9 Summary
This chapter has explored techniques for extracting limited forms of semantic con-
tent from texts.
• Relations among entities can be extracted by pattern-based approaches, su-
pervised learning methods when annotated training data is available, lightly
supervised bootstrapping methods when small numbers of seed tuples or
seed patterns are available, distant supervision when a database of relations
is available, and unsupervised or Open IE methods.
• Reasoning about time can be facilitated by detection and normalization of
temporal expressions.
H ISTORICAL N OTES 475
• Events can be ordered in time using sequence models and classifiers trained
on temporally- and event-labeled data like the TimeBank corpus.
Historical Notes
The earliest work on information extraction addressed the template-filling task in the
context of the Frump system (DeJong, 1982). Later work was stimulated by the U.S.
government-sponsored MUC conferences (Sundheim 1991, Sundheim 1992, Sund-
heim 1993, Sundheim 1995). Early MUC systems like CIRCUS system (Lehnert
et al., 1991) and SCISOR (Jacobs and Rau, 1990) were quite influential and inspired
later systems like FASTUS (Hobbs et al., 1997). Chinchor et al. (1993) describe the
MUC evaluation techniques.
Due to the difficulty of porting systems from one domain to another, attention
shifted to machine learning approaches. Early supervised learning approaches to
IE (Cardie 1993, Cardie 1994, Riloff 1993, Soderland et al. 1995, Huffman 1996)
focused on automating the knowledge acquisition process, mainly for finite-state
rule-based systems. Their success, and the earlier success of HMM-based speech
recognition, led to the use of sequence labeling (HMMs: Bikel et al. 1997; MEMMs
McCallum et al. 2000; CRFs: Lafferty et al. 2001), and a wide exploration of fea-
tures (Zhou et al., 2005). Neural approaches followed from the pioneering results of
Collobert et al. (2011), who applied a CRF on top of a convolutional net.
Progress in this area continues to be stimulated by formal evaluations with shared
benchmark datasets, including the Automatic Content Extraction (ACE) evaluations
of 2000-2007 on named entity recognition, relation extraction, and temporal ex-
KBP pressions1 , the KBP (Knowledge Base Population) evaluations (Ji et al. 2010, Sur-
slot filling deanu 2013) of relation extraction tasks like slot filling (extracting attributes (‘slots’)
like age, birthplace, and spouse for a given entity) and a series of SemEval work-
shops (Hendrickx et al., 2009).
Semisupervised relation extraction was first proposed by Hearst (1992b), and
extended by systems like AutoSlog-TS (Riloff, 1996), DIPRE (Brin, 1998), SNOW-
BALL (Agichtein and Gravano, 2000), and Jones et al. (1999). The distant super-
vision algorithm we describe was drawn from Mintz et al. (2009), who first used
the term ‘distant supervision’ (which was suggested to them by Chris Manning)
but similar ideas had occurred in earlier systems like Craven and Kumlien (1999)
and Morgan et al. (2004) under the name weakly labeled data, as well as in Snow
et al. (2005) and Wu and Weld (2007). Among the many extensions are Wu and
Weld (2010), Riedel et al. (2010), and Ritter et al. (2013). Open IE systems include
K NOW I TA LL Etzioni et al. (2005), TextRunner (Banko et al., 2007), and R E V ERB
(Fader et al., 2011). See Riedel et al. (2013) for a universal schema that combines
the advantages of distant supervision and Open IE.
1 [Link]/speech/tests/ace/
476 C HAPTER 20 • I NFORMATION E XTRACTION : R ELATIONS , E VENTS , AND T IME
Exercises
20.1 Acronym expansion, the process of associating a phrase with an acronym, can
be accomplished by a simple form of relational analysis. Develop a system
based on the relation analysis approaches described in this chapter to populate
a database of acronym expansions. If you focus on English Three Letter
Acronyms (TLAs) you can evaluate your system’s performance by comparing
it to Wikipedia’s TLA page.
20.2 Acquire the CMU seminar corpus and develop a template-filling system by
using any of the techniques mentioned in Section 20.8. Analyze how well
your system performs as compared with state-of-the-art results on this corpus.
20.3 A useful functionality in newer email and calendar applications is the ability
to associate temporal expressions connected with events in email (doctor’s
appointments, meeting planning, party invitations, etc.) with specific calendar
entries. Collect a corpus of email containing temporal expressions related to
event planning. How do these expressions compare to the kinds of expressions
commonly found in news text that we’ve been discussing in this chapter?
20.4 For the following sentences, give FOL translations that capture the temporal
relationships between the events.
1. When Mary’s flight departed, I ate lunch.
2. When Mary’s flight departed, I had eaten lunch.
CHAPTER
Sometime between the 7th and 4th centuries BCE, the Indian grammarian Pān.ini1
wrote a famous treatise on Sanskrit grammar, the As.t.ādhyāyı̄ (‘8 books’), a treatise
that has been called “one of the greatest monuments of hu-
man intelligence” (Bloomfield, 1933, 11). The work de-
scribes the linguistics of the Sanskrit language in the form
of 3959 sutras, each very efficiently (since it had to be
memorized!) expressing part of a formal rule system that
brilliantly prefigured modern mechanisms of formal lan-
guage theory (Penn and Kiparsky, 2012). One set of rules
describes the kārakas, semantic relationships between a
verb and noun arguments, roles like agent, instrument, or
destination. Pā[Link]’s work was the earliest we know of
that modeled the linguistic realization of events and their
participants. This task of understanding how participants relate to events—being
able to answer the question “Who did what to whom” (and perhaps also “when and
where”)—is a central question of natural language processing.
Let’s move forward 2.5 millennia to the present and consider the very mundane
goal of understanding text about a purchase of stock by XYZ Corporation. This
purchasing event and its participants can be described by a wide variety of surface
forms. The event can be described by a verb (sold, bought) or a noun (purchase),
and XYZ Corp can be the syntactic subject (of bought), the indirect object (of sold),
or in a genitive or noun compound relation (with the noun purchase) despite having
notionally the same role in all of them:
• XYZ corporation bought the stock.
• They sold the stock to XYZ corporation.
• The stock was bought by XYZ corporation.
• The purchase of the stock by XYZ corporation...
• The stock purchase by XYZ corporation...
In this chapter we introduce a level of representation that captures the common-
ality between these sentences: there was a purchase event, the participants were
XYZ Corp and some stock, and XYZ Corp was the buyer. These shallow semantic
representations , semantic roles, express the role that arguments of a predicate take
in the event, codified in databases like PropBank and FrameNet. We’ll introduce
semantic role labeling, the task of assigning roles to spans in sentences, and selec-
tional restrictions, the preferences that predicates express about their arguments,
such as the fact that the theme of eat is generally something edible.
1 Figure shows a birch bark manuscript from Kashmir of the Rupavatra, a grammatical textbook based
on the Sanskrit grammar of Panini. Image from the Wellcome Collection.
478 C HAPTER 21 • S EMANTIC ROLE L ABELING
Although thematic roles are one of the oldest linguistic models, as we saw above,
their modern formulation is due to Fillmore (1968) and Gruber (1965). Although
there is no universally agreed-upon set of roles, Figs. 21.1 and 21.2 list some the-
matic roles that have been used in various computational papers, together with rough
definitions and examples. Most thematic role sets have about a dozen roles, but we’ll
see sets with smaller numbers of roles with even more abstract meanings, and sets
with very large numbers of roles that are specific to situations. We’ll use the general
semantic roles term semantic roles for all sets of roles, whether small or large.
Semantic roles thus help generalize over different surface realizations of pred-
icate arguments. For example, while the AGENT is often realized as the subject of
the sentence, in other cases the THEME can be the subject. Consider these possible
realizations of the thematic arguments of the verb break:
(21.3) John broke the window.
AGENT THEME
(21.4) John broke the window with a rock.
AGENT THEME INSTRUMENT
(21.5) The rock broke the window.
INSTRUMENT THEME
(21.6) The window broke.
THEME
(21.7) The window was broken by John.
THEME AGENT
These examples suggest that break has (at least) the possible arguments AGENT,
THEME , and INSTRUMENT. The set of thematic role arguments taken by a verb is
thematic grid often called the thematic grid, θ -grid, or case frame. We can see that there are
case frame (among others) the following possibilities for the realization of these arguments of
break:
AGENT/Subject, THEME /Object
AGENT/Subject, THEME /Object, INSTRUMENT/PPwith
INSTRUMENT/Subject, THEME /Object
THEME /Subject
It turns out that many verbs allow their thematic roles to be realized in various
syntactic positions. For example, verbs like give can realize the THEME and GOAL
arguments in two different ways:
(21.8) a. Doris gave the book to Cary.
AGENT THEME GOAL
These multiple argument structure realizations (the fact that break can take AGENT,
INSTRUMENT, or THEME as subject, and give can realize its THEME and GOAL in
verb either order) are called verb alternations or diathesis alternations. The alternation
alternation
dative we showed above for give, the dative alternation, seems to occur with particular se-
alternation
mantic classes of verbs, including “verbs of future having” (advance, allocate, offer,
480 C HAPTER 21 • S EMANTIC ROLE L ABELING
owe), “send verbs” (forward, hand, mail), “verbs of throwing” (kick, pass, throw),
and so on. Levin (1993) lists for 3100 English verbs the semantic classes to which
they belong (47 high-level classes, divided into 193 more specific classes) and the
various alternations in which they participate. These lists of verb classes have been
incorporated into the online resource VerbNet (Kipper et al., 2000), which links each
verb to both WordNet and FrameNet entries.
(21.14) [Arg0 Big Fruit Co. ] increased [Arg1 the price of bananas].
(21.15) [Arg1 The price of bananas] was increased again [Arg0 by Big Fruit Co. ]
(21.16) [Arg1 The price of bananas] increased [Arg2 5%].
PropBank also has a number of non-numbered arguments called ArgMs, (ArgM-
TMP, ArgM-LOC, etc.) which represent modification or adjunct meanings. These
are relatively stable across predicates, so aren’t listed with each frame file. Data
labeled with these modifiers can be helpful in training systems to detect temporal,
location, or directional modification across predicates. Some of the ArgMs include:
TMP when? yesterday evening, now
LOC where? at the museum, in San Francisco
DIR where to/from? down, to Bangkok
MNR how? clearly, with much enthusiasm
PRP/CAU why? because ... , in response to the ruling
REC themselves, each other
ADV miscellaneous
PRD secondary predication ...ate the meat raw
NomBank While PropBank focuses on verbs, a related project, NomBank (Meyers et al.,
2004) adds annotations to noun predicates. For example the noun agreement in
Apple’s agreement with IBM would be labeled with Apple as the Arg0 and IBM as
the Arg2. This allows semantic role labelers to assign labels to arguments of both
verbal and nominal predicates.
21.5 FrameNet
While making inferences about the semantic commonalities across different sen-
tences with increase is useful, it would be even more useful if we could make such
inferences in many more situations, across different verbs, and also between verbs
and nouns. For example, we’d like to extract the similarity among these three sen-
tences:
(21.17) [Arg1 The price of bananas] increased [Arg2 5%].
(21.18) [Arg1 The price of bananas] rose [Arg2 5%].
(21.19) There has been a [Arg2 5%] rise [Arg1 in the price of bananas].
Note that the second example uses the different verb rise, and the third example
uses the noun rather than the verb rise. We’d like a system to recognize that the
price of bananas is what went up, and that 5% is the amount it went up, no matter
whether the 5% appears as the object of the verb increased or as a nominal modifier
of the noun rise.
FrameNet The FrameNet project is another semantic-role-labeling project that attempts
to address just these kinds of problems (Baker et al. 1998, Fillmore et al. 2003,
Fillmore and Baker 2009, Ruppenhofer et al. 2016). Whereas roles in the PropBank
project are specific to an individual verb, roles in the FrameNet project are specific
to a frame.
What is a frame? Consider the following set of words:
reservation, flight, travel, buy, price, cost, fare, rates, meal, plane
There are many individual lexical relations of hyponymy, synonymy, and so on
between many of the words in this list. The resulting set of relations does not,
21.5 • F RAME N ET 483
however, add up to a complete account of how these words are related. They are
clearly all defined with respect to a coherent chunk of common-sense background
information concerning air travel.
frame We call the holistic background knowledge that unites these words a frame (Fill-
more, 1985). The idea that groups of words are defined with respect to some back-
ground information is widespread in artificial intelligence and cognitive science,
model where besides frame we see related works like a model (Johnson-Laird, 1983), or
script even script (Schank and Abelson, 1977).
A frame in FrameNet is a background knowledge structure that defines a set of
frame elements frame-specific semantic roles, called frame elements, and includes a set of predi-
cates that use these roles. Each word evokes a frame and profiles some aspect of the
frame and its elements. The FrameNet dataset includes a set of frames and frame
elements, the lexical units associated with each frame, and a set of labeled exam-
ple sentences. For example, the change position on a scale frame is defined as
follows:
This frame consists of words that indicate the change of an Item’s posi-
tion on a scale (the Attribute) from a starting point (Initial value) to an
end point (Final value).
Some of the semantic roles (frame elements) in the frame are defined as in
core roles Fig. 21.3. Note that these are separated into core roles, which are frame specific, and
non-core roles non-core roles, which are more like the Arg-M arguments in PropBank, expressing
more general properties of time, location, and so on.
Core Roles
ATTRIBUTE The ATTRIBUTE is a scalar property that the I TEM possesses.
D IFFERENCE The distance by which an I TEM changes its position on the scale.
F INAL STATE A description that presents the I TEM’s state after the change in the ATTRIBUTE’s
value as an independent predication.
F INAL VALUE The position on the scale where the I TEM ends up.
I NITIAL STATE A description that presents the I TEM’s state before the change in the AT-
TRIBUTE ’s value as an independent predication.
I NITIAL VALUE The initial position on the scale from which the I TEM moves away.
I TEM The entity that has a position on the scale.
VALUE RANGE A portion of the scale, typically identified by its end points, along which the
values of the ATTRIBUTE fluctuate.
Some Non-Core Roles
D URATION The length of time over which the change takes place.
S PEED The rate of change of the VALUE.
G ROUP The G ROUP in which an I TEM changes the value of an
ATTRIBUTE in a specified way.
Figure 21.3 The frame elements in the change position on a scale frame from the FrameNet Labelers
Guide (Ruppenhofer et al., 2016).
(21.24) a steady increase [I NITIAL VALUE from 9.5] [F INAL VALUE to 14.3] [I TEM
in dividends]
(21.25) a [D IFFERENCE 5%] [I TEM dividend] increase...
Note from these example sentences that the frame includes target words like rise,
fall, and increase. In fact, the complete frame consists of the following words:
VERBS: dwindle move soar escalation shift
advance edge mushroom swell explosion tumble
climb explode plummet swing fall
decline fall reach triple fluctuation ADVERBS:
decrease fluctuate rise tumble gain increasingly
diminish gain rocket growth
dip grow shift NOUNS: hike
double increase skyrocket decline increase
drop jump slide decrease rise
FrameNet also codes relationships between frames, allowing frames to inherit
from each other, or representing relations between frames like causation (and gen-
eralizations among frame elements in different frames can be represented by inheri-
tance as well). Thus, there is a Cause change of position on a scale frame that is
linked to the Change of position on a scale frame by the cause relation, but that
adds an AGENT role and is used for causative examples such as the following:
(21.26) [AGENT They] raised [I TEM the price of their soda] [D IFFERENCE by 2%].
Together, these two frames would allow an understanding system to extract the
common event semantics of all the verbal and nominal causative and non-causative
usages.
FrameNets have also been developed for many other languages including Span-
ish, German, Japanese, Portuguese, Italian, and Chinese.
Figure 21.5 shows a parse of (21.28) above. The parse is then traversed to find all
words that are predicates.
For each of these predicates, the algorithm examines each node in the parse
tree and uses supervised classification to decide the semantic role (if any) it plays
for this predicate. Given a labeled training set such as PropBank or FrameNet, a
feature vector is extracted for each node, using feature templates described in the
next subsection. A 1-of-N classifier is then trained to predict a semantic role for
each constituent given these features, where N is the number of potential semantic
roles plus an extra NONE role for non-role constituents. Any standard classification
algorithms can be used. Finally, for each test sentence to be labeled, the classifier is
run on each relevant constituent.
parse ← PARSE(words)
for each predicate in parse do
for each node in parse do
featurevector ← E XTRACT F EATURES(node, predicate, parse)
C LASSIFY N ODE(node, featurevector, parse)
NP-SBJ = ARG0 VP
issued DT JJ NN IN NP
noon yesterday
Figure 21.5 Parse tree for a PropBank sentence, showing the PropBank argument labels. The dotted line
shows the path feature NP↑S↓VP↓VBD for ARG0, the NP-SBJ constituent The San Francisco Examiner.
The separation of identification and classification may lead to better use of fea-
tures (different features may be useful for the two tasks) or to computational effi-
ciency.
Global Optimization
The classification algorithm of Fig. 21.5 classifies each argument separately (‘lo-
cally’), making the simplifying assumption that each argument of a predicate can be
labeled independently. This assumption is false; there are interactions between argu-
ments that require a more ‘global’ assignment of labels to constituents. For example,
constituents in FrameNet and PropBank are required to be non-overlapping. More
significantly, the semantic roles of constituents are not independent. For example
PropBank does not allow multiple identical arguments; two constituents of the same
verb cannot both be labeled ARG 0 .
Role labeling systems thus often add a fourth step to deal with global consistency
across the labels in a sentence. For example, the local classifiers can return a list of
possible labels associated with probabilities for each constituent, and a second-pass
Viterbi decoding or re-ranking approach can be used to choose the best consensus
label. Integer linear programming (ILP) is another common way to choose a solution
that conforms best to multiple constraints.
Other features are often used in addition, such as sets of n-grams inside the
constituent, or more complex versions of the path features (the upward or downward
halves, or whether particular nodes occur in the path).
It’s also possible to use dependency parses instead of constituency parses as the
basis of features, for example using dependency parse paths instead of constituency
paths.
Softmax
concatenate
with predicate
ENCODER
As with all the taggers, the goal is to compute the highest probability tag se-
quence ŷ, given the input sequence of words w:
ŷ = argmax P(y|w)
y∈T
Fig. 21.6 shows a sketch of a standard algorithm from He et al. (2017). Here each
input word is mapped to pretrained embeddings, and then each token is concatenated
with the predicate embedding and then passed through a feedforward network with
a softmax which outputs a distribution over each SRL label. For decoding, a CRF
layer can be used instead of the MLP layer on top of the biLSTM output to do global
inference, but in practice this doesn’t seem to provide much benefit.
488 C HAPTER 21 • S EMANTIC ROLE L ABELING
With this representation, all we know about y, the filler of the THEME role, is that
it is associated with an Eating event through the Theme relation. To stipulate the
selectional restriction that y must be something edible, we simply add a new term to
that effect:
When a phrase like ate a hamburger is encountered, a semantic analyzer can form
the following kind of representation:
Sense 1
hamburger, beefburger --
(a fried cake of minced beef served on a bun)
=> sandwich
=> snack food
=> dish
=> nutriment, nourishment, nutrition...
=> food, nutrient
=> substance
=> matter
=> physical entity
=> entity
Figure 21.7 Evidence from WordNet that hamburgers are edible.
(21.36) But it fell apart in 1931, perhaps because people realized you can’t eat
gold for lunch if you’re hungry.
(21.37) In his two championship trials, Mr. Kulkarni ate glass on an empty
stomach, accompanied only by water and tea.
Modern systems for selectional preferences therefore specify the relation be-
tween a predicate and its possible arguments with soft constraints of some kind.
Selectional Association
selectional
One of the most influential has been the selectional association model of Resnik
preference (1993). Resnik defines the idea of selectional preference strength as the general
strength
amount of information that a predicate tells us about the semantic class of its argu-
ments. For example, the verb eat tells us a lot about the semantic class of its direct
objects, since they tend to be edible. The verb be, by contrast, tells us less about
its direct objects. The selectional preference strength can be defined by the differ-
ence in information between two distributions: the distribution of expected semantic
classes P(c) (how likely is it that a direct object will fall into class c) and the dis-
tribution of expected semantic classes for the particular verb P(c|v) (how likely is
it that the direct object of the specific verb v will fall into semantic class c). The
greater the difference between these distributions, the more information the verb
is giving us about possible objects. The difference between these two distributions
relative entropy can be quantified by relative entropy, or the Kullback-Leibler divergence (Kullback
KL divergence and Leibler, 1951). The Kullback-Leibler or KL divergence D(P||Q) expresses the
21.7 • S ELECTIONAL R ESTRICTIONS 491
The selectional preference SR (v) uses the KL divergence to express how much in-
formation, in bits, the verb v expresses about the possible semantic class of its argu-
ment.
SR (v) = D(P(c|v)||P(c))
X P(c|v)
= P(c|v) log (21.39)
c
P(c)
selectional Resnik then defines the selectional association of a particular class and verb as the
association
relative contribution of that class to the general selectional preference of the verb:
1 P(c|v)
AR (v, c) = P(c|v) log (21.40)
SR (v) P(c)
The selectional association is thus a probabilistic measure of the strength of asso-
ciation between a predicate and a class dominating the argument to the predicate.
Resnik estimates the probabilities for these associations by parsing a corpus, count-
ing all the times each predicate occurs with each argument word, and assuming
that each word is a partial observation of all the WordNet concepts containing the
word. The following table from Resnik (1996) shows some sample high and low
selectional associations for verbs and some WordNet semantic classes of their direct
objects.
Direct Object Direct Object
Verb Semantic Class Assoc Semantic Class Assoc
read WRITING 6.80 ACTIVITY -.20
write WRITING 7.26 COMMERCE 0
see ENTITY 5.79 METHOD -0.01
Primitive Definition
ATRANS The abstract transfer of possession or control from one entity to
another
P TRANS The physical transfer of an object from one location to another
M TRANS The transfer of mental concepts between entities or within an
entity
M BUILD The creation of new information within an entity
P ROPEL The application of physical force to move an object
M OVE The integral movement of a body part by an animal
I NGEST The taking in of a substance by an animal
E XPEL The expulsion of something from an animal
S PEAK The action of producing a sound
ATTEND The action of focusing a sense organ
Figure 21.8 A set of conceptual dependency primitives.
Below is an example sentence along with its CD representation. The verb brought
is translated into the two primitives ATRANS and PTRANS to indicate that the waiter
both physically conveyed the check to Mary and passed control of it to her. Note
that CD also associates a fixed set of thematic roles with each primitive to represent
the various participants in the action.
(21.47) The waiter brought Mary the check.
21.9 Summary
• Semantic roles are abstract models of the role an argument plays in the event
described by the predicate.
• Thematic roles are a model of semantic roles based on a single finite list of
roles. Other semantic role models include per-verb semantic role lists and
proto-agent/proto-patient, both of which are implemented in PropBank,
and per-frame role lists, implemented in FrameNet.
494 C HAPTER 21 • S EMANTIC ROLE L ABELING
• Semantic role labeling is the task of assigning semantic role labels to the
constituents of a sentence. The task is generally treated as a supervised ma-
chine learning task, with models trained on PropBank or FrameNet. Algo-
rithms generally start by parsing a sentence and then automatically tag each
parse tree node with a semantic role. Neural models map straight from words
end-to-end.
• Semantic selectional restrictions allow words (particularly predicates) to post
constraints on the semantic properties of their argument words. Selectional
preference models (like selectional association or simple conditional proba-
bility) allow a weight or probability to be assigned to the association between
a predicate and an argument word or class.
Historical Notes
Although the idea of semantic roles dates back to Pā[Link], they were re-introduced
into modern linguistics by Gruber (1965), Fillmore (1966) and Fillmore (1968). Fill-
more had become interested in argument structure by studying Lucien Tesnière’s
groundbreaking Éléments de Syntaxe Structurale (Tesnière, 1959) in which the term
‘dependency’ was introduced and the foundations were laid for dependency gram-
mar. Following Tesnière’s terminology, Fillmore first referred to argument roles as
actants (Fillmore, 1966) but quickly switched to the term case, (see Fillmore (2003))
and proposed a universal list of semantic roles or cases (Agent, Patient, Instrument,
etc.), that could be taken on by the arguments of predicates. Verbs would be listed in
the lexicon with their case frame, the list of obligatory (or optional) case arguments.
The idea that semantic roles could provide an intermediate level of semantic
representation that could help map from syntactic parse structures to deeper, more
fully-specified representations of meaning was quickly adopted in natural language
processing, and systems for extracting case frames were created for machine transla-
tion (Wilks, 1973), question-answering (Hendrix et al., 1973), spoken-language pro-
cessing (Nash-Webber, 1975), and dialogue systems (Bobrow et al., 1977). General-
purpose semantic role labelers were developed. The earliest ones (Simmons, 1973)
first parsed a sentence by means of an ATN (Augmented Transition Network) parser.
Each verb then had a set of rules specifying how the parse should be mapped to se-
mantic roles. These rules mainly made reference to grammatical functions (subject,
object, complement of specific prepositions) but also checked constituent internal
features such as the animacy of head nouns. Later systems assigned roles from pre-
built parse trees, again by using dictionaries with verb-specific case frames (Levin
1977, Marcus 1980).
By 1977 case representation was widely used and taught in AI and NLP courses,
and was described as a standard of natural language processing in the first edition of
Winston’s 1977 textbook Artificial Intelligence.
In the 1980s Fillmore proposed his model of frame semantics, later describing
the intuition as follows:
“The idea behind frame semantics is that speakers are aware of possi-
bly quite complex situation types, packages of connected expectations,
that go by various names—frames, schemas, scenarios, scripts, cultural
narratives, memes—and the words in our language are understood with
such frames as their presupposed background.” (Fillmore, 2012, p. 712)
H ISTORICAL N OTES 495
The word frame seemed to be in the air for a suite of related notions proposed at
about the same time by Minsky (1974), Hymes (1974), and Goffman (1974), as
well as related notions with other names like scripts (Schank and Abelson, 1975)
and schemata (Bobrow and Norman, 1975) (see Tannen (1979) for a comparison).
Fillmore was also influenced by the semantic field theorists and by a visit to the Yale
AI lab where he took notice of the lists of slots and fillers used by early information
extraction systems like DeJong (1982) and Schank and Abelson (1977). In the 1990s
Fillmore drew on these insights to begin the FrameNet corpus annotation project.
At the same time, Beth Levin drew on her early case frame dictionaries (Levin,
1977) to develop her book which summarized sets of verb classes defined by shared
argument realizations (Levin, 1993). The VerbNet project built on this work (Kipper
et al., 2000), leading soon afterwards to the PropBank semantic-role-labeled corpus
created by Martha Palmer and colleagues (Palmer et al., 2005).
The combination of rich linguistic annotation and corpus-based approach in-
stantiated in FrameNet and PropBank led to a revival of automatic approaches to
semantic role labeling, first on FrameNet (Gildea and Jurafsky, 2000) and then on
PropBank data (Gildea and Palmer, 2002, inter alia). The problem first addressed in
the 1970s by handwritten rules was thus now generally recast as one of supervised
machine learning enabled by large and consistent databases. Many popular features
used for role labeling are defined in Gildea and Jurafsky (2002), Surdeanu et al.
(2003), Xue and Palmer (2004), Pradhan et al. (2005), Che et al. (2009), and Zhao
et al. (2009). The use of dependency rather than constituency parses was introduced
in the CoNLL-2008 shared task (Surdeanu et al., 2008). For surveys see Palmer
et al. (2010) and Màrquez et al. (2008).
The use of neural approaches to semantic role labeling was pioneered by Col-
lobert et al. (2011), who applied a CRF on top of a convolutional net. Early work
like Foland, Jr. and Martin (2015) focused on using dependency features. Later work
eschewed syntactic features altogether; Zhou and Xu (2015b) introduced the use of
a stacked (6-8 layer) biLSTM architecture, and (He et al., 2017) showed how to
augment the biLSTM architecture with highway networks and also replace the CRF
with A* decoding that make it possible to apply a wide variety of global constraints
in SRL decoding.
Most semantic role labeling schemes only work within a single sentence, fo-
cusing on the object of the verbal (or nominal, in the case of NomBank) predicate.
However, in many cases, a verbal or nominal predicate may have an implicit argu-
implicit
argument ment: one that appears only in a contextual sentence, or perhaps not at all and must
be inferred. In the two sentences This house has a new owner. The sale was finalized
10 days ago. the sale in the second sentence has no A RG 1, but a reasonable reader
would infer that the Arg1 should be the house mentioned in the prior sentence. Find-
iSRL ing these arguments, implicit argument detection (sometimes shortened as iSRL)
was introduced by Gerber and Chai (2010) and Ruppenhofer et al. (2010). See Do
et al. (2017) for more recent neural models.
To avoid the need for huge labeled training sets, unsupervised approaches for
semantic role labeling attempt to induce the set of semantic roles by clustering over
arguments. The task was pioneered by Riloff and Schmelzenbach (1998) and Swier
and Stevenson (2004); see Grenager and Manning (2006), Titov and Klementiev
(2012), Lang and Lapata (2014), Woodsend and Lapata (2015), and Titov and Khod-
dam (2014).
Recent innovations in frame labeling include connotation frames, which mark
richer information about the argument of predicates. Connotation frames mark the
496 C HAPTER 21 • S EMANTIC ROLE L ABELING
sentiment of the writer or reader toward the arguments (for example using the verb
survive in he survived a bombing expresses the writer’s sympathy toward the subject
he and negative sentiment toward the bombing. See Chapter 22 for more details.
Selectional preference has been widely studied beyond the selectional associa-
tion models of Resnik (1993) and Resnik (1996). Methods have included clustering
(Rooth et al., 1999), discriminative learning (Bergsma et al., 2008a), and topic mod-
els (Séaghdha 2010, Ritter et al. 2010), and constraints can be expressed at the level
of words or classes (Agirre and Martinez, 2001). Selectional preferences have also
been successfully integrated into semantic role labeling (Erk 2007, Zapirain et al.
2013, Do et al. 2017).
Exercises
CHAPTER
affective In this chapter we turn to tools for interpreting affective meaning, extending our
study of sentiment analysis in Appendix K. We use the word ‘affective’, following
the tradition in affective computing (Picard, 1995) to mean emotion, sentiment, per-
subjectivity sonality, mood, and attitudes. Affective meaning is closely related to subjectivity,
the study of a speaker or writer’s evaluations, opinions, emotions, and speculations
(Wiebe et al., 1999).
How should affective meaning be defined? One influential typology of affec-
tive states comes from Scherer (2000), who defines each class of affective states by
factors like its cognitive realization and time course (Fig. 22.1).
We can design extractors for each of these kinds of affective states. Appendix K
already introduced sentiment analysis, the task of extracting the positive or negative
orientation that a writer expresses in a text. This corresponds in Scherer’s typology
to the extraction of attitudes: figuring out what people like or dislike, from affect-
rich texts like consumer reviews of books or movies, newspaper editorials, or public
sentiment in blogs or tweets.
Detecting emotion and moods is useful for detecting whether a student is con-
fused, engaged, or certain when interacting with a tutorial system, whether a caller
to a help line is frustrated, whether someone’s blog posts or tweets indicated depres-
sion. Detecting emotions like fear in novels, for example, could help us trace what
groups or situations are feared and how that changes over time.
498 C HAPTER 22 • L EXICONS FOR S ENTIMENT, A FFECT, AND C ONNOTATION
1962, Plutchik 1962), a model dating back to Darwin. Perhaps the most well-known
of this family of theories are the 6 emotions proposed by Ekman (e.g., Ekman 1999)
to be universally present in all cultures: surprise, happiness, anger, fear, disgust,
sadness. Another atomic theory is the Plutchik (1980) wheel of emotion, consisting
of 8 basic emotions in four opposing pairs: joy–sadness, anger–fear, trust–disgust,
and anticipation–surprise, together with the emotions derived from them, shown in
Fig. 22.2.
The second class of emotion theories widely used in NLP views emotion as a
space in 2 or 3 dimensions (Russell, 1980). Most models include the two dimensions
valence and arousal, and many add a third, dominance. These can be defined as:
valence: the pleasantness of the stimulus
arousal: the level of alertness, activeness, or energy provoked by the stimulus
dominance: the degree of control or dominance exerted by the stimulus or the
emotion
Sentiment can be viewed as a special case of this second view of emotions as points
in space. In particular, the valence dimension, measuring how pleasant or unpleasant
a word is, is often used directly as a measure of sentiment.
In these lexicon-based models of affect, the affective meaning of a word is gen-
erally fixed, irrespective of the linguistic context in which a word is used, or the
dialect or culture of the speaker. By contrast, other models in affective science repre-
sent emotions as much richer processes involving cognition (Barrett et al., 2007). In
appraisal theory, for example, emotions are complex processes, in which a person
considers how an event is congruent with their goals, taking into account variables
like the agency, certainty, urgency, novelty and control associated with the event
(Moors et al., 2013). Computational models in NLP taking into account these richer
theories of emotion will likely play an important role in future work.
500 C HAPTER 22 • L EXICONS FOR S ENTIMENT, A FFECT, AND C ONNOTATION
Positive admire, amazing, assure, celebration, charm, eager, enthusiastic, excellent, fancy, fan-
tastic, frolic, graceful, happy, joy, luck, majesty, mercy, nice, patience, perfect, proud,
rejoice, relief, respect, satisfactorily, sensational, super, terrific, thank, vivid, wise, won-
derful, zest
Negative abominable, anger, anxious, bad, catastrophe, cheap, complaint, condescending, deceit,
defective, disappointment, embarrass, fake, fear, filthy, fool, guilt, hate, idiot, inflict, lazy,
miserable, mourn, nervous, objection, pest, plot, reject, scream, silly, terrible, unfriendly,
vile, wicked
Figure 22.3 Some words with consistent sentiment across the General Inquirer (Stone et al., 1966), the
MPQA Subjectivity lexicon (Wilson et al., 2005), and the polarity lexicon of Hu and Liu (2004).
Slightly more general than these sentiment lexicons are lexicons that assign each
word a value on all three affective dimensions. The NRC Valence, Arousal, and
Dominance (VAD) lexicon (Mohammad, 2018a) assigns valence, arousal, and dom-
inance scores to 20,000 words. Some examples are shown in Fig. 22.4.
EmoLex The NRC Word-Emotion Association Lexicon, also called EmoLex (Moham-
mad and Turney, 2013), uses the Plutchik (1980) 8 basic emotions defined above.
The lexicon includes around 14,000 words including words from prior lexicons as
well as frequent nouns, verbs, adverbs and adjectives. Values from the lexicon for
some sample words:
22.3 • C REATING A FFECT L EXICONS BY H UMAN L ABELING 501
anticipation
negative
surprise
positive
sadness
disgust
anger
trust
fear
joy
Word
reward 0 1 0 0 1 0 1 1 1 0
worry 0 1 0 1 0 1 0 0 0 1
tenderness 0 0 0 0 1 0 0 0 1 0
sweetheart 0 1 0 0 1 1 0 1 1 0
suddenly 0 0 0 0 0 0 1 0 0 0
thirst 0 1 0 0 0 1 1 0 0 0
garbage 0 0 1 0 0 0 0 0 0 1
For a smaller set of 5,814 words, the NRC Emotion/Affect Intensity Lexicon
(Mohammad, 2018b) contains real-valued scores of association for anger, fear, joy,
and sadness; Fig. 22.5 shows examples.
LIWC LIWC, Linguistic Inquiry and Word Count, is a widely used set of 73 lex-
icons containing over 2300 words (Pennebaker et al., 2007), designed to capture
aspects of lexical meaning relevant for social psychological tasks. In addition to
sentiment-related lexicons like ones for negative emotion (bad, weird, hate, prob-
lem, tough) and positive emotion (love, nice, sweet), LIWC includes lexicons for
categories like anger, sadness, cognitive mechanisms, perception, tentative, and in-
hibition, shown in Fig. 22.6.
There are various other hand-built affective lexicons. The General Inquirer in-
cludes additional lexicons for dimensions like strong vs. weak, active vs. passive,
overstated vs. understated, as well as lexicons for categories like pleasure, pain,
virtue, vice, motivation, and cognitive orientation.
concrete Another useful feature for various tasks is the distinction between concrete
abstract words like banana or bathrobe and abstract words like belief and although. The
lexicon in Brysbaert et al. (2014) used crowdsourcing to assign a rating from 1 to 5
of the concreteness of 40,000 words, thus assigning banana, bathrobe, and bagel 5,
belief 1.19, although 1.07, and in between words like brisk a 2.5.
Positive Negative
Emotion Emotion Insight Inhibition Family Negate
appreciat* anger* aware* avoid* brother* aren’t
comfort* bore* believe careful* cousin* cannot
great cry decid* hesitat* daughter* didn’t
happy despair* feel limit* family neither
interest fail* figur* oppos* father* never
joy* fear know prevent* grandf* no
perfect* griev* knew reluctan* grandm* nobod*
please* hate* means safe* husband none
safe* panic* notice* stop mom nor
terrific suffers recogni* stubborn* mother nothing
value terrify sense wait niece* nowhere
wow* violent* think wary wife without
Figure 22.6 Samples from 5 of the 73 lexical categories in LIWC (Pennebaker et al., 2007).
The * means the previous letters are a word prefix and all words with that prefix are included
in the category.
tators. Let’s take a look at some of the methodological choices for two crowdsourced
emotion lexicons.
The NRC Emotion Lexicon (EmoLex) (Mohammad and Turney, 2013), labeled
emotions in two steps. To ensure that the annotators were judging the correct sense
of the word, they first answered a multiple-choice synonym question that primed
the correct sense of the word (without requiring the annotator to read a potentially
confusing sense definition). These were created automatically using the headwords
associated with the thesaurus category of the sense in question in the Macquarie
dictionary and the headwords of 3 random distractor categories. An example:
Which word is closest in meaning (most related) to startle?
• automobile
• shake
• honesty
• entertain
For each word (e.g. startle), the annotator was then asked to rate how associated
that word is with each of the 8 emotions (joy, fear, anger, etc.). The associations
were rated on a scale of not, weakly, moderately, and strongly associated. Outlier
ratings were removed, and then each term was assigned the class chosen by the ma-
jority of the annotators, with ties broken by choosing the stronger intensity, and then
the 4 levels were mapped into a binary label for each word (no and weak mapped to
0, moderate and strong mapped to 1).
The NRC VAD Lexicon (Mohammad, 2018a) was built by selecting words and
emoticons from prior lexicons and annotating them with crowd-sourcing using best-
best-worst
scaling worst scaling (Louviere et al. 2015, Kiritchenko and Mohammad 2017). In best-
worst scaling, annotators are given N items (usually 4) and are asked which item is
the best (highest) and which is the worst (lowest) in terms of some property. The
set of words used to describe the ends of the scales are taken from prior literature.
For valence, for example, the raters were asked:
Q1. Which of the four words below is associated with the MOST happi-
ness / pleasure / positiveness / satisfaction / contentedness / hopefulness
OR LEAST unhappiness / annoyance / negativeness / dissatisfaction /
22.4 • S EMI - SUPERVISED I NDUCTION OF A FFECT L EXICONS 503
on a specific corpus (for example using a financial corpus if a finance lexicon is the
goal), or we can finetune off-the-shelf embeddings to a corpus. Fine-tuning is espe-
cially important if we have a very specific genre of text but don’t have enough data
to train good embeddings. In finetuning, we begin with off-the-shelf embeddings
like word2vec, and continue training them on the small target corpus.
Once we have embeddings for each pole word, we create an embedding that
represents each pole by taking the centroid of the embeddings of each of the seed
words; recall that the centroid is the multidimensional version of the mean. Given
a set of embeddings for the positive seed words S+ = {E(w+ +
1 ), E(w2 ), ..., E(wn )},
+
− − − −
and embeddings for the negative seed words S = {E(w1 ), E(w2 ), ..., E(wm )}, the
pole centroids are:
n
1X
V+ = E(w+
i )
n
1
m
1X
V− = E(w−
i ) (22.1)
m
1
The semantic axis defined by the poles is computed just by subtracting the two vec-
tors:
Vaxis = V+ − V− (22.2)
Vaxis , the semantic axis, is a vector in the direction of positive sentiment. Finally,
we compute (via cosine similarity) the angle between the vector in the direction of
positive sentiment and the direction of w’s embedding. A higher cosine means that
w is more aligned with S+ than S− .
score(w) = cos E(w), Vaxis
E(w) · Vaxis
= (22.3)
kE(w)kkVaxis k
then choose a node to move to with probability proportional to the edge prob-
ability. A word’s polarity score for a seed set is proportional to the probability
of a random walk from the seed set landing on that word (Fig. 22.7).
4. Create word scores: We walk from both positive and negative seed sets,
resulting in positive (rawscore+ (wi )) and negative (rawscore− (wi )) raw label
scores. We then combine these values into a positive-polarity score as:
rawscore+ (wi )
score+ (wi ) = (22.5)
rawscore+ (wi ) + rawscore− (wi )
It’s often helpful to standardize the scores to have zero mean and unit variance
within a corpus.
5. Assign confidence to each score: Because sentiment scores are influenced by
the seed set, we’d like to know how much the score of a word would change if
a different seed set is used. We can use bootstrap sampling to get confidence
regions, by computing the propagation B times over random subsets of the
positive and negative seed sets (for example using B = 50 and choosing 7 of
the 10 seed words each time). The standard deviation of the bootstrap sampled
polarity scores gives a confidence measure.
loathe loathe
like like
abhor abhor
(a) (b)
Figure 22.7 Intuition of the S ENT P ROP algorithm. (a) Run random walks from the seed words. (b) Assign
polarity scores (shown here as colors green or red) based on the frequency of random walk visits.
associated review score: a value that may range from 1 star to 5 stars, or scoring 1
to 10. Fig. 22.9 shows samples extracted from restaurant, book, and movie reviews.
We can use this review score as supervision: positive words are more likely to
appear in 5-star reviews; negative words in 1-star reviews. And instead of just a
binary polarity, this kind of supervision allows us to assign a word a more complex
representation of its polarity: its distribution over stars (or other scores).
Thus in a ten-star system we could represent the sentiment of each word as a
10-tuple, each number a score representing the word’s association with that polarity
level. This association can be a raw count, or a likelihood P(w|c), or some other
function of the count, for each class c from 1 to 10.
For example, we could compute the IMDb likelihood of a word like disap-
point(ed/ing) occurring in a 1 star review by dividing the number of times disap-
point(ed/ing) occurs in 1-star reviews in the IMDb dataset (8,557) by the total num-
ber of words occurring in 1-star reviews (25,395,214), so the IMDb estimate of
P(disappointing|1) is .0003.
A slight modification of this weighting, the normalized likelihood, can be used
as an illuminating visualization (Potts, 2011)1
count(w, c)
P(w|c) = P
w∈C count(w, c)
P(w|c)
PottsScore(w) = P (22.6)
c P(w|c)
Dividing the IMDb estimate P(disappointing|1) of .0003 by the sum of the likeli-
hood P(w|c) over all categories gives a Potts score of 0.10. The word disappointing
thus is associated with the vector [.10, .12, .14, .14, .13, .11, .08, .06, .06, .05]. The
1 Each element of the Potts score of a word w and category c can be shown to be a variant of the
pointwise mutual information pmi(w, c) without the log term; see Exercise 22.1.
IMDB
Cat = 0.3
Cat^2 = -
0.09
0.05
Potts diagram Potts diagram (Potts, 2011) is a visualization of these word scores, representing the
-0.50
-0.39
-0.28
prior sentiment of a word as a distribution over the rating categories.
Fig. 22.10 shows the Potts diagrams for 3 positive and 3 negative scalar adjec-
tives. Note that the curve for strongly positive scalars have the shape of the letter
J, while strongly negative scalars look like a reverse J. By contrast, weakly posi-
tive and negative scalars have a hump-shape, with the maximum either below the
“Potts&diagrams”
mean (weakly negative words like disappointing) or above the mean (weakly pos-
itive words like good). These shapes offer an illuminating typology of affective
IMDB
Potts,&Christopher.& 2011.&NSF&wor
restructuring& adjectives. Ca
meaning. Cat^
-0.50
-0.39
-0.28
1 2 3 4 5 6 7 8 9 10 1 2 3 4
1 2 3 4 5 6 7 8 9 10 rating
rating 1 2 3 4 5 6 7 8 9 10
rating
absolutely f
great bad
IMDB
1 2 3 4 5 6 7 8 9 10 1 2 3 4Ca
rating Cat
1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
rating rating
utterly p
excellent terrible 0.13
0.09
0.05
1 2 3 4 5 6 7 8 9 10 1 2 3 4
-0.50
-0.39
-0.28
rating
1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
rating rating
Figure 22.10 Potts diagrams (Potts, 2011) for positive and negative scalar adjectives, show-
ing the J-shape and reverse J-shape for strongly positive and negative adjectives, and the
hump-shape for more weakly polarized adjectives.
Fig. 22.11 shows the Potts diagrams for emphasizing and attenuating adverbs.
Note that emphatics tend to have a J-shape (most likely to occur in the most posi-
tive reviews) or a U-shape (most likely to occur in the strongly positive and nega-
tive). Attenuators all have the hump-shape, emphasizing the middle of the scale and
downplaying both extremes. The diagrams can be used both as a typology of lexical
sentiment, and also play a role in modeling sentiment compositionality.
In addition to functions like posterior P(c|w), likelihood P(w|c), or normalized
likelihood (Eq. 22.6) many other functions of the count of a word occurring with a
sentiment label have been used. We’ll introduce some of these on page 512, includ-
ing ideas like normalizing the counts per writer in Eq. 22.14.
0.
0.
0.
0.
0.
-0.
-0.
0.
0.
0.
-0.
-0.
0.
Category Category Category
fairly/r
Cat^2 509
S ENTIMENTCat
Cat = -0.13 (p = 0.284)
(p < 0.001)
= 0.2 (p = 0.265)
= -4.16 (p = 0.007)
OpenTable – 2,829 tokens Goodreads – 1,806
Cat = -0.87 (p
Cat^2 = -5.74 (p
0.35
0.31
-0.50
-0.39
-0.28
-0.17
-0.06
0.06
0.17
0.28
0.39
0.50
-0.50
-0.25
0.00
0.25
0.50
-0.50
-0.25
0.00
Category Category Category
1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
5 6 7 8 9 10 rating
rating 1 2 3 4 5 6 7 8 9 10 rating
rating
absolutely fairly
great bad pretty/r
IMDB – 176,264 tokens OpenTable – 8,982 tokens Goodreads – 11,89
1 2 3 4 5 6 7 8 9 10 1 2 3 4Cat5= -0.43
6 7(p 8< 0.001)
9 10 Cat = -0.64 (p = 0.035) Cat = -0.71 (p
rating Cat^2 = -3.6 (p < 0.001)
rating Cat^2 = -4.47 (p = 0.007) Cat^2 = -4.59 (p
5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
0.34
rating rating 0.32
utterly pretty
xcellent terrible 0.13
0.19
0.15
0.14
0.09
0.05 0.08 0.07
1 2 3 4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
-0.50
-0.39
-0.28
-0.17
-0.06
0.06
0.17
0.28
0.39
0.50
-0.50
-0.25
0.00
0.25
0.50
-0.50
-0.25
0.00
rating rating
Category Category Category
4 5 6 7 8 9 10 1 2 3 4 5 6 7 8 9 10
rating rating Figure 22.11 Potts diagrams (Potts, 2011) for emphatic and attenuating adverbs.
j:
Pi (horrible) P j (horrible)
lor(horrible) = log − log
1 − Pi (horrible) 1 − P j (horrible)
i j
f (horrible) f (horrible)
= log ni
− log nj
i
f (horrible) j
f (horrible)
1− 1 −
ni nj
i j
f (horrible) f (horrible)
= log i i − log (22.8)
n − f (horrible) n j − f j (horrible)
The Dirichlet intuition is to use a large background corpus to get a prior estimate of
what we expect the frequency of each word w to be. We’ll do this very simply by
adding the counts from that corpus to the numerator and denominator, so that we’re
essentially shrinking the counts toward that prior. It’s like asking how large are the
differences between i and j given what we would expect given their frequencies in
a well-estimated large background corpus.
The method estimates the difference between the frequency of word w in two
(i− j)
corpora i and j via the prior-modified log odds ratio for w, δw , which is estimated
as:
!
(i− j) fwi + αw fwj + αw
δw = log i − log (22.9)
n + α0 − ( fwi + αw ) n j + α0 − ( fwj + αw )
(where ni is the size of corpus i, n j is the size of corpus j, fwi is the count of word
w in corpus i, fwj is the count of word w in corpus j, α0 is the scaled size of the
background corpus, and αw is the scaled count of word w in the background corpus.)
In addition, Monroe et al. (2008) make use of an estimate for the variance of the
log–odds–ratio:
1 1
(i− j)
σ 2 δ̂w ≈ i + j (22.10)
fw + αw fw + αw
The final statistic for a word is then the z–score of its log–odds–ratio:
(i− j)
δ̂w
r (22.11)
(i− j)
σ 2 δ̂w
The Monroe et al. (2008) method thus modifies the commonly used log odds ratio
in two ways: it uses the z-scores of the log odds ratio, which controls for the amount
of variance in a word’s frequency, and it uses counts from a background corpus to
provide a prior count for words.
Fig. 22.12 shows the method applied to a dataset of restaurant reviews from
Yelp, comparing the words used in 1-star reviews to the words used in 5-star reviews
(Jurafsky et al., 2014). The largest difference is in obvious sentiment words, with the
1-star reviews using negative sentiment words like worse, bad, awful and the 5-star
reviews using positive sentiment words like great, best, amazing. But there are other
illuminating differences. 1-star reviews use logical negation (no, not), while 5-star
reviews use emphatics and emphasize universality (very, highly, every, always). 1-
star reviews use first person plurals (we, us, our) while 5 star reviews use the second
person. 1-star reviews talk about people (manager, waiter, customer) while 5-star
reviews talk about dessert and properties of expensive restaurants like courses and
atmosphere. See Jurafsky et al. (2014) for more details.
22.6 • U SING L EXICONS FOR S ENTIMENT R ECOGNITION 511
If supervised training data is available, these counts computed from sentiment lex-
icons, sometimes weighted or normalized in various ways, can also be used as fea-
tures in a classifier along with other lexical or non-lexical features. We return to
such algorithms in Section 22.7.
512 C HAPTER 22 • L EXICONS FOR S ENTIMENT, A FFECT, AND C ONNOTATION
Various weights can be used for the features, including the raw count in the training
set, or some normalized probability or log probability. Schwartz et al. (2013), for
example, turn feature counts into phrase likelihoods by normalizing them by each
subject’s total word use.
freq(phrase, subject)
p(phrase|subject) = X (22.14)
freq(phrase0 , subject)
phrase0 ∈vocab(subject)
If the training data is sparser, or not as similar to the test set, any of the lexicons
we’ve discussed can play a helpful role, either alone or in combination with all the
words and n-grams.
Many possible values can be used for lexicon features. The simplest is just an
indicator function, in which the value of a feature fL takes the value 1 if a particular
text has any word from the relevant lexicon L. Using the notation of Appendix K, in
which a feature value is defined for a particular output class c and document x.
1 if ∃w : w ∈ L & w ∈ x & class = c
fL (c, x) =
0 otherwise
Alternatively the value of a feature fL for a particular lexicon L can be the total
number of word tokens in the document that occur in L:
X
fL = count(w)
w∈L
For lexica in which each word is associated with a score or weight, the count can be
multiplied by a weight θwL :
X
fL = θwL count(w)
w∈L
22.8 • L EXICON - BASED METHODS FOR E NTITY-C ENTRIC A FFECT 513
weakly Rachel Dent Gordan Batman Joker powerfully weakly Rachel Joker Dent Gordan Batm
negative Joker Dent Gordan Rachel Batman positive negative Joker Gordan Batman Dent Rach
Agency Score
Agency Score
Figure 22.13 Power (dominance), sentiment (valence) and agency (arousal) for Figure 2: Power, sentiment, and agency sco
characters
in the movie TheFigure 1: Power,
Dark Knight sentiment,
computed and agency
from embeddings scores
trained onfor
thechar-
NRC VADacters
Lexicon.
in The Dark Night as learned throug
acters(Batman)
Note the protagonist in The Dark Night
and the as learned
antagonist through
(the Joker) thehigh
have regres-
power and agency
ELMo embeddings. These scores reflect th
sion model with ELMo embeddings. Scores generally
scores but differ in sentiment, while the love interest Rachel has low power and agency but
terns as the regression model with greater
high sentiment. align with character archetypes, i.e. the antagonist has
between characters.
the lowest sentiment score.
(22.16) the teenager ... survived the Boston Marathon bombing”
By using the verb
ment violate
haveinresulted
(22.15), in
thehis
author is expressing
effective removaltheir vey with
sympathies
from Dent (ally to Batman who turns
Country B, portraying Country B as a victim, and expressing antagonism toward
Rachel Dawes (primary love interest).
the industry. While articles about the #MeToo
the agent Country A. By contrast, in using the verb survive, the author of (22.16) is
itate extracting example sentences, we
expressing thatmovement
the bombingportray men experience,
is a negative like Weinstein assubject
and the unpow- of the sentence,
the teenager, iserful, we can character.
a sympathetic speculate These
that the corpora
aspects used to areinstance
of connotation inherent
of these entities in the narrative
in the meaningtrain
of theELMo and BERT
verbs violate portrayasthem
and survive, shownasinpowerful.
Fig. 22.14. and average across instances to obtain
Thus, in a corpus where traditional power roles score for the document.9 To maximiz
haveRole2”
Connotation Frame for “Role1 survives been inverted, the Connotation embeddingsFrame for extracted by capturing every mention of an entit
“Role1 violates Role2”
from ELMo and BERT perform worse than ran- form co-reference resolution by hand.
)
S(
le1
)
S(
le1
Writer
+
ite
dom, struc-
r→
Writer
+
ite
r→
r→
r→
ite
ite
S(role1→role2) tures in the data they are trained on. Further ev-
le2
le2
S(
S(role1→role2)
S(
)
)
_
beddings generally capture power _ poorly as com- +
Figures 1 and 2 show results.
+ ence, we show the entity scores as co
Reader pared to the unmasked embeddings (Table 2), Reader
they outperform the unmasked embeddings one polar opposite pair identified by
(a) (b) on this
task, and even outperform the regression model and ASP show s
Figure 22.14 Connotation frames for survive and violate. (a) For the frequency
survive, the writerbaseline
and reader have positive
sentiment toward Role1, the subject, in and
onenegative
setting. Nevertheless,
sentiment toward Role2, theythedo notobject.
direct outper- terns. the
(b) For violate, Batman has high power, while R
writer and reader have positive sentiment form Fieldinsteadettoward
al. (2019),
Role2, likely because
the direct object. they do not low power. Additionally, the Joker is
capture affect information as well as the unmasked with the most negative sentiment, but
The connotation frame lexicons
embeddings (Table 2). of Rashkin et al. (2016) and Rashkinest etagency.
al. Throughout the plot sum
(2017) also express other connotative aspects of the predicate toward each argu-
movie progresses by the Joker taking
ment, including the effect (something bad happened to x) value: (x is valuable), and
4.3 Qualitative Document-level Analysis sive action and the other characters r
mental state: (x is distressed by the event). Connotation frames can also mark the
We can see this dynamic reflected in t
power differential Finally,
betweenwe the qualitatively
arguments (using analyzethe how well our
verb implore means that the
theme argument has greater poweraffect
than thedimensions
agent), and the profile score, as a high-powered, hi
method captures by agency
analyzing of each argument
(waited is low singleagency). Fig. 22.15 in shows a visualization from Sap low-sentiment character, who is the pri
et al. (2017).
documents detail. We conduct this anal-
Connotation frames can be built by hand (Sap et al., 2017), or they can be driver.
learnedIn general, ASP shows a greater
ysis in a domain where we expect entities to fulfill
by supervised learning (Rashkin et al., 2016), for example using hand-labeled train- characters than the regression m
between
traditional power roles and where entity portray-
hypothesize that this occurs because AS
als are known. Following Bamman et al. (2013),
the dimensions of interest, while the reg
we analyze the Wikipedia plot summary of the
proach captures other confounds, such
movie The Dark Knight,7 focusing on Batman
(protagonist),8 the Joker (antagonist), Jim Gordan 9
When we used this averaging metric in othe
22.10 • S UMMARY 515
VERB
AGENT THEME
implore
Figure 22.15 The connotation frames of Sap et al. (2017), showing that the verb implore
implies the agent has Figure 2: The
lower power formal
than notation
the theme of the connotation
(in contrast, say, with a verb like demanded),
and showing the low frames
level of of power
agency of and [Link] The
the subject firstFigure
waited. example
from Sap et al. (2017).
shows the relative power differential implied by
the verb “implored”, i.e., the agent (“he”) is in
ing data to supervise classifiers
a position forpower
of less each than
of thetheindividual
theme (“the relations,
tri- e.g., 3:
Figure whether
Sample verbs in the connotation frame
S(writer → Role1)bunal”).
is + or In-, contrast,
and then“He improving
demanded accuracy withconstraints
via global
the tribunal high annotator agreement. Size is indicativ
across all [Link] mercy” implies that the agent has authority of verb frequency in our corpus (bigger = mor
over the theme. The second example shows the frequent), color differences are only for legibility
low level of agency implied by the verb “waited”.
one another. For example, if the agent “dom
22.10 Summary interactive demo website of our findings (see Fig- inates” the theme (denoted as power(AG>TH)
ure 5 in the appendix for a screenshot).2 Further- then the agent is implied to have a level of contro
more, as will be seen in Section 4.1, connotation over the theme. Alternatively, if the agent “hon
• Many kinds of affective states can be distinguished,
frames offer new insights that complement and de- including emotions, moods,
ors” the theme (denoted as power(AG<TH)), th
attitudes (which include sentiment), interpersonal
viate from the well-known Bechdel test (Bechdel, stance, and personality.
writer implies that the theme is more important o
authoritative. We used AMT crowdsourcing to la
• Emotion can1986). In particular,
be represented we find
by fixed that units
atomic high-agency
often called basic emo-
women through the lens of connotation frames are bel 1700 transitive verbs for power differential
tions, or as points in space defined by dimensions like valenceWith and arousal.
three annotators per verb, the inter-annotato
rare in modern films. It is, in part, because some
connotational
• Words have movies (e.g., Snow aspects related
White) to thesepass
accidentally affective agreement
the states, andisthis
0.34 (Krippendorff’s ↵).
connotationalBechdel
aspect test
of word meaning
and also caneven
because be represented
movies within lexicons.
Agency The agency attributed to the agent of th
strong female characters are not entirely free from verbtodenotes whether the action being describe
• Affective lexicons can be built by hand, using crowd sourcing label the
the deeply ingrained biases in social norms. implies that the agent is powerful, decisive, an
affective content of each word.
capable of pushing forward their own storyline
• Lexicons can2 beConnotation Frames of Power
built with semi-supervised, and
bootstrapping from
For seed words
example, a person who is described as “ex
Agencylike embedding cosine.
using similarity metrics periencing” things does not seem as active and de
• Lexicons canWebecreate twoinnew
learned connotation
a fully relations,
supervised powerwhencisive
manner, as someone who is described as “determin
a convenient
and agency (examples in Figure 3), as an expan- ing” things. AMT workers labeled 2000 trans
training signal can be found in the world, such as ratings assigned by users on
3 tive verbs for implying high/moderate/low agenc
a review [Link] of the existing connotation frame lexicons.
Three AMT crowdworkers annotated the verbs (inter-annotator agreement of 0.27). We denot
• Words can bewith
assigned weightstoinavoid
placeholders a lexicon
genderbybias
using various
in the con- functions of word
high agency as agency(AG)=+, and low agenc
counts in training texts, and ratio metrics like log
text (e.g., X rescued Y; an example task is shownodds ratio
as informative
agency( AG )= .
Dirichlet prior.
in the appendix in Figure 7). We define the anno- Pairwise agreements on a hard constraint ar
tated constructs as follows: 56% and 51%
• Affect can be detected, just like sentiment, by using standard supervised text for power and agency, respec
classificationPower
techniques, using allMany
Differentials the words
verbs or bigrams
imply tively.
in a text
the au- Despite this, agreements reach 96% an
as features.
Additional features can beofdrawn
thority levels fromand
the agent counts of words
theme 94% when moderate labels are counted as agree
relativeintolexicons.
ing with either high or low labels, showing that an
• Lexicons can also
2
be used to detect affect in a rule-based
[Link] ˜msap/ classifier by picking
notators rarely strongly disagree with one anothe
movie-bias/.
the simple majority
3 sentiment based on counts of words in each lexicon.
Some contributing factors in the lower KA score
The lexicons and a demo are available at http://
• [Link]/ ˜msap/movie-bias/.
frames express richer relations include
of affective meaning that the subtlety of choosing between neutra
a pred-
icate encodes about its arguments.
516 C HAPTER 22 • L EXICONS FOR S ENTIMENT, A FFECT, AND C ONNOTATION
Historical Notes
The idea of formally representing the subjective meaning of words began with Os-
good et al. (1957), the same pioneering study that first proposed the vector space
model of meaning described in Chapter 5. Osgood et al. (1957) had participants rate
words on various scales, and ran factor analysis on the ratings. The most significant
factor they uncovered was the evaluative dimension, which distinguished between
pairs like good/bad, valuable/worthless, pleasant/unpleasant. This work influenced
the development of early dictionaries of sentiment and affective meaning in the field
of content analysis (Stone et al., 1966).
subjectivity Wiebe (1994) began an influential line of work on detecting subjectivity in text,
beginning with the task of identifying subjective sentences and the subjective char-
acters who are described in the text as holding private states, beliefs or attitudes.
Learned sentiment lexicons such as the polarity lexicons of Hatzivassiloglou and
McKeown (1997) were shown to be a useful feature in subjectivity detection (Hatzi-
vassiloglou and Wiebe 2000, Wiebe 2000).
The term sentiment seems to have been introduced in 2001 by Das and Chen
(2001), to describe the task of measuring market sentiment by looking at the words in
stock trading message boards. In the same paper Das and Chen (2001) also proposed
the use of a sentiment lexicon. The list of words in the lexicon was created by
hand, but each word was assigned weights according to how much it discriminated
a particular class (say buy versus sell) by maximizing across-class variation and
minimizing within-class variation. The term sentiment, and the use of lexicons,
caught on quite quickly (e.g., inter alia, Turney 2002). Pang et al. (2002) first showed
the power of using all the words without a sentiment lexicon; see also Wang and
Manning (2012).
Most of the semi-supervised methods we describe for extending sentiment dic-
tionaries drew on the early idea that synonyms and antonyms tend to co-occur in the
same sentence (Miller and Charles 1991, Justeson and Katz 1991, Riloff and Shep-
herd 1997). Other semi-supervised methods for learning cues to affective mean-
ing rely on information extraction techniques, like the AutoSlog pattern extractors
(Riloff and Wiebe, 2003). Graph based algorithms for sentiment were first sug-
gested by Hatzivassiloglou and McKeown (1997), and graph propagation became a
standard method (Zhu and Ghahramani 2002, Zhu et al. 2003, Zhou et al. 2004a,
Velikovich et al. 2010). Crowdsourcing can also be used to improve precision by
filtering the result of semi-supervised lexicon learning (Riloff and Shepherd 1997,
Fast et al. 2016).
Much recent work focuses on ways to learn embeddings that directly encode sen-
timent or other properties, such as the D ENSIFIER algorithm of Rothe et al. (2016)
that learns to transform the embedding space to focus on sentiment (or other) infor-
mation.
Exercises
22.1 Show that the relationship between a word w and a category c in the Potts
Score in Eq. 22.6 is a variant of the pointwise mutual information pmi(w, c)
without the log term.
CHAPTER
Discourse Model
Lotsabucks
V
Megabucks
$ pay refer (access)
refer (evoke)
“Victoria” corefer “she”
Figure 23.1 How mentions evoke and access discourse entities in a discourse model.
Reference in a text to an entity that has been previously introduced into the
anaphora discourse is called anaphora, and the referring expression used is said to be an
anaphor anaphor, or anaphoric.2 In passage (23.1), the pronouns she and her and the defi-
nite NP the 38-year-old are therefore anaphoric. The anaphor corefers with a prior
antecedent mention (in this case Victoria Chen) that is called the antecedent. Not every refer-
ring expression is an antecedent. An entity that has only a single mention in a text
singleton (like Lotsabucks in (23.1)) is called a singleton.
coreference In this chapter we focus on the task of coreference resolution. Coreference
resolution
resolution is the task of determining whether two mentions corefer, by which we
mean they refer to the same entity in the discourse model (the same discourse entity).
coreference The set of coreferring expressions is often called a coreference chain or a cluster.
chain
cluster For example, in processing (23.1), a coreference resolution algorithm would need
to find at least four coreference chains, corresponding to the four entities in the
discourse model in Fig. 23.1.
1. {Victoria Chen, her, the 38-year-old, She}
2. {Megabucks Banking, the company, Megabucks}
3. {her pay}
4. {Lotsabucks}
Note that mentions can be nested; for example the mention her is syntactically
part of another mention, her pay, referring to a completely different discourse entity.
Coreference resolution thus comprises two tasks (although they are often per-
formed jointly): (1) identifying the mentions, and (2) clustering them into corefer-
ence chains/discourse entities.
We said that two mentions corefered if they are associated with the same dis-
course entity. But often we’d like to go further, deciding which real world entity is
associated with this discourse entity. For example, the mention Washington might
refer to the US state, or the capital city, or the person George Washington; the inter-
pretation of the sentence will of course be very different for each of these. The task
entity linking of entity linking (Ji and Grishman, 2011) or entity resolution is the task of mapping
a discourse entity to some real-world individual.3 We usually operationalize entity
2 We will follow the common NLP usage of anaphor to mean any mention that has an antecedent, rather
than the more narrow usage to mean only mentions (like pronouns) whose interpretation depends on the
antecedent (under the narrower interpretation, repeated names are not anaphors).
3 Computational linguistics/NLP thus differs in its use of the term reference from the field of formal
semantics, which uses the words reference and coreference to describe the relation between a mention
and a real-world entity. By contrast, we follow the functional linguistics tradition in which a mention
refers to a discourse entity (Webber, 1978) and the relation between a discourse entity and the real world
individual requires an additional step of linking.
519
inferrables: these introduce entities that are neither hearer-old nor discourse-old,
but the hearer can infer their existence by reasoning based on other entities
that are in the discourse. Consider the following examples:
(23.18) I went to a superb restaurant yesterday. The chef had just opened it.
(23.19) Mix flour, butter and water. Knead the dough until shiny.
Neither the chef nor the dough were in the discourse model based on the first
bridging sentence of either example, but the reader can make a bridging inference
inference
that these entities should be added to the discourse model and associated with
the restaurant and the ingredients, based on world knowledge that restaurants
have chefs and dough is the result of mixing flour and liquid (Haviland and
Clark 1974, Webber and Baldwin 1992, Nissim et al. 2004, Hou et al. 2018).
The form of an NP gives strong clues to its information status. We often talk
given-new about an entity’s position on the given-new dimension, the extent to which the refer-
ent is given (salient in the discourse, easier for the hearer to call to mind, predictable
by the hearer), versus new (non-salient in the discourse, unpredictable) (Chafe 1976,
accessible Prince 1981, Gundel et al. 1993). A referent that is very accessible (Ariel, 2001)
i.e., very salient in the hearer’s mind or easy to call to mind, can be referred to with
less linguistic material. For example pronouns are used only when the referent has
salience a high degree of activation or salience in the discourse model.4 By contrast, less
salient entities, like a new referent being introduced to the discourse, will need to be
introduced with a longer and more explicit referring expression to help the hearer
recover the referent.
Thus when an entity is first introduced into a discourse its mentions are likely
to have full names, titles or roles, or appositive or restrictive relative clauses, as in
the introduction of our protagonist in (23.1): Victoria Chen, CFO of Megabucks
Banking. As an entity is discussed over a discourse, it becomes more salient to the
hearer and its mentions on average typically becomes shorter and less informative,
for example with a shortened name (for example Ms. Chen), a definite description
(the 38-year-old), or a pronoun (she or her) (Hawkins 1978). However, this change
in length is not monotonic, and is sensitive to discourse structure (Grosz 1977b,
Reichman 1985, Fox 1993).
singular they Second, singular they has become much more common, in which they is used to
describe singular individuals, often useful because they is gender neutral. Although
recently increasing, singular they is quite old, part of English for many centuries.5
Person Agreement: English distinguishes between first, second, and third person,
and a pronoun’s antecedent must agree with the pronoun in person. Thus a third
person pronoun (he, she, they, him, her, them, his, her, their) must have a third person
antecedent (one of the above or any other noun phrase). However, phenomena like
quotation can cause exceptions; in this example I, my, and she are coreferent:
(23.32) “I voted for Nader because he was most aligned with my values,” she said.
Gender or Noun Class Agreement: In many languages, all nouns have grammat-
ical gender or noun class6 and pronouns generally agree with the grammatical gender
of their antecedent. In English this occurs only with third-person singular pronouns,
which distinguish between male (he, him, his), female (she, her), and nonpersonal
(it) grammatical genders. Non-binary pronouns like ze or hir may also occur in more
recent texts. Knowing which gender to associate with a name in text can be complex,
and may require world knowledge about the individual. Some examples:
(23.33) Maryam has a theorem. She is exciting. (she=Maryam, not the theorem)
(23.34) Maryam has a theorem. It is exciting. (it=the theorem, not Maryam)
Binding Theory Constraints: The binding theory is a name for syntactic con-
straints on the relations between a mention and an antecedent in the same sentence
reflexive (Chomsky, 1981). Oversimplifying a bit, reflexive pronouns like himself and her-
self corefer with the subject of the most immediate clause that contains them (23.35),
whereas nonreflexives cannot corefer with this subject (23.36).
(23.35) Janet bought herself a bottle of fish sauce. [herself=Janet]
(23.36) Janet bought her a bottle of fish sauce. [her6=Janet]
Grammatical Role: Entities mentioned in subject position are more salient than
those in object position, which are in turn more salient than those mentioned in
oblique positions. Thus although the first sentence in (23.38) and (23.39) expresses
roughly the same propositional content, the preferred referent for the pronoun he
varies with the subject—Billy Bones in (23.38) and Jim Hawkins in (23.39).
(23.38) Billy Bones went to the bar with Jim Hawkins. He called for a glass of
rum. [ he = Billy ]
(23.39) Jim Hawkins went to the bar with Billy Bones. He called for a glass of
rum. [ he = Jim ]
5 Here’s a bound pronoun example from Shakespeare’s Comedy of Errors: There’s not a man I meet but
doth salute me As if I were their well-acquainted friend
6 The word “gender” is generally only used for languages with 2 or 3 noun classes, like most Indo-
European languages; many languages, like the Bantu languages or Chinese, have a much larger number
of noun classes.
23.2 • C OREFERENCE TASKS AND DATASETS 525
Verb Semantics: Some verbs semantically emphasize one of their arguments, bi-
asing the interpretation of subsequent pronouns. Compare (23.40) and (23.41).
(23.40) John telephoned Bill. He lost the laptop.
(23.41) John criticized Bill. He lost the laptop.
These examples differ only in the verb used in the first sentence, yet “he” in (23.40)
is typically resolved to John, whereas “he” in (23.41) is resolved to Bill. This may
be partly due to the link between implicit causality and saliency: the implicit cause
of a “criticizing” event is its object, whereas the implicit cause of a “telephoning”
event is its subject. In such verbs, the entity which is the implicit cause may be more
salient.
Selectional Restrictions: Many other kinds of semantic knowledge can play a role
in referent preference. For example, the selectional restrictions that a verb places on
its arguments (Chapter 21) can help eliminate referents, as in (23.42).
(23.42) I ate the soup in my new bowl after cooking it for hours
There are two possible referents for it, the soup and the bowl. The verb eat, however,
requires that its direct object denote something edible, and this constraint can rule
out bowl as a possible referent.
Exactly what counts as a mention and what links are annotated differs from task
to task and dataset to dataset. For example some coreference datasets do not label
singletons, making the task much simpler. Resolvers can achieve much higher scores
on corpora without singletons, since singletons constitute the majority of mentions in
running text, and they are often hard to distinguish from non-referential NPs. Some
tasks use gold mention-detection (i.e. the system is given human-labeled mention
boundaries and the task is just to cluster these gold mentions), which eliminates the
need to detect and segment mentions from running text.
Coreference is usually evaluated by the CoNLL F1 score, which combines three
metrics: MUC, B3 , and CEAFe ; Section 23.8 gives the details.
Let’s mention a few characteristics of one popular coreference dataset, OntoNotes
(Pradhan et al. 2007c, Pradhan et al. 2007a), and the CoNLL 2012 Shared Task
based on it (Pradhan et al., 2012a). OntoNotes contains hand-annotated Chinese
and English coreference datasets of roughly one million words each, consisting of
newswire, magazine articles, broadcast news, broadcast conversations, web data and
conversational speech data, as well as about 300,000 words of annotated Arabic
newswire. The most important distinguishing characteristic of OntoNotes is that
it does not label singletons, simplifying the coreference task, since singletons rep-
resent 60%-70% of all entities. In other ways, it is similar to other coreference
datasets. Referring expression NPs that are coreferent are marked as mentions, but
generics and pleonastic pronouns are not marked. Appositive clauses are not marked
as separate mentions, but they are included in the mention. Thus in the NP, “Richard
Godown, president of the Industrial Biotechnology Association” the mention is the
entire phrase. Prenominal modifiers are annotated as separate entities only if they
are proper nouns. Thus wheat is not an entity in wheat fields, but UN is an entity in
UN policy (but not adjectives like American in American policy).
A number of corpora mark richer discourse phenomena. The ISNotes corpus
annotates a portion of OntoNotes for information status, include bridging examples
(Hou et al., 2018). The LitBank coreference corpus (Bamman et al., 2020) contains
coreference annotations for 210,532 tokens from 100 different literary novels, in-
cluding singletons and quantified and negated noun phrases. The AnCora-CO coref-
erence corpus (Recasens and Martı́, 2010) contains 400,000 words each of Spanish
(AnCora-CO-Es) and Catalan (AnCora-CO-Ca) news data, and includes labels for
complex phenomena like discourse deixis in both languages. The ARRAU corpus
(Uryupina et al., 2020) contains 350,000 words of English marking all NPs, which
means singleton clusters are available. ARRAU includes diverse genres like dialog
(the TRAINS data) and fiction (the Pear Stories), and has labels for bridging refer-
ences, discourse deixis, generics, and ambiguous anaphoric relations.
It is Modaladjective that S
It is Modaladjective (for NP) to VP
It is Cogv-ed that S
It seems/appears/means/follows (that) S
as make it in advance, but make them in Hollywood did not occur at all). These
n-gram contexts can be used as features in a supervised anaphoricity classifier.
p(coref|”Victoria Chen”,”she”)
Victoria Chen Megabucks Banking her her pay the 37-year-old she
p(coref|”Megabucks Banking”,”she”)
Figure 23.2 For each pair of a mention (like she), and a potential antecedent mention (like
Victoria Chen or her), the mention-pair classifier assigns a probability of a coreference link.
For each prior mention (Victoria Chen, Megabucks Banking, her, etc.), the binary
classifier computes a probability: whether or not the mention is the antecedent of
she. We want this probability to be high for actual antecedents (Victoria Chen, her,
the 38-year-old) and low for non-antecedents (Megabucks Banking, her pay).
Early classifiers used hand-built features (Section 23.5); more recent classifiers
use neural representation learning (Section 23.6)
For training, we need a heuristic for selecting training samples; since most pairs
of mentions in a document are not coreferent, selecting every pair would lead to
a massive overabundance of negative samples. The most common heuristic, from
(Soon et al., 2001), is to choose the closest antecedent as a positive example, and all
pairs in between as the negative examples. More formally, for each anaphor mention
mi we create
• one positive instance (mi , m j ) where m j is the closest antecedent to mi , and
530 C HAPTER 23 • C OREFERENCE R ESOLUTION AND E NTITY L INKING
} One or more
of these
should be high
}
ϵ Victoria Chen Megabucks Banking her her pay the 37-year-old she
All of these
should be low
p(ϵ|”she”) p(”Megabucks Banking”|she”) p(”her pay”|she”)
Figure 23.3 For each candidate anaphoric mention (like she), the mention-ranking system assigns a proba-
bility distribution over all previous mentions plus the special dummy mention .
At test time, for a given mention i the model computes one softmax over all the
antecedents (plus ) giving a probability for each candidate antecedent (or none).
23.5 • C LASSIFIERS USING HAND - BUILT FEATURES 531
Fig. 23.3 shows an example of the computation for the single candidate anaphor
she.
Once the antecedent is classified for each anaphor, transitive closure can be run
over the pairwise decisions to get a complete clustering.
Training is trickier in the mention-ranking model than the mention-pair model,
because for each anaphor we don’t know which of all the possible gold antecedents
to use for training. Instead, the best antecedent for each mention is latent; that
is, for each mention we have a whole cluster of legal gold antecedents to choose
from. Early work used heuristics to choose an antecedent, for example choosing the
closest antecedent as the gold antecedent and all non-antecedents in a window of
two sentences as the negative examples (Denis and Baldridge, 2008). Various kinds
of ways to model latent antecedents exist (Fernandes et al. 2012, Chang et al. 2013,
Durrett and Klein 2013). The simplest way is to give credit to any legal antecedent
by summing over all of them, with a loss function that optimizes the likelihood of
all correct antecedents from the gold clustering (Lee et al., 2017b). We’ll see the
details in Section 23.6.
Mention-ranking models can be implemented with hand-build features or with
neural representation learning (which might also incorporate some hand-built fea-
tures). we’ll explore both directions in Section 23.5 and Section 23.6.
form as well as neural ones. Nonetheless, they are still sometimes useful to build
lightweight systems when compute or data are sparse, and the features themselves
are useful for error analysis even in neural systems.
Given an anaphor mention and a potential antecedent mention, feature based
classifiers make use of three types of features: (i) features of the anaphor, (ii) features
of the candidate antecedent, and (iii) features of the relationship between the pair.
Entity-based models can make additional use of two additional classes: (iv) feature
of all mentions from the antecedent’s entity cluster, and (v) features of the relation
between the anaphor and the mentions in the antecedent entity cluster.
Figure 23.4 shows a selection of commonly used features, and shows the value
that would be computed for the potential anaphor “she” and potential antecedent
“Victoria Chen” in our example sentence, repeated below:
(23.47) Victoria Chen, CFO of Megabucks Banking, saw her pay jump to $2.3
million, as the 38-year-old also became the company’s president. It is
widely known that she came to Megabucks from rival Lotsabucks.
Features that prior work has found to be particularly useful are exact string
match, entity headword agreement, mention distance, as well as (for pronouns) exact
attribute match and i-within-i, and (for nominals and proper names) word inclusion
and cosine. For lexical features (like head words) it is common to only use words
that appear enough times (>20 times).
23.6 • A NEURAL MENTION - RANKING ALGORITHM 533
This score s(i, j) includes three factors that we’ll define below: m(i); whether span
i is a mention; m( j); whether span j is a mention; and c(i, j); whether j is the
antecedent of i:
For the dummy antecedent , the score s(i, ) is fixed to 0. This way if any non-
dummy scores are positive, the model predicts the highest-scoring antecedent, but if
all the scores are negative it abstains.
embedding for the first (start) token of the span, the encoder output for the last (end)
token of the span, and a third vector which is an attention-based representation:
The goal of the attention vector is to represent which word/token is the likely
syntactic head-word of the span; we saw in the prior section that head-words are
a useful feature; a matching head-word is a good indicator of coreference. The
attention representation is computed as usual; the system learns a weight vector wα ,
and computes its dot product with the hidden state ht transformed by a FFN:
And then the attention distribution is used to create a vector hATT(i) which is an
attention-weighted sum of the embeddings et of each of the words in span i:
END
X(i)
hATT(i) = ai,t · et (23.53)
t= START (i)
Fig. 23.5 shows the computation of the span representation and the mention
score.
General Electric Electric said the the Postal Service Service contacted the the company
Mention score (m)
Encodings (h)
Encoder
Figure 23.5 Computation of the span representation g (and the mention score m) in a BERT version of the
e2e-coref model (Lee et al. 2017b, Joshi et al. 2019). The model considers all spans up to a maximum width of
say 10; the figure shows a small subset of the bigram and trigram spans.
At inference time, this mention score m is used as a filter to keep only the best few
mentions.
23.6 • A NEURAL MENTION - RANKING ALGORITHM 535
We then compute the antecedent score for high-scoring mentions. The antecedent
score c(i, j) takes as input a representation of the spans i and j, but also the element-
wise similarity of the two spans to each other gi ◦ g j (here ◦ is element-wise mul-
tiplication). Fig. 23.6 shows the computation of the score s for the three possible
antecedents of the company in the example sentence from Fig. 23.5.
Figure 23.6 The computation of the score s for the three possible antecedents of the com-
pany in the example sentence from Fig. 23.5. Figure after Lee et al. (2017b).
Given the set of mentions, the joint distribution of antecedents for each docu-
ment is computed in a forward pass, and we can then do transitive closure on the
antecedents to create a final clustering for the document.
Fig. 23.7 shows example predictions from the model, showing the attention
weights, which Lee et al. (2017b) find correlate with traditional semantic heads.
Note that the model gets the second example wrong, presumably because attendants
and pilot likely have nearby word embeddings.
Figure 23.7 Sample predictions from the Lee et al. (2017b) model, with one cluster per
example, showing one correct example and one mistake. Bold, parenthesized spans are men-
tions in the predicted cluster. The amount of red color on a word indicates the head-finding
attention weight ai,t in Eq. 23.52. Figure adapted from Lee et al. (2017b).
23.6.3 Learning
For training, we don’t have a single gold antecedent for each mention; instead the
coreference labeling only gives us each entire cluster of coreferent mentions; so a
mention only has a latent antecedent. We therefore use a loss function that maxi-
mizes the sum of the coreference probability of any of the legal antecedents. For a
given mention i with possible antecedents Y (i), let GOLD(i) be the set of mentions
in the gold cluster containing i. Since the set of mentions occurring before i is Y (i),
the set of mentions in that gold cluster that also occur before i is Y (i) ∩ GOLD(i). We
536 C HAPTER 23 • C OREFERENCE R ESOLUTION AND E NTITY L INKING
where in(x) is the set of Wikipedia pages pointing to x and W is the set of all Wiki-
pedia pages in the collection.
The vote given by anchor b to the candidate annotation a → X is the average,
over all the possible entities of b, of their relatedness to X, weighted by their prior
probability:
1 X
vote(b, X) = rel(X,Y )p(Y |b) (23.60)
|E(b)|
Y ∈E(b)
The total relatedness score for a → X is the sum of the votes of all the other anchors
detected in q:
X
relatedness(a → X) = vote(b, X) (23.61)
b∈Xq \a
1 X
coherence(a → X) = rel(B, X)
|S| − 1
B∈S\X
coherence(a → X) + linkprob(a)
score(a → X) = (23.62)
2
Finally, pairs are pruned if score(a → X) < λ , where the threshold λ is set on a
held-out set.
It then computes the likelihood of each span [i, j] in q being an entity mention, in
a way similar to the span-based algorithm we saw for the reader above. First we
compute the score for i/ j being the start/end of a mention:
Figure 23.8 A sketch of the inference process in the ELQ algorithm for entity linking in
questions (Li et al., 2020). Each candidate question mention span and candidate entity are
separately encoded, and then scored by the entity/span dot product.
where wstart and wend are vectors learned during training. Next, another trainable
embedding, wmention is used to compute a score for each token being part of a men-
tion:
Mention spans can be linked to entities by computing, for each entity e and span
[i, j], the dot product similarity between the span encoding (the average of the token
embeddings) and the entity encoding.
X j
1
yi, j = qt
( j − i + 1)
t=i
s(e, [i, j]) = x·e yi, j (23.68)
Finally, we take a softmax to get a distribution over entities for each span:
Training The ELQ mention detection and entity linking algorithm is fully super-
vised. This means, unlike the anchor dictionary algorithms from Section 23.7.1,
540 C HAPTER 23 • C OREFERENCE R ESOLUTION AND E NTITY L INKING
it requires datasets with entity boundaries marked and linked. Two such labeled
datasets are WebQuestionsSP (Yih et al., 2016), an extension of the WebQuestions
(Berant et al., 2013) dataset derived from Google search questions, and GraphQues-
tions (Su et al., 2016). Both have had entity spans in the questions marked and
linked (Sorokin and Gurevych 2018, Li et al. 2020) resulting in entity-labeled ver-
sions WebQSPEL and GraphQEL (Li et al., 2020).
Given a training set, the ELQ mention detection and entity linking phases are
trained jointly, optimizing the sum of their losses. The mention detection loss is
a binary cross-entropy loss, with L the length of the passage and N the number of
candidates:
1 X
LMD = − y[i, j] log p([i, j]) + (1 − y[i, j] ) log(1 − p([i, j])) (23.70)
N
1≤i≤ j≤min(i+L−1,n)
with y[i, j] = 1 if [i, j] is a gold mention span, else 0. The entity linking loss is:
LED = −logp(eg |[i, j]) (23.71)
where eg is the gold entity for mention [i, j].
The weight wi for each entity can be set to different values to produce different
versions of the algorithm.
Following a proposal from Denis and Baldridge (2009), the CoNLL coreference
competitions were scored based on the average of MUC, CEAF-e, and B3 (Pradhan
et al. 2011, Pradhan et al. 2012b), and so it is common in many evaluation campaigns
to report an average of these 3 metrics. See Luo and Pradhan (2016) for a detailed
description of the entire set of metrics; reference implementations of these should
be used rather than attempting to reimplement from scratch (Pradhan et al., 2014).
Alternative metrics have been proposed that deal with particular coreference do-
mains or tasks. For example, consider the task of resolving mentions to named
entities (persons, organizations, geopolitical entities), which might be useful for in-
formation extraction or knowledge base completion. A hypothesis chain that cor-
rectly contains all the pronouns referring to an entity, but has no version of the name
itself, or is linked with a wrong name, is not useful for this task. We might instead
want a metric that weights each mention by how informative it is (with names being
most informative) (Chen and Ng, 2013) or a metric that considers a hypothesis to
match a gold chain only if it contains at least one variant of a name (the NEC F1
metric of Agarwal et al. (2019)).
(23.74) The trophy didn’t fit into the suitcase because it was too small.
Question: What was too small? Answer: The suitcase
The problems have the following characteristics:
1. The problems each have two parties
2. A pronoun preferentially refers to one of the parties, but could grammatically
also refer to the other
3. A question asks which party the pronoun refers to
4. If one word in the question is changed, the human-preferred answer changes
to the other party
The kind of world knowledge that might be needed to solve the problems can
vary. In the trophy/suitcase example, it is knowledge about the physical world; that
a bigger object cannot fit into a smaller object. In the original Winograd sentence,
it is stereotypes about social actors like politicians and protesters. In examples like
the following, it is knowledge about human actions like turn-taking or thanking.
(23.75) Bill passed the gameboy to John because his turn was [over/next]. Whose
turn was [over/next]? Answers: Bill/John
(23.76) Joan made sure to thank Susan for all the help she had [given/received].
Who had [given/received] help? Answers: Susan/Joan.
Although the Winograd Schema was designed to require common-sense rea-
soning, a large percentage of the original set of problems can be solved by pre-
trained language models, fine-tuned on Winograd Schema sentences (Kocijan et al.,
2019). Large pretrained language models encode an enormous amount of world or
common-sense knowledge! The current trend is therefore to propose new datasets
with increasingly difficult Winograd-like coreference resolution problems like K NOW R EF
(Emami et al., 2019), with examples like:
(23.77) Marcus is undoubtedly faster than Jarrett right now but in [his] prime the
gap wasn’t all that big.
In the end, it seems likely that some combination of language modeling and knowl-
edge will prove fruitful; indeed, it seems that knowledge-based models overfit less
to lexical idiosyncracies in Winograd Schema training sets (Trichelair et al., 2018),
(23.78) The secretary called the physiciani and told himi about a new patient
[pro-stereotypical]
(23.79) The secretary called the physiciani and told heri about a new patient
[anti-stereotypical]
Zhao et al. (2018a) consider a coreference system to be biased if it is more accu-
rate at linking pronouns consistent with gender stereotypical occupations (e.g., him
with physician in (23.78)) than linking pronouns inconsistent with gender-stereotypical
occupations (e.g., her with physician in (23.79)). They show that coreference sys-
tems of all architectures (rule-based, feature-based machine learned, and end-to-
end-neural) all show significant bias, performing on average 21 F1 points worse in
the anti-stereotypical cases.
One possible source of this bias is that female entities are significantly un-
derrepresented in the OntoNotes dataset, used to train most coreference systems.
Zhao et al. (2018a) propose a way to overcome this bias: they generate a second
gender-swapped dataset in which all male entities in OntoNotes are replaced with
female ones and vice versa, and retrain coreference systems on the combined orig-
inal and swapped OntoNotes data, also using debiased GloVE embeddings (Boluk-
basi et al., 2016). The resulting coreference systems no longer exhibit bias on the
WinoBias dataset, without significantly impacting OntoNotes coreference accuracy.
In a follow-up paper, Zhao et al. (2019) show that the same biases exist in ELMo
contextualized word vector representations and coref systems that use them. They
showed that retraining ELMo with data augmentation again reduces or removes bias
in coreference systems on WinoBias.
Webster et al. (2018) introduces another dataset, GAP, and the task of Gendered
Pronoun Resolution as a tool for developing improved coreference algorithms for
gendered pronouns. GAP is a gender-balanced labeled corpus of 4,454 sentences
with gendered ambiguous pronouns (by contrast, only 20% of the gendered pro-
nouns in the English OntoNotes training data are feminine). The examples were
created by drawing on naturally occurring sentences from Wikipedia pages to create
hard to resolve cases with two named entities of the same gender and an ambiguous
pronoun that may refer to either person (or neither), like the following:
(23.80) In May, Fujisawa joined Mari Motohashi’s rink as the team’s skip, moving
back from Karuizawa to Kitami where she had spent her junior days.
Webster et al. (2018) show that modern coreference algorithms perform signif-
icantly worse on resolving feminine pronouns than masculine pronouns in GAP.
Kurita et al. (2019) shows that a system based on BERT contextualized word repre-
sentations shows similar bias.
23.11 Summary
This chapter introduced the task of coreference resolution.
• This is the task of linking together mentions in text which corefer, i.e. refer
to the same discourse entity in the discourse model, resulting in a set of
coreference chains (also called clusters or entities).
• Mentions can be definite NPs or indefinite NPs, pronouns (including zero
pronouns) or names.
544 C HAPTER 23 • C OREFERENCE R ESOLUTION AND E NTITY L INKING
Historical Notes
Coreference has been part of natural language processing since the 1970s (Woods
et al. 1972, Winograd 1972). The discourse model and the entity-centric foundation
of coreference was formulated by Karttunen (1969) (at the 3rd COLING confer-
ence), playing a role also in linguistic semantics (Heim 1982, Kamp 1981). But
it was Bonnie Webber’s 1978 dissertation and following work (Webber 1983) that
explored the model’s computational aspects, providing fundamental insights into
how entities are represented in the discourse model and the ways in which they can
license subsequent reference. Many of the examples she provided continue to chal-
lenge theories of reference to this day.
Hobbs
algorithm The Hobbs algorithm9 is a tree-search algorithm that was the first in a long
series of syntax-based methods for identifying reference robustly in naturally occur-
ring text. The input to the Hobbs algorithm is a pronoun to be resolved, together
with a syntactic (constituency) parse of the sentences up to and including the cur-
rent sentence. The details of the algorithm depend on the grammar used, but can be
understood from a simplified version due to Kehler et al. (2004) that just searches
through the list of NPs in the current and prior sentences. This simplified Hobbs
algorithm searches NPs in the following order: “(i) in the current sentence from
right-to-left, starting with the first NP to the left of the pronoun, (ii) in the previous
sentence from left-to-right, (iii) in two sentences prior from left-to-right, and (iv) in
9 The simpler of two algorithms presented originally in Hobbs (1978).
H ISTORICAL N OTES 545
the current sentence from left-to-right, starting with the first noun group to the right
of the pronoun (for cataphora). The first noun group that agrees with the pronoun
with respect to number, gender, and person is chosen as the antecedent” (Kehler
et al., 2004).
Lappin and Leass (1994) was an influential entity-based system that used weights
to combine syntactic and other features, extended soon after by Kennedy and Bogu-
raev (1996) whose system avoids the need for full syntactic parses.
Approximately contemporaneously centering (Grosz et al., 1995) was applied
to pronominal anaphora resolution by Brennan et al. (1987), and a wide variety of
work followed focused on centering’s use in coreference (Kameyama 1986, Di Eu-
genio 1990, Walker et al. 1994, Di Eugenio 1996, Strube and Hahn 1996, Kehler
1997a, Tetreault 2001, Iida et al. 2003). Kehler and Rohde (2013) show how center-
ing can be integrated with coherence-driven theories of pronoun interpretation. See
Chapter 24 for the use of centering in measuring discourse coherence.
Coreference competitions as part of the US DARPA-sponsored MUC confer-
ences provided early labeled coreference datasets (the 1995 MUC-6 and 1998 MUC-
7 corpora), and set the tone for much later work, choosing to focus exclusively
on the simplest cases of identity coreference (ignoring difficult cases like bridging,
metonymy, and part-whole) and drawing the community toward supervised machine
learning and metrics like the MUC metric (Vilain et al., 1995). The later ACE eval-
uations produced labeled coreference corpora in English, Chinese, and Arabic that
were widely used for model training and evaluation.
This DARPA work influenced the community toward supervised learning begin-
ning in the mid-90s (Connolly et al. 1994, Aone and Bennett 1995, McCarthy and
Lehnert 1995). Soon et al. (2001) laid out a set of basic features, extended by Ng and
Cardie (2002b), and a series of machine learning models followed over the next 15
years. These often focused separately on pronominal anaphora resolution (Kehler
et al. 2004, Bergsma and Lin 2006), full NP coreference (Cardie and Wagstaff 1999,
Ng and Cardie 2002b, Ng 2005a) and definite NP reference (Poesio and Vieira 1998,
Vieira and Poesio 2000), as well as separate anaphoricity detection (Bean and Riloff
1999, Bean and Riloff 2004, Ng and Cardie 2002a, Ng 2004), or singleton detection
(de Marneffe et al., 2015).
The move from mention-pair to mention-ranking approaches was pioneered by
Yang et al. (2003) and Iida et al. (2003) who proposed pairwise ranking methods,
then extended by Denis and Baldridge (2008) who proposed to do ranking via a soft-
max over all prior mentions. The idea of doing mention detection, anaphoricity, and
coreference jointly in a single end-to-end model grew out of the early proposal of Ng
(2005b) to use a dummy antecedent for mention-ranking, allowing ‘non-referential’
to be a choice for coreference classifiers, Denis and Baldridge’s 2007 joint system
combining anaphoricity classifier probabilities with coreference probabilities, the
Denis and Baldridge (2008) ranking model, and the Rahman and Ng (2009) pro-
posal to train the two models jointly with a single objective.
Simple rule-based systems for coreference returned to prominence in the 2010s,
partly because of their ability to encode entity-based features in a high-precision way
(Zhou et al. 2004b, Haghighi and Klein 2009, Raghunathan et al. 2010, Lee et al.
2011, Lee et al. 2013, Hajishirzi et al. 2013) but in the end they suffered from an
inability to deal with the semantics necessary to correctly handle cases of common
noun coreference.
A return to supervised learning led to a number of advances in mention-ranking
models which were also extended into neural architectures, for example using re-
546 C HAPTER 23 • C OREFERENCE R ESOLUTION AND E NTITY L INKING
Exercises
CHAPTER
24 Discourse Coherence
And even in our wildest and most wandering reveries, nay in our very dreams,
we shall find, if we reflect, that the imagination ran not altogether at adven-
tures, but that there was still a connection upheld among the different ideas,
which succeeded each other. Were the loosest and freest conversation to be
transcribed, there would immediately be transcribed, there would immediately
be observed something which connected it in all its transitions.
David Hume, An enquiry concerning human understanding, 1748
Orson Welles’ movie Citizen Kane was groundbreaking in many ways, perhaps most
notably in its structure. The story of the life of fictional media magnate Charles
Foster Kane, the movie does not proceed in chronological order through Kane’s
life. Instead, the film begins with Kane’s death (famously murmuring “Rosebud”)
and is structured around flashbacks to his life inserted among scenes of a reporter
investigating his death. The novel idea that the structure of a movie does not have
to linearly follow the structure of the real timeline made apparent for 20th century
cinematography the infinite possibilities and impact of different kinds of coherent
narrative structures.
But coherent structure is not just a fact about movies or works of art. Like
movies, language does not normally consist of isolated, unrelated sentences, but
instead of collocated, structured, coherent groups of sentences. We refer to such
discourse a coherent structured group of sentences as a discourse, and we use the word co-
coherence herence to refer to the relationship between sentences that makes real discourses
different than just random assemblages of sentences. The chapter you are now read-
ing is an example of a discourse, as is a news article, a conversation, a thread on
social media, a Wikipedia page, and your favorite novel.
What makes a discourse coherent? If you created a text by taking random sen-
tences each from many different sources and pasted them together, would that be a
local coherent discourse? Almost certainly not. Real discourses exhibit both local coher-
global ence and global coherence. Let’s consider three ways in which real discourses are
locally coherent;
First, sentences or clauses in real discourses are related to nearby sentences in
systematic ways. Consider this example from Hobbs (1979):
(24.1) John took a train from Paris to Istanbul. He likes spinach.
This sequence is incoherent because it is unclear to a reader why the second
sentence follows the first; what does liking spinach have to do with train trips? In
fact, a reader might go to some effort to try to figure out how the discourse could be
coherent; perhaps there is a French spinach shortage? The very fact that hearers try
to identify such connections suggests that human discourse comprehension involves
the need to establish this kind of coherence.
By contrast, in the following coherent example:
(24.2) Jane took a train from Paris to Istanbul. She had to attend a conference.
548 C HAPTER 24 • D ISCOURSE C OHERENCE
the second sentence gives a REASON for Jane’s action in the first sentence. Struc-
tured relationships like REASON that hold between text units are called coherence
coherence relations, and coherent discourses are structured by many such coherence relations.
relations
Coherence relations are introduced in Section 24.1.
A second way a discourse can be locally coherent is by virtue of being “about”
someone or something. In a coherent discourse some entities are salient, and the
discourse focuses on them and doesn’t go back and forth between multiple entities.
This is called entity-based coherence. Consider the following incoherent passage,
in which the salient entity seems to wildly swing from John to Jenny to the piano
store to the living room, back to Jenny, then the piano again:
(24.3) John wanted to buy a piano for his living room.
Jenny also wanted to buy a piano.
He went to the piano store.
It was nearby.
The living room was on the second floor.
She didn’t find anything she liked.
The piano he bought was hard to get up to that floor.
Entity-based coherence models measure this kind of coherence by tracking salient
Centering
Theory entities across a discourse. For example Centering Theory (Grosz et al., 1995), the
most influential theory of entity-based coherence, keeps track of which entities in
the discourse model are salient at any point (salient entities are more likely to be
pronominalized or to appear in prominent syntactic positions like subject or object).
In Centering Theory, transitions between sentences that maintain the same salient
entity are considered more coherent than ones that repeatedly shift between entities.
entity grid The entity grid model of coherence (Barzilay and Lapata, 2008) is a commonly
used model that realizes some of the intuitions of the Centering Theory framework.
Entity-based coherence is introduced in Section 24.3.
topically Finally, discourses can be locally coherent by being topically coherent: nearby
coherent
sentences are generally about the same topic and use the same or similar vocab-
ulary to discuss these topics. Because topically coherent discourses draw from a
single semantic field or topic, they tend to exhibit the surface property known as
lexical cohesion lexical cohesion (Halliday and Hasan, 1976): the sharing of identical or semanti-
cally related words in nearby sentences. For example, the fact that the words house,
chimney, garret, closet, and window— all of which belong to the same semantic
field— appear in the two sentences in (24.4), or that they share the identical word
shingled, is a cue that the two are tied together as a discourse:
(24.4) Before winter I built a chimney, and shingled the sides of my house...
I have thus a tight shingled and plastered house... with a garret and a
closet, a large window on each side....
In addition to the local coherence between adjacent or nearby sentences, dis-
courses also exhibit global coherence. Many genres of text are associated with
particular conventional discourse structures. Academic articles might have sections
describing the Methodology or Results. Stories might follow conventional plotlines
or motifs. Persuasive essays have a particular claim they are trying to argue for,
and an essay might express this claim together with a structured set of premises that
support the argument and demolish potential counterarguments. We’ll introduce
versions of each of these kinds of global coherence.
Why do we care about the local or global coherence of a discourse? Since co-
herence is a property of a well-written text, coherence detection plays a part in any
24.1 • C OHERENCE R ELATIONS 549
task that requires measuring the quality of a text. For example coherence can help
in pedagogical tasks like essay grading or essay quality measurement that are trying
to grade how well-written a human essay is (Somasundaran et al. 2014, Feng et al.
2014, Lai and Tetreault 2018). Coherence can also help for summarization; knowing
the coherence relationship between sentences can help know how to select informa-
tion from them. Finally, detecting incoherent text may even play a role in mental
health tasks like measuring symptoms of schizophrenia or other kinds of disordered
language (Ditman and Kuperberg 2010, Elvevåg et al. 2007, Bedi et al. 2015, Iter
et al. 2018).
Elaboration: The satellite gives additional information or detail about the situation
presented in the nucleus.
(24.8) [NUC Dorothy was from Kansas.] [SAT She lived in the midst of the great
Kansas prairies.]
Evidence: The satellite gives additional information or detail about the situation
presented in the nucleus. The information is presented with the goal of convince the
reader to accept the information presented in the nucleus.
(24.9) [NUC Kevin must be here.] [SAT His car is parked outside.]
550 C HAPTER 24 • D ISCOURSE C OHERENCE
Attribution: The satellite gives the source of attribution for an instance of reported
speech in the nucleus.
(24.10) [SAT Analysts estimated] [NUC that sales at U.S. stores declined in the
quarter, too]
evidence
We can also talk about the coherence of a larger text by considering the hierar-
chical structure between coherence relations. Figure 24.1 shows the rhetorical struc-
ture of a paragraph from Marcu (2000a) for the text in (24.12) from the Scientific
American magazine.
(24.12) With its distant orbit–50 percent farther from the sun than Earth–and slim
atmospheric blanket, Mars experiences frigid weather conditions. Surface
temperatures typically average about -60 degrees Celsius (-76 degrees
Fahrenheit) at the equator and can dip to -123 degrees C near the poles. Only
the midday sun at tropical latitudes is warm enough to thaw ice on occasion,
but any liquid water formed in this way would evaporate almost instantly
because of the low atmospheric pressure.
Title 2-9
(1)
evidence
Mars
2-3 4-9
background elaboration-additional
Figure 24.1 A discourse tree for the Scientific American text in (24.12), from Marcu (2000a). Note that
asymmetric relations are represented with a curved arrow from the satellite to the nucleus.
The leaves in the Fig. 24.1 tree correspond to text spans of a sentence, clause or
EDU phrase that are called elementary discourse units or EDUs in RST; these units can
also be referred to as discourse segments. Because these units may correspond to
arbitrary spans of text, determining the boundaries of an EDU is an important task
for extracting coherence relations. Roughly speaking, one can think of discourse
24.1 • C OHERENCE R ELATIONS 551
Class Type
Example
TEMPORAL The parishioners of St. Michael and All Angels stop to chat at
SYNCHRONOUS
the church door, as members here always have. (Implicit while)
In the tower, five men and women pull rhythmically on ropes
attached to the same five bells that first sounded here in 1614.
CONTINGENCY REASON Also unlike Mr. Ruder, Mr. Breeden appears to be in a position
to get somewhere with his agenda. (implicit=because) As a for-
mer White House aide who worked closely with Congress,
he is savvy in the ways of Washington.
COMPARISON CONTRAST The U.S. wants the removal of what it perceives as barriers to
investment; Japan denies there are real barriers.
EXPANSION CONJUNCTION Not only do the actors stand outside their characters and make
it clear they are at odds with them, but they often literally stand
on their heads.
Figure 24.2 The four high-level semantic distinctions in the PDTB sense hierarchy
Temporal Comparison
• Asynchronous • Contrast (Juxtaposition, Opposition)
• Synchronous (Precedence, Succession) •Pragmatic Contrast (Juxtaposition, Opposition)
• Concession (Expectation, Contra-expectation)
• Pragmatic Concession
Contingency Expansion
• Cause (Reason, Result) • Exception
• Pragmatic Cause (Justification) • Instantiation
• Condition (Hypothetical, General, Unreal • Restatement (Specification, Equivalence, Generalization)
Present/Past, Factual Present/Past)
• Pragmatic Condition (Relevance, Implicit As- • Alternative (Conjunction, Disjunction, Chosen Alterna-
sertion) tive)
• List
Figure 24.3 The PDTB sense hierarchy. There are four top-level classes, 16 types, and 23 subtypes (not all
types have subtypes). 11 of the 16 types are commonly used for implicit ¯ argument classification; the 5 types in
italics are too rare in implicit labeling to be used.
EDU break 0 0 0 1
softmax
linear layer
ENCODER
Mr. Rambo says that …
Figure 24.4 Predicting EDU segment beginnings from encoded text.
Figure
Figure 24.5example
1: An ExampleofRST
RSTdiscourse tree,tree,
discourse showing four{e
where EDUs. Figure from Yu et al. (2018).
1 , e2 , e3 , e4 } are EDUs, attr and elab are
discourse relation labels, and arrows indicate the nuclearities of discourse relations.
Step Stack Queue Action Relation
2Previous
Transition-based
The Discourse
RSTParsing
decoder is then
transition-based a feedforward network W that outputs an action o based on a
discourse parsing studies exploit statistical models, using manually-
concatenation of the top three subtrees on the stack (so , s1 , s2 ) plus the first EDU in
designed
We follow Jidiscrete features
and Eisenstein
the (q(Sagae,
queue(2014),0 ):
2009;aHeilman
exploiting and Sagae,
transition-based framework2015; forWang et al., 2017).
RST discourse [Link] this work, we
The framework
propose is conceptually simple
a transition-based neuraland flexible
model fortoRST
support arbitraryparsing,
discourse features, which
which has been widely
follows an encoder-decoder
t , ht , ht , he ) (24.20) a
used in a number
framework. of NLP
Given tasks (Zhu
an input sequenceet al., of
2013;
o =Dyer
EDUsW(h {eet1s0al., 2015;
, ...,s2enZhang
, e2s1 q0 theetencoder
}, al., 2016). In addition,
computes the input represen-
transition-based model formalizes a certain task into predicting a sequence e of actions, which is essential
tations {h1 , h2 , ...,
e e e
hn },
where theand the decoder
representation of predicts
the EDU next on thestepqueue actions
h comes conditioned
directly on
fromthetheencoder outputs.
similar to sequence-to-sequence models proposed recently (Bahdanau et q0 al., 2014). In the following,
we first describe encoder, and the three hidden vectors representing partial trees
the transition system for RST discourse parsing, and then introduce our neural network are computed by
2.2.1 Encoder average pooling over the encoder output for the EDUs in those trees:
model by its encoder and decoder parts, respectively. Thirdly, we present our proposed dynamic oracle
We follow
strategy Li toet enhance
aiming al. (2016), using hierarchical
the transition-based [Link]-LSTMs
Then j to encode the the source method
EDU inputs, where the
t 1 weX introduce integration of
first-layer is used to represent sequencial
implicit syntax features. Finally we describe the training words inside
hs = method of our of EDUs, e and the second layer
hkneural network models.(24.21) is used to represent
j−i+1
sequencial EDUs. Given an input sentence {w1 , w2 , ...,k=i wm }, first we represent each word by its form
2.1 The Transition-based System
(e.g., wi ) and POS tag (e.g. ti ), concatenating their neural embeddings. By this way, the input vectors
The transition-based
of the framework converts
first-layer Bi-LSTM are {xw a structural learning problem
w into a sequence ofemb(t
action predic-
1 , x2 , ..., xm }, where xi = emb(wi ) i ), and then we apply
w w
tions, whose key point is a transition system. A transition system consists of two parts: states and actions.
Bi-LSTM directly, obtaining:
The states are used to store partially-parsed results and the actions are used to control state transitions.
w w w w w w
24.2 • D ISCOURSE S TRUCTURE PARSING 555
Training first maps each RST gold parse tree into a sequence of oracle actions, and
then uses the standard cross-entropy loss (with l2 regularization) to train the system
to take such actions. Give a state S and oracle action a, we first compute the decoder
output using Eq. 24.20, apply a softmax to get probabilities:
exp(oa )
pa = P (24.22)
exp(oa0 )
a0 ∈A
λ
LCE () = − log(pa ) + ||Θ||2 (24.23)
2
RST discourse parsers are evaluated on the test section of the RST Discourse Tree-
bank, either with gold EDUs or end-to-end, using the RST-Pareval metrics (Marcu,
2000b). It is standard to first transform the gold RST trees into right-branching bi-
nary trees, and to report four metrics: trees with no labels (S for Span), labeled
with nuclei (N), with relations (R), or both (F for Full), for each metric computing
micro-averaged F1 over all spans from all documents (Marcu 2000b, Morey et al.
2017).
24.3.1 Centering
Centering
Theory Centering Theory (Grosz et al., 1995) is a theory of both discourse salience and
discourse coherence. As a model of discourse salience, Centering proposes that at
any given point in the discourse one of the entities in the discourse model is salient:
it is being “centered” on. As a model of discourse coherence, Centering proposes
that discourses in which adjacent sentences CONTINUE to maintain the same salient
entity are more coherent than those which SHIFT back and forth between multiple
entities (we will see that CONTINUE and SHIFT are technical terms in the theory).
The following two texts from Grosz et al. (1995) which have exactly the same
propositional content but different saliences, can help in understanding the main
Centering intuition.
(24.28) a. John went to his favorite music store to buy a piano.
b. He had frequented the store for many years.
c. He was excited that he could finally buy a piano.
d. He arrived just as the store was closing for the day.
24.3 • C ENTERING AND E NTITY-BASED C OHERENCE 557
entity. In a RETAIN relation, the speaker intends to SHIFT to a new entity in a future
utterance and meanwhile places the current entity in a lower rank C f . In a SHIFT
relation, the speaker is shifting to a new salient entity.
Let’s walk though the start of (24.28) again, repeated as (24.30), showing the
representations after each utterance is processed.
(24.30) John went to his favorite music store to buy a piano. (U1 )
He was excited that he could finally buy a piano. (U2 )
He arrived just as the store was closing for the day. (U3 )
It was closing just as John arrived (U4 )
Using the grammatical role hierarchy to order the C f , for sentence U1 we get:
C f (U1 ): {John, music store, piano}
C p (U1 ): John
Cb (U1 ): undefined
and then for sentence U2 :
C f (U2 ): {John, piano}
C p (U2 ): John
Cb (U2 ): John
Result: Continue (C p (U2 )=Cb (U2 ); Cb (U1 ) undefined)
The transition from U1 to U2 is thus a CONTINUE. Completing this example is left
as exercise (1) for the reader
Government
Competitors
Department
3.3 Grid Construction: Linguistic Dimensions
Microsoft
Netscape
Evidence
Earnings
Products
Software
Markets
Brands
Tactics
Case
Trial
Suit
One of the central research issues in developing entity-based models of coherence is
determining what sources 1 S ofO linguistic
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resulting grid, columns that are dense (like the column for Microsoft) in-
Current 2). approaches recast coreference resolution as a classification task. A pair
dicate entities that are mentioned often in the texts; sparse columns (like the column
of NPs is classified as coreferring or not based on constraints that are learned from
foranearnings)
annotated indicate
corpus. entities
A separate that are mentioned rarely.
3.2 Entity Grids as Feature Vectorsclustering mechanism then coordinates the possibly
In the entity pairwise
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classifications andmeasured
constructs by apatterns
partition of local
on theentity
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sition. For example, Department is a subject in sentence
A fundamental assumption underlying our approach is that the distribution of system.
our experiments, we employ Ng and Cardie’s (2002) 1,
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sentences. Grids (70.4
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of length two). Fig. in 24.10
Table shows
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sparse columns which will consist mostly of gaps (see markets and earnings in Table
One
row d1would
Table ),3 and 2further expect that entities corresponding to dense columns are more often
other documents.
subjects
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feature-vector document representationwill be less pronounced
using all transitionsin low-coherence
of length two given texts.
syntactic
Inspiredcategories S , O , X , and
by Centering –.
Theory, our analysis revolves around patterns of local entity
transitions. A local entity transition is a sequence {S, O, X, –}n that represents entity
SS SO SX S– OS OO OX O– XS XO XX X– –S –O –X ––
occurrences and their syntactic roles in n adjacent sentences. Local transitions can be
d1 .01
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.59
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the probability
d3 .02 0 0 .03 .09 0 .09 .06 0 0 0 .05 .03 .07 .17 .39
transition [ S –] in the grid from Table 1 is 0.08 (computed as a ratio of its frequency
[i.e., six]
Figure 24.10 divided by the
A feature totalfor
vector number of transitions
representing documentsofusing length all two [i.e., 75]).
transitions [Link]
of length
can thusdbe
Document 1 isviewed
the textas inaFig.
distribution
24.9. Figure defined over transition
from Barzilay and Lapatatypes.(2008).
8
We can now go one step further and represent each text by a fixed set of transition
sequences using a standard feature vector notation. Each grid rendering j of a document
The transitions and their probabilities can then be used as features for a machine
di corresponds to a feature vector Φ(x ij ) = (p1 (x ij ), p2 (x ij ), . . . , pm (x ij )), where m is the
learning model. This model can be a text classifier trained to produce human-labeled
number of all predefined entity transitions, and pt (x ij ) the probability of transition t
coherence
in grid x ijscores (for example
. This feature vector from humans labeling
representation is usefully each text as coherent
amenable to machine or learning
inco-
herent).
algorithms But such (see data is expensive in
our experiments to gather.
SectionsBarzilay and Lapata (2005)
4–6). Furthermore, it allows introduced
the consid-
a eration
simplifying of large innovation:
numbers of coherence
transitions models
whichcan couldbe potentially
trained by uncoverself-supervision:
novel entity
distribution
trained patterns relevant
to distinguish the natural for coherence
original order assessment or other in
of sentences coherence-related
a discourse from tasks.
Note that considerable latitude is available when specifying the transition types to
be included in a feature vector. These can be all transitions of a given length (e.g., two
or three) or the most frequent transitions within a document collection. An example of
7
560 C HAPTER 24 • D ISCOURSE C OHERENCE
a subtopic have high cosine with each other, but not with sentences in a neighboring
subtopic.
A third early model, the LSA Coherence method of Foltz et al. (1998) was the
first to use embeddings, modeling the coherence between two sentences as the co-
sine between their LSA sentence embedding vectors1 , computing embeddings for a
sentence s by summing the embeddings of its words w:
sim(s,t) = cos(s, t)
X X
= cos( w, w) (24.31)
w∈s w∈t
and defining the overall coherence of a text as the average similarity over all pairs of
adjacent sentences si and si+1 :
n−1
1 X
coherence(T ) = cos(si , si+1 ) (24.32)
n−1
i=1
E p(s0 |si ) is the expectation with respect to the negative sampling distribution con-
ditioned on si : given a sentence si the algorithms samples a negative sentence s0
1 See Chapter 5 for more on LSA embeddings; they are computed by applying SVD to the term-
document matrix (each cell weighted by log frequency and normalized by entropy), and then the first
300 dimensions are used as the embedding.
562 C HAPTER 24 • D ISCOURSE C OHERENCE
Figure 24.11 The architecture of the LCD model of document coherence, showing the
computation of the score for a pair of sentences s and t. Figure from Xu et al. (2019).
uniformly over the other sentences in the same document. L is a loss function that
takes two scores, one for a positive pair and one for a negative pair, with the goal of
encouraging f + = fθ (si , si+1 ) to be high and f − = fθ (si , s0 )) to be low. Fig. 24.11
use the margin loss l( f + , f − ) = max(0, η − f + + f − ) where η is the margin hyper-
parameter.
Xu et al. (2019) also give a useful baseline algorithm that itself has quite high
performance in measuring perplexity: train an RNN language model on the data,
and compute the log likelihood of sentence si in two ways, once given the preceding
context (conditional log likelihood) and once with no context (marginal log likeli-
hood). The difference between these values tells us how much the preceding context
improved the predictability of si , a predictability measure of coherence.
Training models to predict longer contexts than just consecutive pairs of sen-
tences can result in even stronger discourse representations. For example a Trans-
former language model trained with a contrastive sentence objective to predict text
up to a distance of ±2 sentences improves performance on various discourse coher-
ence tasks (Iter et al., 2020).
Language-model style models are generally evaluated by the methods of Sec-
tion 24.3.3, although they can also be evaluated on the RST and PDTB coherence
relation tasks.
is pursued”) that have to occur in particular order, along with other components.
Propp shows that the plots of each of the fairy tales he studies can be represented as
a sequence of these functions, different tales choosing different subsets of functions,
but always in the same order. Indeed Lakoff (1972) showed that Propp’s model
amounted to a discourse grammar of stories, and in recent computational work Fin-
layson (2016) demonstrates that some of these Proppian functions could be induced
from corpora of folktale texts by detecting events that have similar actions across
stories. Bamman et al. (2013) showed that generalizations over dramatis personae
could be induced from movie plot summaries on Wikipedia. Their model induced
latent personae from features like the actions the character takes (e.g., Villains stran-
gle), the actions done to them (e.g., Villains are foiled and arrested) or the descriptive
words used of them (Villains are evil).
In this section we introduce two kinds of such global discourse structure that
have been widely studied computationally. The first is the structure of arguments:
the way people attempt to convince each other in persuasive essays by offering
claims and supporting premises. The second is somewhat related: the structure of
scientific papers, and the way authors present their goals, results, and relationship to
prior work in their papers.
Figure
Figure 2
24.12 Argumentation structure of a persuasive essay. Arrows indicate argumentation relations, ei-
Argumentation structure of the example essay. Arrows indicate argumentative relations.
ther of SUPPORT (with arrowheads) or ATTACK (with circleheads); P denotes premises. Figure from Stab and
Arrowheads denote argumentative support relations and circleheads attack relations. Dashed
Gurevych (2017).relations that are encoded in the stance attributes of claims. “P” denotes premises.
lines indicate
sive essay will have only a single main claim, with premises spread throughout 629the
text, without the local coherence we see in coherence relations.
Algorithms for detecting argumentation structure often include classifiers for
distinguishing claims, premises, or non-argumentation, together with relation clas-
sifiers for deciding if two spans have the SUPPORT, ATTACK, or neither relation
(Peldszus and Stede, 2013). While these are the main focus of much computational
work, there is also preliminary efforts on annotating and detecting richer semantic
relationships (Park and Cardie 2014, Hidey et al. 2017) such as detecting argumen-
argumentation tation schemes, larger-scale structures for argument like argument from example,
schemes
or argument from cause to effect, or argument from consequences (Feng and
Hirst, 2011).
Another important line of research is studying how these argument structure (or
other features) are associated with the success or persuasiveness of an argument
(Habernal and Gurevych 2016, Tan et al. 2016, Hidey et al. 2017. Indeed, while it
is Aristotle’s logos that is most related to discourse structure, Aristotle’s ethos and
pathos techniques are particularly relevant in the detection of mechanisms of this
persuasion sort of persuasion. For example scholars have investigated the linguistic realization
of features studied by social scientists like reciprocity (people return favors), social
proof (people follow others’ choices), authority (people are influenced by those
with power), and scarcity (people value things that are scarce), all of which can
be brought up in a persuasive argument (Cialdini, 1984). Rosenthal and McKeown
(2017) showed that these features could be combined with argumentation structure
to predict who influences whom on social media, Althoff et al. (2014) found that
linguistic models of reciprocity and authority predicted success in online requests,
while the semisupervised model of Yang et al. (2019) detected mentions of scarcity,
commitment, and social identity to predict the success of peer-to-peer lending plat-
forms.
See Stede and Schneider (2018) for a comprehensive survey of argument mining.
24.6 • S UMMARY 565
Teufel et al. (1999) and Teufel et al. (2009) develop labeled corpora of scientific
articles from computational linguistics and chemistry, which can be used as supervi-
sion for training standard sentence-classification architecture to assign the 15 labels.
24.6 Summary
In this chapter we introduced local and global models for discourse coherence.
• Discourses are not arbitrary collections of sentences; they must be coherent.
Among the factors that make a discourse coherent are coherence relations
between the sentences, entity-based coherence, and topical coherence.
• Various sets of coherence relations and rhetorical relations have been pro-
posed. The relations in Rhetorical Structure Theory (RST) hold between
spans of text and are structured into a tree. Because of this, shift-reduce
and other parsing algorithms are generally used to assign these structures.
The Penn Discourse Treebank (PDTB) labels only relations between pairs of
spans, and the labels are generally assigned by sequence models.
• Entity-based coherence captures the intuition that discourses are about an
entity, and continue mentioning the entity from sentence to sentence. Cen-
tering Theory is a family of models describing how salience is modeled for
566 C HAPTER 24 • D ISCOURSE C OHERENCE
Historical Notes
Coherence relations arose from the independent development of a number of schol-
ars, including Hobbs (1979) idea that coherence relations play an inferential role for
the hearer, and the investigations by Mann and Thompson (1987) of the discourse
structure of large texts. Other approaches to coherence relations and their extrac-
SDRT tion include Segmented Discourse Representation Theory (SDRT) (Asher and Las-
carides 2003, Baldridge et al. 2007) and the Linguistic Discourse Model (Polanyi
1988, Scha and Polanyi 1988, Polanyi et al. 2004). Wolf and Gibson (2005) argue
that coherence structure includes crossed bracketings, which make it impossible to
represent as a tree, and propose a graph representation instead. A compendium of
over 350 relations that have been proposed in the literature can be found in Hovy
(1990).
RST parsing was first proposed by Marcu (1997), and early work was rule-based,
focused on discourse markers (Marcu, 2000a). The creation of the RST Discourse
TreeBank (Carlson et al. 2001, Carlson and Marcu 2001) enabled a wide variety
of machine learning algorithms, beginning with the shift-reduce parser of Marcu
(1999) that used decision trees to choose actions, and continuing with a wide variety
of machine learned parsing methods (Soricut and Marcu 2003, Sagae 2009, Hernault
et al. 2010, Feng and Hirst 2014, Surdeanu et al. 2015, Joty et al. 2015) and chunkers
(Sporleder and Lapata, 2005). Subba and Di Eugenio (2009) integrated sophisticated
semantic information into RST parsing. Ji and Eisenstein (2014) first applied neural
models to RST parsing neural models, leading to the modern set of neural RST
models (Li et al. 2014, Li et al. 2016, Braud et al. 2017, Yu et al. 2018, inter alia) as
well as neural segmenters (Wang et al. 2018b). and neural PDTB parsing models (Ji
and Eisenstein 2015, Qin et al. 2016, Qin et al. 2017).
Barzilay and Lapata (2005) pioneered the idea of self-supervision for coher-
ence: training a coherence model to distinguish true orderings of sentences from
random permutations. Li et al. (2014) first applied this paradigm to neural sentence-
representation, and many neural self-supervised models followed (Li and Jurafsky
2017, Logeswaran et al. 2018, Lai and Tetreault 2018, Xu et al. 2019, Iter et al.
2020)
Another aspect of global coherence is the global topic structure of a text, the way
the topics shift over the course of the document. Barzilay and Lee (2004) introduced
an HMM model for capturing topics for coherence, and later work expanded this
intuition (Soricut and Marcu 2006, Elsner et al. 2007, Louis and Nenkova 2012, Li
and Jurafsky 2017).
The relationship between explicit and implicit discourse connectives has been
a fruitful one for research. Marcu and Echihabi (2002) first proposed to use sen-
H ISTORICAL N OTES 567
tences with explicit relations to help provide training data for implicit relations, by
removing the explicit relations and trying to re-predict them as a way of improv-
ing performance on implicit connectives; this idea was refined by Sporleder and
Lascarides (2005), (Pitler et al., 2009), and Rutherford and Xue (2015). This rela-
tionship can also be used as a way to create discourse-aware representations. The
DisSent algorithm (Nie et al., 2019) creates the task of predicting explicit discourse
markers between two sentences. They show that representations learned to be good
at this task also function as powerful sentence representations for other discourse
tasks.
The idea of entity-based coherence seems to have arisen in multiple fields in the
mid-1970s, in functional linguistics (Chafe, 1976), in the psychology of discourse
processing (Kintsch and Van Dijk, 1978), and in the roughly contemporaneous work
of Grosz, Sidner, Joshi, and their colleagues. Grosz (1977a) addressed the focus
of attention that conversational participants maintain as the discourse unfolds. She
defined two levels of focus; entities relevant to the entire discourse were said to
be in global focus, whereas entities that are locally in focus (i.e., most central to
a particular utterance) were said to be in immediate focus. Sidner 1979; 1983 de-
scribed a method for tracking (immediate) discourse foci and their use in resolving
pronouns and demonstrative noun phrases. She made a distinction between the cur-
rent discourse focus and potential foci, which are the predecessors to the backward-
and forward-looking centers of Centering theory, respectively. The name and further
roots of the centering approach lie in papers by Joshi and Kuhn (1979) and Joshi and
Weinstein (1981), who addressed the relationship between immediate focus and the
inferences required to integrate the current utterance into the discourse model. Grosz
et al. (1983) integrated this work with the prior work of Sidner and Grosz. This led
to a manuscript on centering which, while widely circulated since 1986, remained
unpublished until Grosz et al. (1995). A collection of centering papers appears in
Walker et al. (1998). See Karamanis et al. (2004) and Poesio et al. (2004) for a
deeper exploration of centering and its parameterizations, and the History section of
Chapter 23 for more on the use of centering on coreference.
The grid model of entity-based coherence was first proposed by Barzilay and
Lapata (2005) drawing on earlier work by Lapata (2003) and Barzilay, and then
extended by them Barzilay and Lapata (2008) and others with additional features
(Elsner and Charniak 2008, 2011, Feng et al. 2014, Lin et al. 2011) a model that
projects entities into a global graph for the discourse (Guinaudeau and Strube 2013,
Mesgar and Strube 2016), and a convolutional model to capture longer-range entity
dependencies (Nguyen and Joty, 2017).
Theories of discourse coherence have also been used in algorithms for interpret-
ing discourse-level linguistic phenomena, including verb phrase ellipsis and gap-
ping (Asher 1993, Kehler 1993), and tense interpretation (Lascarides and Asher
1993, Kehler 1994, Kehler 2000). An extensive investigation into the relationship
between coherence relations and discourse connectives can be found in Knott and
Dale (1994).
Useful surveys of discourse processing and structure include Stede (2011) and
Webber et al. (2012).
Andy Kehler wrote the Discourse chapter for the 2000 first edition of this text-
book, which we used as the starting point for the second-edition chapter, and there
are some remnants of Andy’s lovely prose still in this third-edition coherence chap-
ter.
568 C HAPTER 24 • D ISCOURSE C OHERENCE
Exercises
24.1 Finish the Centering Theory processing of the last two utterances of (24.30),
and show how (24.29) would be processed. Does the algorithm indeed mark
(24.29) as less coherent?
24.2 Select an editorial column from your favorite newspaper, and determine the
discourse structure for a 10–20 sentence portion. What problems did you
encounter? Were you helped by superficial cues the speaker included (e.g.,
discourse connectives) in any places?
CHAPTER
Conversation is an intricate and complex joint activity, and conversations have struc-
ture. This is true of all conversations, whether they are conversations between people
or conversations between people and language models. Understanding the structure
of human conversations is an important social science and linguistic task. The con-
cepts we introduce in studying human conversation can also be a useful tool for
analyzing human-LLM conversations.
[This draft is the initial stub of a chapter that will introduce different kinds of
conversational structure and how to annotate them computationally.]
25.1.1 Turns
turn A dialogue is a sequence of turns (C1 , A2 , C3 , and so on), each a single contribution
from one speaker to the dialogue (as if in a game: I take a turn, then you take a turn,
then me, and so on). There are 20 turns in Fig. 25.1. A turn can consist of a sentence
(like C1 ), although it might be as short as a single word (C13 ) or as long as multiple
sentences (A10 ).
Turn structure has important implications for spoken dialogue. A human has
to know when to stop talking; the client interrupts (in A16 and C17 ), so a system
that was performing this role must know to stop talking (and that the user might be
making a correction).
The same issues come up for LLMs; a system also has to know when to start
talking. For example, most of the time in conversation, speakers start their turns
almost immediately after the other speaker finishes, without a long pause, because
people are can usually predict when the other person is about to finish talking.
Spoken language models must also detect whether a user is done speaking, so
endpointing they can process the utterance and respond. This task—called endpointing or end-
point detection— can be quite challenging because of noise and because people
often pause in the middle of turns.
25.1.3 Grounding
A dialogue is not just a series of independent speech acts, but rather a collective act
performed by the speaker and the hearer. Like all collective acts, it’s important for
common
ground the participants to establish what they both agree on, called the common ground
grounding (Stalnaker, 1978). Speakers do this by grounding each other’s utterances. Ground-
25.1 • P ROPERTIES OF H UMAN C ONVERSATION 571
ing means acknowledging that the hearer has understood the speaker (Clark, 1996).
(People need grounding for non-linguistic actions as well; the reason an elevator but-
ton lights up when it’s pressed is to acknowledge that the elevator has indeed been
called, essentially grounding your action of pushing the button (Norman, 1988).)
Grounding is also important when the hearer needs to indicate that the speaker
has not succeeded in performing an action. If the hearer has problems in under-
standing, she must indicate these problems to the speaker, again so that mutual un-
derstanding can eventually be achieved.
How is closure achieved? Clark and Schaefer (1989) introduce the idea that each
contribution joint linguistic act or contribution has two phases, called presentation and accep-
tance. In the first phase, a speaker presents the hearer with an utterance, performing
a sort of speech act. In the acceptance phase, the hearer has to ground the utterance,
indicating to the speaker whether understanding was achieved.
What methods can the hearer B use to ground the speaker A’s utterance? Clark
and Schaefer (1989) discuss a continuum of methods ordered from weakest to strongest:
Continued attention: B shows she is continuing to attend and therefore remains satisfied with
A’s presentation.
Next contribution: B starts in on the next relevant contribution.
Acknowledgment: B nods or says a continuer like uh-huh, yeah, or the like, or an assess-
ment like that’s great.
Demonstration: B demonstrates all or part of what she has understood A to mean, for
example, by reformulating (paraphrasing) A’s utterance or by collabo-
rative completion of A’s utterance.
Display: B displays verbatim all or part of A’s presentation.
continuer The word uh-huh here is a continuer, also often called an acknowledgment to-
backchannel ken or a backchannel. A continuer is a (short) optional utterance that acknowledges
the content of the utterance of the other and that doesn’t require an acknowledgment
by the other (Yngve, 1970; Jefferson, 1984; Schegloff, 1982; Ward and Tsukahara,
2000).
composed of a first pair part and a second pair part (Schegloff, 1968), and these
expectations can help systems decide what actions to take.
However, dialogue acts aren’t always followed immediately by their second pair
side sequence part. The two parts can be separated by a side sequence (Jefferson 1972) or sub-
subdialogue dialogue. For example utterances C17 to A20 constitute a correction subdialogue
(Litman 1985, Litman and Allen 1987, Chu-Carroll and Carberry 1998):
C17 : #Act. . . actually#, what day of the week is the 15th?
A18 : It’s a Friday.
C19 : Uh hmm. I would consider staying there an extra day til Sunday.
A20 : OK. . . OK. On Sunday I have . . .
The question in C17 interrupts the prior discourse, in which the agent was looking
for a May 15 return flight. The agent must answer the question and also realize that
‘’I would consider staying...til Sunday” means that the client would probably like to
change their plan, and now go back to finding return flights, but for the 17th.
Another side sequence is the clarification question, which can form a subdia-
logue between a REQUEST and a RESPONSE. This is especially common in dialogue
systems where speech recognition errors causes the system to have to ask for clari-
fications or repetitions like the following:
User: What do you have going to UNKNOWN WORD on the 5th?
System: Let’s see, going where on the 5th?
User: Going to Hong Kong.
System: OK, here are some flights...
presequence In addition to side-sequences, questions often have presequences, like the fol-
lowing example where a user starts with a question about the system’s capabilities
(“Can you make train reservations”) before making a request.
User: Can you make train reservations?
System: Yes I can.
User: Great, I’d like to reserve a seat on the 4pm train to New York.
25.1.5 Initiative
Sometimes a conversation is completely controlled by one participant. For exam-
ple a reporter interviewing a chef might ask questions, and the chef responds. We
initiative say that the reporter in this case has the conversational initiative (Carbonell, 1970;
Nickerson, 1976). In normal human-human dialogue, however, it’s more common
for initiative to shift back and forth between the participants, as they sometimes
answer questions, sometimes ask them, sometimes take the conversations in new di-
rections, sometimes not. You may ask me a question, and then I respond asking you
to clarify something you said, which leads the conversation in all sorts of ways. We
call such interactions mixed initiative (Carbonell, 1970).
Full mixed initiative, while the norm for human-human conversations, can be
difficult for dialogue systems. The most primitive dialogue systems tend to use
system-initiative, where the system asks a question and the user can’t do anything
until they answer it, or user-initiative like simple search engines, where the user
specifies a query and the system passively responds. Even modern large language
model-based dialogue systems, which come much closer to using full mixed initia-
tive, often don’t have completely natural initiative switching. Getting this right is an
important goal for modern systems.
25.2 • D IALOG ACTS AND C ORPORA 573
Tag Example
T HANK Thanks
G REET Hello Dan
I NTRODUCE It’s me again
B YE Alright bye
R EQUEST-C OMMENT How does that look?
S UGGEST from thirteenth through seventeenth June
R EJECT No Friday I’m booked all day
ACCEPT Saturday sounds fine
R EQUEST-S UGGEST What is a good day of the week for you?
I NIT I wanted to make an appointment with you
G IVE R EASON Because I have meetings all afternoon
F EEDBACK Okay
D ELIBERATE Let me check my calendar here
C ONFIRM Okay, that would be wonderful
C LARIFY Okay, do you mean Tuesday the 23rd?
D IGRESS [we could meet for lunch] and eat lots of ice cream
M OTIVATE We should go to visit our subsidiary in Munich
G ARBAGE Oops, I-
Figure 25.2 The 18 high-level dialogue acts for a meeting scheduling task, from the
Verbmobil-1 system (Jekat et al., 1995).
the work of Clark and Schaefer (1989), Allwood et al. (1992), and (Allwood, 1995),
each utterance is tagged for two types of functions, forward-looking functions like
speech act functions, and backward-looking functions, like grounding and answer-
ing, which “look back” to the interlocutor’s previous utterance (Allen and Core,
1997; Walker et al., 1996; Carletta et al., 1997; Core et al., 1999).
25.2 • D IALOG ACTS AND C ORPORA 575
577
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606 Bibliography
607
608 Subject Index
simplifying assumptions interpretable, 89 latent semantic analysis, LSI, see latent semantic
for POS tagging, interval algebra, 464 117 analysis
388 intonation phrases, 313 lateral sound, 310 LSTM, 401
states, 386 intrinsic evaluation, 43 layer norm, 179 LUNAR, 251
transition probabilities, inversion transduction LDC, 18
386 grammar (ITG), 277 learning rate, 75 machine learning
Hobbs algorithm, 544 inverted index, 239 lemma, 96 for NER, 398
Hobbs tree search algorithm IO, 212, 384 Levenshtein distance, 30 textbooks, 94
for pronoun IOB tagging lexical machine translation, 253
resolution, 544 for temporal expressions, category, 405 macroaveraging, 83
hold, as dialogue act, 575 469 cohesion, 548, 560 MAE, 20
homonymy, 206 IPA, 305, 331 gap, 257 Mandarin, 255, 306
hot languages, 258 IR, 235 semantics, 96 Manhattan distance
HuBERT, 346 idf term weighting, 236 stress, 312 in L1 regularization, 91
Hungarian term weighting, 236 trigger, in IE, 468 manner adverb, 380
part-of-speech tagging, IRB, 168 lexico-syntactic pattern, manner of articulation, 309
398 is-a, 453 454 Markov, 39
hybrid, 359 ISO 8601, 470 lexicon, 404 assumption, 39
hypernym, 453 isolating language, 9, 257 LibriSpeech, 335 Markov assumption, 385
lexico-syntactic patterns iSRL, 495 light verbs, 463 Markov chain, 57, 385
for, 454 ITG (inversion transduction linear chain CRF, 392, 393 formal definition of, 386
hyperparameter, 77 grammar), 277 linear interpolation for initial distribution, 386
hyperparameters, 143 n-grams, 53 n-gram as, 385
Hz as unit of measure, 315 linearly separable, 123
Japanese, 255, 257, 305, states, 386
306 Linguistic Data transition probabilities,
IBM Models, 277 Consortium, 18 386
IBM Thomas J. Watson Linguistic Discourse Markov model, 39
k-means, 348 model, 566
Research Center, formal definition of, 386
Kaldi, 360 Link Grammar, 449
58, 358 history, 58
KBP, 475 List (as coherence relation),
idf term weighting, 236 Marx, G., 403
KenLM, 43, 58 550
immediately dominates, Masked Language
kernel, 337 listen attend and spell, 341
405 Modeling, 200
key, 175 LIWC, 501
implicature, 573 mass nouns, 379
KL divergence, 490 LM, 37
implicit argument, 495 max-pooling, 133
Klatt formant synthesizer, LOB corpus, 400
in-context learning, 193 maxent, 93
373 localization, 253
indefinite reference, 520 maxim, Gricean, 573
Kleene *, 22 locative, 380
induction heads, 194 maximum entropy, 93
sneakiness of matching locative adverb, 380
inference-based learning, maximum spanning tree,
zero things, 22 log
445 442
Kleene +, 23 why used for
inflectional morphemes, 8 Mayan, 257
knowledge claim, 565 probabilities, 42
infoboxes, 453 MBR, 267
knowledge graphs, 451 why used to compress
information McNemar’s test, 357
Korean, 306 speech, 316, 326
structure, 521 mean
Koryak, 9 log likelihood ratio, 509
status, 521 element-wise, 288
Kullback-Leibler log odds ratio, 509
information extraction (IE), mean average precision,
divergence, 490 log probabilities, 42, 42
451 241
KV cache, 191 logistic function, 64
bootstrapping, 457 mean opinion score, 371
information gain, 94 logistic regression mean reciprocal rank, 250
for feature selection, 94 L* pitch accent, 314 conditional maximum mean-pooling, 133
Information retrieval, 235 L+H* pitch accent, 314 likelihood mechanical indexing, 116
information retrieval, 234 L1 regularization, 91 estimation, 73 Mechanical Turk, 361
initiative, 572 L2 regularization, 90 Gaussian priors, 91 mel, 328
inner ear, 322 labeled precision, 421 learning in, 72 frequency cepstral
inner product, 102 labeled recall, 421 regularization, 91 coefficients, 329
instance, word, 5 labial place of articulation, relation to neural scale, 318
Institutional Review Board, 308 networks, 127 memory networks, 195
168 labiodental consonants, 308 logit, 65, 187 mention detection, 526
Instruction tuning, 217 language logit lens, 194 mention-pair, 529
intensity of sound, 318 identification, 372 logos, 563 mentions, 517
intercept, 64 universal, 254 long short-term memory, MERT, for training in MT,
Interjections, 380 language model, 37 292 277
intermediate phrase, 313 language model:coined by, lookahead in regex, 28 Message Understanding
International Phonetic 58 LoRA, 192 Conference, 473
Alphabet, 305, 331 language modeling head, loss, 72 METEOR, 278
interpolated precision, 241 186 loudness, 319 metonymy, 546
interpolation Laplace smoothing, 50 low frame rate, 342 MFCC, 329
in smoothing, 53 larynx, 307 LPC (Linear Predictive microaveraging, 83
interpretability, 193 lasso regression, 91 Coding), 333, 358 Microsoft .wav format, 316
Subject Index 611
precision-recall curve, 241 random sampling, 156 Rhetorical Structure self-supervised, 346
preference-based learning, range, regular expression, Theory, see RST self-supervision, 105, 284
222 22 rhyme, syllable, 311 self-training, 158
premises, 563 ranking, 271 Riau Indonesian, 380 semantic drift in IE, 458
prepositional phrase rarefaction, 315, 325 ridge regression, 91 semantic feature, 117
constituency, 406 RDF, 453 rime semantic field, 97
prepositions, 380 RDF triple, 453 syllable, 311 semantic relations in IE,
presequences, 572 Read speech, 335 RMS amplitude, 317 452
pretokenization, 16 reading comprehension, RNN-T, 354 table, 453
pretraining, 146 248 role-filler extraction, 473 semantic role, 478, 478,
primitive decomposition, Reason (as coherence root, 8 480
492 relation), 549 Rosebud, sled named, 547 Semantic role labeling, 484
principle of contrast, 97 Recall, 82 rounded vowels, 311 semantics
pro-drop languages, 258 recall RST, 549 lexical, 96
probabilistic context-free for MT evaluation, 278 TreeBank, 551, 566 semivowel, 308
grammars, 425 in NER, 214, 397 rules sense
productions, 404 receptive field, 339 context-free, 404 word, 206
projective, 430 reconstruction loss, 368 context-free, expansion, sentence
prominence, phonetic, 313 rectangular, 326 405 error rate, 355
prominent word, 312 reduced vowels, 313 context-free, sample, 406 segmentation, 19
prompt, 151 reduction, phonetic, 313 Russian sentence separation, 297
prompt engineering, 151 reference fusion language, 9, 257 SentencePiece, 260
pronoun, 380 bound pronouns, 520 verb-framed, 257 sentiment, 98
bound, 520 cataphora, 520 RVQ, 367 origin of term, 516
demonstrative, 521 definite, 520 sentiment analysis, 61
non-binary, 524 generics, 523 SentiWordNet, 506
personal, 380 S as start symbol in CFG,
indefinite, 520 406 sequence labeling, 378
possessive, 380 reference point, 465 SFT, 217
wh-, 380 salience, in discourse
referent, 517 model, 522 SGNS, 104
pronunciation dictionary, accessing of, 517 Shakespeare
306 Sampling, 47
evoking of, 517 sampling, 155 n-gram approximations
CELEX, 306 referential density, 258 to, 48
CMU, 306 of analog waveform, 315,
reflexive, 524 325 shallow discourse parsing,
PropBank, 481 reformulation, 571 555
proper noun, 379 rate, 315, 325
regex satellite, 257, 549 sibilant sound, 310
prosodic phrasing, 313 regular expression, 21 side sequence, 572
Prosody, 312 satellite-framed language,
regression 257 sigmoid, 64, 120
prosody lasso, 91 significance test
accented syllables, 312 saturated, 122
ridge, 91 scaling laws, 190 MAPSSWE for ASR,
reduced vowels, 313 regular expression, 21, 34 356
PROTO - AGENT, 480 schwa, 313
substitutions, 26 SCISOR, 475 McNemar’s, 357
PROTO - PATIENT , 480
regularization, 90 sclite, 355 similarity, 97
prototype
relatedness, 97 sclite package, 35 cosine, 103
in clustering and VQ,
relation extraction, 451 script singleton, 518
350
relative Schankian, 483 singular they, 524
pseudoword, 492
temporal expression, 468 scripts, 472 skip-gram, 104
PTRANS, 493
relative entropy, 490 SDRT (Segmented slot filling, 475
punctuation
relative frequency, 41 Discourse smoothing, 50, 50
for numbers
release, stop, 309 Representation add-one, 50
cross-linguistically,
18 relevance, 573 Theory), 566 interpolation, 53
for sentence ReLU, 121 search engine, 235 Laplace, 50
segmentation, 19 reporting events, 463 search tree, 264 linear interpolation, 53
tokenization, 18 representation learning, 95 second-order softmax, 70, 127
treated as words, 5 representational harm, 114 co-occurrence, 111 source-filter model, 323
treated as words in LM, representational harms, 88 seed pattern in IE, 457 SOV language, 255
49 rescore, 345 seed tuples, 457 spam detection, 61
residual segmentation span, 419
in RVQ, 367 sentence, 19 Spanish, 306
quantization, 316, 326 residual stream, 177 selectional association, 491 Speaker diarization, 372
query, 175, 235 residual vector selectional preference speaker identification, 372
in IR, 235 quantization, 367 strength, 490 speaker recognition, 372
question resolve, 382 selectional preferences speaker verification, 372
rise, 313 Resource Management, 358 pseudowords for spectrogram, 322
questions retrieval-augmented evaluation, 492 spectrum, 320
factoid, 233 generation, 246 selectional restriction, 488 speech
ReVerb, 461 representing with events, telephone bandwidth,
Radio Rex, 334 reward, 225 489 316, 326
RAG, 234, 246 rewrite, 405 violations in WSD, 490 speech acts, 570
Subject Index 613