Department of Electrical & Computer Engineering
ELEC 691X/498X – Broadcast Signal Transmission
Fall 2015
Instructor: Dr. Reza Soleymani, Office: EV‐5.125,
Telephone: 848‐2424 ext.: 4103.
Office Hours: Wednesday, Thursday, 14:00 – 15:00
Time: Tuesday, 2:45 to 5:30
Room: H 411
Slide 1
Department of Electrical & Computer Engineering
In this lecture we cover the following topics:
• Sampling.
• Quantization.
• Digital Interfaces: SDI, ASI, etc.
• Picture Compression: JPEG.
• Moving Picture Compression: MPEG.
Slide 2
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Slide 3
Lecture 2:
Sampling
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Nyquist theorem:
• The number of samples per second, i.e., the sampling rate
should be more than or equal to twice the highest
frequency of the analog signal:
2 ,
or,
.
• If there will be aliasing.
Slide 4
Lecture 2:
Critical Sampling
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Slide 5
Lecture 2:
Aliasing
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Slide 6
Lecture 2:
Sampling: Voice and Audio
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• Human voice has frequencies up to 3.4 kHz. So, the
minimum sampling frequency for voice is 6.8 k
samples/sec. But to make filtering easier a sampling
rate of 8 ksps is used, i.e., one sample every 125
• For an audio signal a frequency range of 20 to 20,000
Hz. is considered. This is the range of frequencies
human auditory system can detect. So, at least 40,000
samples per second is required. Usually a sampling
rate of 44.1 ksps is used for audio.
Slide 7
Lecture 2:
Sampling: Image
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While sound is a one dimensional, temporal (time
varying) signal, an image is a two-dimensional spatial
signal. Therefore, the sampling interval instead of
having dimension of time, has the dimension of length.
The horizontal and vertical resolution is defined in term
of and , respectively or their inverse and .
They represent the number of samples (pixels) in each
row and column. The number of pixels depends on the
frequency in X and Y dimension. A “busy” image has
higher frequency components hence needing more
samples.
Slide 8
Lecture 2:
Sampling: Video
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Video or moving picture is a sequence of images in time, so in
addition to two spatial dimensions it is also a function of time,
, , . The number of pixels in a frame determines the spatial
resolution. The temporal resolution is determined by the number
of images shown (frames) per second.
In the previous lecture, we saw that in order to prevent flickering
and at the same time not to increase the data rate, a frame
sometimes is divided into two fields (odd and even fields) and
fields are shown in an interlaced fashion. This is called
interlaced scan as opposed to progressive scan where the whole
frame is scanned. For example 480p denotes a format with 480
lines per frame and 480 640 pixels per line if the aspect
ratio is 4/3 and 853 pixels per line for the aspect ratio of 16/9.
Slide 9
Lecture 2:
Sampling: Video
Department of Electrical & Computer Engineering
Video or moving picture is a sequence of images in time, so in
addition to two spatial dimensions it is also a function of time,
, , . The number of pixels in a frame determines the spatial
resolution. The temporal resolution is determined by the number
of images shown (frames) per second.
In the previous lecture, we saw that in order to prevent flickering
and at the same time not to increase the data rate, a frame
sometimes is divided into two fields (odd and even fields) and
fields are shown in an interlaced fashion. This is called
interlaced scan as opposed to progressive scan where the whole
frame is scanned. For example 480p denotes a format with 480
lines per frame and 480 640 pixels per line if the aspect
ratio is 4/3 and 853 pixels per line for the aspect ratio of 16/9.
Slide 10
Lecture 2:
Sampling: Video
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• Similarly 720p (HD) is a video format with 1280 720 pixels
per frame.
• 1080i (HD) and 1080p (True HD) refers to a 1920 1080
pixel interlaced or progressive scan format, respectively.
• 2160p or 4K format with 3840 x 2160 pixels called UHDTV.
The number of frames per second can be 25p, 30p, 50i, 60i,
50p/60p, 100p/120p, 300p.
Slide 11
Lecture 2:
Analog to Digital Conversion
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After sampling a source whether audio or video, we need to
convert the voltage level obtained into a finite number of values
so that we can represent each sample of the audio signal or each
pixel with a finite number of bits.
Slide 12
Lecture 2:
ADC: Quantization Error
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An input sample is mapped into a discrete level .
So, the quantization error is . The average
squared error (MSE) will be,
p(x)dx
∑ p(x)dx.
Where , 0, 1, … , are the thresholds and
, 1, … , are the discrete level value. That is if
, then .
Slide 13
Lecture 2:
ADC: Quantization Error
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• For a uniform source is uniformly distributed
between and .
Slide 14
Lecture 2:
ADC: Quantization Error
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• So,
Let the peak-to-peak value of the signal be 2V, i.e., and
. Then and .
Where log is the number of bits required for representing
L levels of the ADC.
Denoting the signal power by , we have the signal-to-
quantization-noise ratio (SQNR) in dB as:
3
10 log 6 .
Slide 15
Lecture 2:
ADC: Quantization Error
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• Exercise 1: a)Find the SQNR in dB for a sinusoidal signal with
amplitude A quantized with an 8 bit uniform quantizer.
b) Find SQNR for a Gaussian source quantized with an 8 bit
uniform quantizer. The probability of overload should be
less than 1%.
c) Find the SQNR for a Gaussian source designed for it.
Compare with what information theory (rate-distortion
theory) suggests.
Slide 16
Lecture 2:
Raw bit rate
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We so far have learnt about the number of samples required to
represent a source whether temporal or spatial (pixels in image
or video) and the fact that each bit added to the ADC gives us
roughly an extra 6 dB of signal quality. Now let’s find what is the
bit rate for transmitting or storing a sampled and quantized
source. We do this for voice, audio and video. This, particularly in
the case of video, gives ridiculously large values. This gives us an
appreciation for the work gone into audio and video
compression as well as advanced digital coding and modulation
techniques that have brought the bit rates into a reasonably low
value allowing us to receive and retrieve audio and video signal
with extremely high quality over a large variety of platforms.
Slide 17
Lecture 2:
Raw bit rate
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• Let’s start with voice signal. We said that voice signals can be
represented by samples taken every 125 . That is they are
sampled at the rate of 8000 samples per second. If we are
content with 48 dB of SQNR, we can represent each sample
with 8 bits. This means that in order to transmit the voice signals
over the phone, we need 64 kbps. This is actually the rate used
at the start of digital telephony. It was called a Voice Channel
(VC). The technique was called Pulse Code Modulation
(PCM). With the advances in voice coding rate were reduced
to 32 kbps (DPCM), 16 kbps (ADPCM) and less than 8 kbps with
Linear Predictive Coding (LPC) techniques such as CELP.
• For audio sampling is at the rate of 44.1 ksps and each sample
is quantized with 16 bits so the total arte is 2 44.1 16 1.411
Mbits/sec. This is called CD format. Using mp3 roughly the
same perceptual can be obtained with 128 kbps (1/11
compression).
Slide 18
Lecture 2:
Raw bit rate
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• For Video, we need three color samples per pixel. So, if the
number of pixels are and and there are , the raw bit
rate will be
3 .
• Let’s take 1080 . Then,
3 8 30 1080 1920 1.5 Gbps.
In the above, we have assumed 8 bit quantization. The industry is
moving towards 10 bit for some applications.
• A Blu-ray disk that has 50 GB capacity can only store 4.5
minutes of raw video. And we have not even added audio
and metadata. So, let’s see what compression can do for us.
Slide 19
Lecture 2:
RGB
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• Any colour can be represented as a linear combination of
Red Green Blue.
• As we saw in the previous lecture the three colour scheme
RGB or Component can be transformed into or
sometimes called YUV or Lab, where the first component is the
luminance and the other two are Chroma. The simplest way to
transform component into composite is Y=R+G+B,
and . However, usually some matrices are used
based on the human visual perception of colour.
Slide 20
Lecture 2:
RGB
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• The first thing we do to reduce the image (or video) size is to
use the fact that the eye is less sensitive to colour than to
brightness. So instead of having two colours for each Y (4:4:4),
we can use two colours for each 4 Y’s (4:2:0) or 4 colurs for
each Y (4:2:2).
Slide 21
Lecture 2:
SDI Interface
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• Serial Digital Interface is a standard developed by The Society
of Motion Picture and Television Engineers (SMPTE) in 1989
(SMPTE 259M). It is used for transferring uncompressed video.
It uses BNC connector. Later high-definition SDI (HD-SDI), was
standardized in SMPTE 292M; this provides a nominal data rate
of 1.5 (1.485) Gbit/s. It is good for transferring a single 4:2:2
video. Later 3G-SDI was developed that can carry a 1080p
50/60 or 1080p 4:4:4.
• 6G-SDI and 12G-SDI have 6 and
12 Gb/s rate, respectively and can
be used to transport 4K video.
Slide 22
Lecture 2:
HDMI Interface
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• HDMI implements the EIA/CEA-861 standards, which define
video formats and waveforms, transport of compressed,
uncompressed, and LPCM audio and auxiliary data. HDMI 2.0
released in 2013 has the maximum bitrate of 6 Gbit/s for each
channel and a total throughput of 18 Gbit/s. This allows HDMI
2.0 to carry 4K resolution at 60 frames per second (fps).
Slide 23
Lecture 2:
ASI
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• While SD-SDI (270 Mbit/s) or HD-SDI (1.485 Gbit/s) carry uncompressed video
an, an ASI (Asynchronous Serial Interface ) signal can carry one or multiple SD,
HD or audio programs that are already compressed. ASI, is a streaming data
format which often carries an MPEG Transport Stream (MPEG-TS). ASI signal
can be at varying transmission speeds and is completely dependent on the
user's engineering requirements. For example, an ATSC has a maximum
bandwidth of 19.392658 Mbit/s. Generally, the ASI signal is the final product of
video compression, either MPEG2 or MPEG4, ready for transmission to a
transmitter or microwave system or other device. Sometimes it is also
converted to fiber, RF or SMPTE310 for other types of transmission. There are
two transmission formats commonly used by the ASI interface: the 188 byte
format and the 204 byte format. The 188 byte format is the more common ASI
transport stream. When optional Reed–Solomon error correction data are
included, the packet can stretch an extra 16 bytes to 204 bytes total.
Slide 24
Lecture 2:
JPEG
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• The goal of video compression techniques is to reduce the bit rate of
the video while keeping the quality as high as possible. The reduction
in bit rate (size of the video file) is achieved by removing the
redundancy in the video signal. Being a three dimensional signal, a
video contains spatial and temporal information and hence spatial
and temporal redundancy. The spatial redundancy is the redundant
content in a frame (intra-frame redundancy) while the temporal
redundancy is the similarity between the consecutive frames (inter-
frame redundancy). A frame full of different objects of varying size
and colour has hardly any redundancy and cannot be compressed
a lot while a quiet frame has lots of redundancy and can be
compressed considerably. Similarly a slowly varying scene, say, a
broadcaster reading the news has hardly any inter-frame variation
and there is a lot of redundancy to be removed hence high
compression ratio. On the other hand in a football match there is a
lot changing between two frames and therefore, we need more bits
to represent the video.
Slide 25
Lecture 2:
JPEG
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• Removing the spatial redundancy is done using transform coding (in
order to find the prominent spectral lines of a frame) and some
statistical techniques such as Huffman Coding in order to assign less
bits to more probable values.
• Removal of the temporal redundancy is through utilization of the
similarity between consecutive frames by sending information in the
form of motion vectors just enough for the decoder to be able to
reconstruct the frame based on the older (fresh or reconstructed)
frames and the incremental data contained in the motion vectors.
• We start the discussion with the spatial compression, i.e., encoding a
frame and will later discuss about inter-frame compression. As a
frame is just a picture, we start with image compression technique
JPEG which is basically the same technique used in different versions
of MPEG.
Slide 26
Lecture 2:
JPEG
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• JPEG was developed by the Joint Photographic Experts Group.
• JPEG uses a lossy compression technique based on the Discrete
Cosine Transform (DCT). DCT converts each frame/field of the video
source from the spatial (2D) domain into the frequency (transform)
domain A perceptual model based loosely on the human
psychovisual system discards high-frequency information, i.e. sharp
transitions in intensity, and colour hue. In the transform domainthe
picture is quantized by discarding or highly compressing the high-
frequency coefficients, which contribute less to the overall picture
than other coefficients and are also characteristically small-values
with high compressibility. The quantized coefficients are then
sequenced and losslessly packed into the output bitstream.
• Lossless coding of transform domain coefficients is done using
Huffman coding.
Slide 27
Lecture 2:
JPEG: Zigzag scan
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• In JPEG, in order to preserve correlation between pixels, zigzag scan
is used on an 8-by-8 group of pixels:
Slide 28
Lecture 2:
JPEG: DCT
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• JPEG was developed by the Joint Photographic Experts Group.
DCT Basis Functions
Slide 29
Lecture 2:
JPEG: DCT
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• Examples of DCT output:
Slide 30
Lecture 2:
JPEG: Entropy Coding
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• The average length of a data stream consisting of a characters
taking value from a given alphabet can be reduced by assigning
shorter representation (less bits) to more frequently appearing
characters. For example in compressing a written text, say in English,
instead of assigning the same number of bits to each letter, we may
assign less bits to more frequent letters such as t or e and more bits to
q or z.
• In order for a code to be useful, it has to be prefix-free, i.e., no
codeword can be prefix for another. These are called prefix codes.
• The optimum prefix codes are designed based on the following rules:
• For a source with alphabet having probabilities , , …
– If then and are the lengths of symbols j and k.
– The two longest codewords (corresponding to the least probable
symbols) have the same length.
– The two longest codewords differ only in the last bit.
• These rules provides us a tool for designing optimal (Huffman) Codes.
Slide 31
Lecture 2:
JPEG: Entropy Coding
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• Morse Code: Note that E and T are represented by shortest strings.
Slide 32
Lecture 2:
JPEG: Huffman Coding
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• The average length of a data stream consisting of characters taking
value from a given alphabet can be reduced by assigning shorter
representation (less bits) to more frequently appearing characters.
For example in compressing a written text, say in English, instead of
assigning the same number of bits to each letter, we may assign less
bits to more frequent letters such as t or e and more bits to q or z.
• In order for a code to be useful, it has to be prefix-free, i.e., no
codeword can be prefix for another. These are called prefix codes.
• The optimum prefix codes are designed based on the following rules:
• For a source with alphabet having probabilities , , …
– If then and are the lengths of symbols j and k.
– The two longest codewords (corresponding to the least probable
symbols) have the same length.
– The two longest codewords differ only in the last bit.
• These rules provides us a tool for designing optimal (Huffman) Codes.
Slide 33
Lecture 2:
JPEG: Huffman Coding
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• Assume that we have a source with m letters with probabilities
, ,…, . Assume we have a code for it with codeword lengths , , … , .
The average length of this code is ∑ .
• Now assume that we combine the two least likely symbols into one and
create a source with m-1 letters:
1, 2, … , 2
′
1
• Assume that we have an optimum code for this new source with m-1
symbols. Let’s assign the first m-2 codewords of this code to the first
m-2 symbols of the original source and expand the m-1st codeword
into two codewords with lengths of ′ 1 for symbols m-1 and m-2.
Then:
′ 1 1
=∑ ′
Slide 34
Lecture 2:
JPEG: Huffman Coding
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• We observe that the average length of the code for m symbols is the
average length of the code for m-1 symbols plus a constant. So, in
order to minimize we need to minimize . We can use this fact
recursively to get smaller and smaller alphabet.
• Consider for example a source with letters A, B, …, G with
probabilities {3/8, 3/16, 3/16, 1/8, 1/16, 1/32, 1/32}.
• L=2.44 bits/symbol.
• H(X)= 2.37 bits/symbol
Slide 35