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Speech/Audio Signal Processing in MATLAB/Simulink

Speech/Audio Signal Processing in MATLAB/Simulink

J. J.-S. Roger Jang ( ) ) CS Dept, TsingTsing-Hua Univ, Taiwan ( ) )
[Link] jang@[Link]

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Speech/Audio Signal Processing in MATLAB/Simulink

About Me
Experiences:
1993-1995: The MathWorks, Inc. 1995-now: CS Dept., Tsing Hua Univ., Taiwan

Research interests
Speech/Audio Signal Processing, Fuzzy Logic, Neural Networks, Pattern Recognition, Biometric Identification, Document Classification, Webbased Technologies

Programming languages:
MATLAB, C, JavaScript, VBScript, Perl

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Outline
Wave file manipulation
Reading, writing, recording ...

Time-domain processing
Delay, filtering, sptools

Frequency-domain processing
Spectrogram

Pitch determination
Auto-correlation, SIFT, AMDF, HPS ...

Others
Formant estimation, speech coding
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Toolbox/Blockset Used
MATLAB Simulink Signal Processing Toolbox DSP Blockset

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MATLAB Primer
Before you start, you need to get familiar with MATLAB. Please read MATLAB Primer at the following page: [Link] asp Exercise: 1. Please plot two curves y=sin(2*t) and y=cos(3*t) in the same figure. 2. Please plot x vs. y where x=sin(2*t) and y=cos(3*t).
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To Read a Wave File

To read a MS .wav file (PCM format only): wavread
y = wavread(file) [] = wavread(file, [n1, n2]) [y, fs, nbits, opts] = wavread(file) [] = wavread(file, n) [y, fs, nbits] = wavread(file)

If the wav file is stereo, y will be a two-column matrix.

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To Read a Wav File

Example (wavRead01.m):
[y, fs] = wavread('[Link]'); plot((1:length(y))/fs, y); xlabel('Time in seconds'); ylabel('Amplitude');

Exercise
1. Plot the waveform of [Link]. Use MATLABs zoom button to find the consecutive curling R occurs. 2. Plot the two-channel waveform in [Link].
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Solution to the Previous Exercise

wavRead02.m:
[y, fs] = wavread([Link]); subplot(2,1,1), plot((1:length(y))/fs, y(:,1)); subplot(2,1,2), plot((1:length(y))/fs, y(:,2));

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To Play Wav Files

To play sound using Windows audio output device: wavplay, sound, soundsc
wavplay(y, fs) wavplay(y, fs, async): non-blocking call wavplay(y, fs, sync): blocking call sound(y, fs) soundsc(): autoscale the sound

Example (wavPlay01.m)
[y, fs] = wavread([Link]); wavplay(y, fs);

Exercise
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Follow the example to play [Link].

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To Read/Play Using DSP Blocks

To read/play sound using DSP Blockset:
DSP Blockset/DSP Sources/From Wave File DSP Blockset/DSP Sinks/To Wave Device

Example:

Frame-based operation!

Exercise:
Create a model as shown above.
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Solution
Solution to the previous exercise: [Link]

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To Write a Wave File

To write MS wave files: wavwrite
wavwrite(y, fs, nbits, wavefile) nbits must be 8 or 16. y must have two columns for stereo data. Amplitude values outside [-1,1] are clipped.

Example (wavWrite01.m)
[y, fs] = wavread([Link]); wavwrite(y, fs*1.2, 8, [Link]); !start [Link]

Exercise
Try out the above example.
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To Record a Wave File

To record wave files:
1. Use the recording utility under WinXP. 2. Use wavrecord under MATLAB. 3. Use From Wave Device under Simulink, under DSP Blocksets/Platform Specific IO/Windows (Win32)

Example
1. Go ahead and try WinXP recording utility! 2. Try wavRecord01.m 3. Try [Link]

Exercise:
Try out the above examples.
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Time-Domain Speech Signals

A typical time-domain plot of speech signals:

Amplitude: volume or intensity Frequency: pitch

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Changing Wave Playback Param.

To control the play of a sound:
Normal: wavplay(y, fs) High volume: wavplay(2*y, fs) Low volume: wavplay(0.5*y, fs) High pitch (and faster): wavplay(y, 1.2*fs) Low pitch (and slower): wavplay(y, 0.8*fs)

Exercise:
Try wavPlay01.m and trace the code. Create wavPlay02.m such that you can record your own voice on the fly.

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Time-Domain Signal Processing

Take-home exrecise:
How to get a high pitch with the same time span?

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Synthetic Sounds
Use a sine wave generator (under DSP blocksets) to produce sounds
Single frequency:

Multiple frequencies:

Amplitude modulation:

Exercise:
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Create the above models.

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Solution
Solution to the previous exercise: sineSource01 sineSource02 sineSource03

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Delay in Speech/Audio
What is a delay in a signal?
y(n) --> y(n-k)

What effects can delay generate?

Echo Reverberation Chorus Flanging

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Single Delay in Audio Signal

Block diagram:
Input u(n)

-k

Output y(n) = u(n) + a*u(n-k)

Simulink model:

Exercise:
Create the above model.
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Multiple Delay in Audio Signal

How to create karaoke effects:
a
Input u(n)

-k

Output y(n)

y(n) = u(n) + a u(n-k) + a 2u(n-2k) + a 3u(n-3k) ...

Simulink model:

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Multiple Delay in Audio Signal

Parameter values:
Feedback gain a < 1 Actual delay time = k/fs

Exercise:
Create the above model and change some parameters to see their effects. Modify the model to take microphone input (so you can start singing karaoke now!) Use a configurable subsystem to include all possible input files and the microphone. (See next page.)

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Multiple Delay in Audio Signal

How to use configurable subsystem block?
1. Create a library (say, [Link])

2. Get a block of configurable subsystem 3. Fill the dialog box with the library name

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Audio Flanging
Flanging sound:
A sound similar to the sound of a jet plane flying overhead, or a "whooshing" sound Pitch modulation due to a variable delay

Simulink demo:
[Link] (all platforms) dspafxf_nt.mdl (for 95/98/NT)

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Audio Flanging
Simulink model:

Original spectrogram:

Modified spectrogram:

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Signal Processing Using sptool

To invoke sptool, type sptool.

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Speech Production
How is speech produced?
Speech is produced when air is forced from the lungs through the vocal cords (glottis) and along the vocal tract.

Analogy to System Theory:

Input: air forced into the vocal cords Output: media vibration System (or filter): vocal tract Pitch frequency: frequency of the input Formant frequency: resonant frequency
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Source Filter Model of Speech

The source-filter model of speech production:
Speech is split into a rapidly varying excitation signal and a slowly varying filter. The envelope of the power spectra contains the vocal tract information.

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Two important characteristics of the model are fundamental (pitch) frequency (f0) and formants 2010/8/26 (F1, F2, F3, ) 28

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Frame Analysis of Speech Signal

Speech wave form :

Zoom in

Overlap Frame

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Spectrogram
Spectrogram (specgram.m) displays short-time frequency contents:

Wave form :

Spectrogram :

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Real-time Spectrogram
Try dspstfft_win32:

Spectrum:

Spectrogram:

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Pitch and Formants

Pitch and formants can be defined visually:
First formant F1 Pitch period = 1/f0 Second formant F2

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Spectrogram Reading
Spectrogram Reading
[Link] ading/spectrogram_reading.html

Waveform:

Spectrogram:

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compute

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Pitch Determination Algorithms

Time-domain:
Auto-correlation AMDF (Average Magnitude Difference Function) Gold-Rabiner algorithm (1969)

Frequency-domain:
Cepstrum (Noll 1964) Harmonic product spectrum (Schroeder 1968)

Others:
SIFT (Simple inverse filter tracking) Maximum likelihood Neural network approach

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Autocorrelation of Each Frame

Let s(k) be a frame of size 128.
 

s(k): s(k-L):
L=30

x(30) = dot prod. of overlapped = sum(s(31:128).*s(1:99)

Autocorrelation x(L):
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Pitch period

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Autocorrelation via DSP Blockset

Real-time autocorrelation demo:

Exercise:
Construct the above model and try it.
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Pitch Tracking via Autocorrelation

Real-time pitch tracking via autocorrelation: [Link]

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Formant Analysis
Characteristics of formants:
Formants are perceptually defined. The corresponding physical property is the frequencies of resonances of the vocal tract. Formant analysis is useful as the position of the first two formants pretty much identifies a vowel.

Computation methods:

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Peak picking on the smoothed spectrum Peak picking on the LP spectrum Factoring for the LP roots Fitting of mixture of Gaussians
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Formant Analysis
Track Draw:
A package for formant synthesis with options to sketch formant tracks on a spectrogram. [Link] [Link]

Formant Location Algorithm

MATLAB code by Michelle Jamrozik [Link]

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Speech Waveform Coding

Time domain coding
PCM: Pulse Code Modulation DPCM: Differential PCM ADPCM: Adaptive Differential PCM ([Link])

Frequency domain coding

Sub-band coding Transform coding

Speech Coding in MATLAB

[Link] ml
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Conclusions
Ideal tools for speech/audio signal processing:
MATLAB Simulink Signal Processing Toolbox DSP Blockset Reliable functions: well-established and tested Visible graphical algorithm design tools High-level programming language yet C-compatible Powerful visualization capabilities

Advantages:

Easy debugging Integrated environment

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References
[1] Discrete-Time Processing of Speech Signals, by Deller, Proakis and Hansen, Prentice Hall, 1993 [2] Fundamentals of Speech Recognition, by Rabiner and Juang, Prentice Hall, 1993 [3] Effects Explained, [Link] [4] TrackDraw, [Link] [Link] [5] Speech Coding in MATLAB, [Link] html
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