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Voice Quality Enhancement

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Voice Quality Enhancement. Shi Chao Zhang, Kevin McCook, & Kurtis Chang. T.A.: Matt Olson ... Build a hands-free speakerphone system. Apply Digital Signal Processing ... – PowerPoint PPT presentation

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Title: Voice Quality Enhancement


1
Voice Quality Enhancement
  • Shi Chao Zhang, Kevin McCook,
  • Kurtis Chang
  • T.A. Matt Olson

2
Objective
  • Build a hands-free speakerphone system
  • Apply Digital Signal Processing
  • Remove acoustic echo from transmission
  • Adaptive Echo Cancellation, Echo Suppression
  • Reduce noise
  • Frequency Domain Spectral Subtraction

3
Design Review
Incoming Signal
Audio Amp.
Computer 1
Input 1
Spectral Subtraction
Speaker
DSP
Output
DSP
Input 2
Computer 2
Microphone Preamp.
mic
Outgoing Signals
4
Digital Signal Processing
  • Texas Instruments TMS320C54X DSP
  • Sampling rate 44.1 kHz
  • Frequency response 0 22.05 kHz
  • Voice Signals Are Bandlimited
  • 300 to 3400 Hz
  • Implement multi-rate processing
  • Increases available processing time
  • Improves efficiency of algorithms.

5
Decimation/Interpolation
  • Changes Sampling Rate of DSP

Processing
6
Echo Canceling
  • Input 1 Incoming Signal
  • Input 2 Signal received by microphone

?
AFIR
LPF
D ?
Input 1

LPF
D ?
Input 2
Output
7
Adaptive Filter Based E.C.
  • Filter converges to transfer function
  • Uses LMS algorithm to update coefficients
  • Uses error to calculate
  • Parameter Step Size affects convergence rate
  • Removes Unwanted Signal Digitally

Unknown Transfer Function
8
Adaptive Filter Design
  • Had to choose a step size
  • 0.06 was ideal step size
  • Had to determine length
  • Longer Adaptive Filter could remove longer
    lasting echos
  • 510 was longest possible length

9
Noise Reduction
  • Conceptual overview
  • Determination of noise for each frequency
  • Subtraction of lowest noise in time period from
    each frequency

10
Spectral Subtraction
  • Algorithmic overview

11
Spectral Subtraction
  • MatLab simulation results

Sample 1
Sample 2
12
Spectral Subtraction
  • DSP simulation
  • FFT and IFFT reduce precision
  • 1024-point FFT-IFFT pair

Sample DSP output
13
Echo Suppression Overview
  • eliminates near-end signal when no double-talk is
    detected
  • uses voice-activity detector to determine
    double-talk state
  • uses comfort noise generation for when there is
    no near-end signal

14
Voice-Activity Detector
  • -uses recursive power algorithm to compare power
    level vs. double-talk
  • threshold
  • -P(i) (1-alpha) P(i-1) alpha input(i)
  • where alpha time constant
  • -uses holdover threshold to prevent clipping of
    speech signals

15
comfort noise generator
  • uses a pseudo-noise generator to produce the
    'random' bits
  • -using a shift register of size N, bits will
    repeat every 2N - 1 times

16
echo suppression initial simulations
  • through research, decided to implement recursive
    power algorithm
  • used matlab to produce some preliminary test
    results and then adjust variables

17
echo suppression dsp implementation and testing
  • after matlab testing, implemented voice activity
    detector in C and
  • comfort noise generation in assembly
  • then, had to adjust all variables to perform well
    with our speakerphone
  • overall, the echo suppression worked successfully

18
Functional Tests
  • Made recordings of unprocessed signal and
    processed signal simultaneously, then compared
    outputs
  • Matched the gain of the microphone and speaker
    signals

19
Circuit Diagram
  • Microphone Preamp

Used integrated circuit for audio amp Motorola
MC34119P
20
Successes
  • Substantial reduction of echo noise through the
    use of an adaptive filter
  • Successful determination of non-speech periods
    through echo suppression

21
Challenges
  • A suitable rate of change for adaptive
    coefficients has to be determined ahead of time
  • The number of coefficients is limited by the
    processing power of the DSP
  • Noise reduction (spectral subtraction) was
    impractical to implement on the DSP due to
    reduction of precision in the FFT algorithm used

22
Recommendations
  • Use Infinite Impulse Response Filters for
    anti-aliasing
  • Uses less memory
  • Use adaptive filter for comfort noise generator
  • Scales noise added to the value of noise
    calculated in spectral subtraction

23
Conclusion
  • The majority of the project worked at the end
  • Had problems converting matlab code to assembly
    for Noise Reduction

24
  • THANK YOU!!
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