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WELCOME
GUIDE:

MIS.AMBIKA SEKHAR
                    GROUP MEMBERS:
                            ATHIRA.P
                            SIRAJ SIDHIK
                            SHAHANA.P.N
PROBLEMS
 speech coding systems is to transmit speech with the
  highest possible quality using the least possible
  channel capacity.
 To save bandwidth in telecoms applications and to
  reduce memory storage requirements.
 Maintain certain levels of complexity to reduce the
  processing delay and cost of implementation.
PRESENTATION OUTLINE
 Section I
 Introduction to speech
 Sub-band coding (SBC)
 Filter Banks
 Section II
 Sub band coder implimentation
 QMF design
 Simulation and result
 Section III
 Conclusion
 Applications
Introduction to Speech
What is the Speech?
o Speech is the primary method of human
    communication.

o        To transmit/store a speech waveform using as
      few bits as possible while retaining high quality
Speech Process
 Production
 Propagation:



 Perception:
  The incoming sounds are deciphered by the listener
 into a received message, thereby completing the chain
 of events that culminated in the transfer of
 information from the speaker to the listener
SUB BAND CODING
  Divides the speech signal into many smaller sub-bands
  and encodes each sub-band separately according to
  some perceptual significance.

  Speech is typically divided into 4 or 8 sub-bands by a
  bank of filters.

  Can be used for coding speech at bit rates in the range
  9.6 kbps to 32 kbps.
 A compression approach where digital filters
 are used to separate the source output into
 different bands of frequencies.


 Each part then can be encoded separately.
FILTERS
 A system that isolates a constituent part corresponding to
 certain frequency is called a filter.

  If it isolates the low frequency components, it is called a
    low- pass filter.

 Similarly, we have high-pass or band –pass filters.

  In general, a filter can be called a subband filter if it isolates
   a number of bands
FILTER BANKS
 Filter banks are essentially a cascade of stages, where each
  stage consists of a low-pass filter and a high-pass filter
 The source output is passed through a bank of filters.
 This filter bank covers the range of frequencies that make
  up the source output.
 The passband of each filter specifies each set of
  frequencies that can pass through.
FILTER BANKS
SUB BAND CODER
 IMPLIMENTATION
MATLAB CODE IMPLIMENTING THE SUBBAND CODER

Function y=subband(x,h0,bits)
           subband decomposition
           y=subband(x, h0, [bits])
                   x=input signal vector
                    h0=basic QMF filter
                     bits= a vector of 2 entries giving the number of bits
                     y=output signal vector
SUB BAND CODING ALGORITHM
1.ANALYSIS
BLOCK DIAGRAM OF A SUB BAND SPEECH ENCODER WITH
THREE FREQUENCY SUBDIVISION
 The speech signal is to be sampled at a rate fs samples
  per second.
 The first frequency subdivision is splits the signal
  spectrum into two equal width segments,low pass
  signal             and a high pass signal
 The second frequency subdivision split the first
  lowpass signal into two equal bands ,a lowpass
  signal          ,,,and a highpass signal
 Finally, the third frequency subdivision splits the
  lowpass signal from the second stage into two equal
  bandwidth signals .
 Thus the signal is subdivided into four frequency
  bands,covering three octaves.
BLOCK DIAGRAM OF SUB BAND SPEECH DECODER WITH THREE
FREQUENCY SUBDIVISION
 The decoding process for the sub band encoded
speech signal is basically the reverse of the encoding
process.
The signal in adjacent lowpass and high pass
frequency bands are interpolated, filterd,and
combined
Quadrature Mirror Filter (QMF)
A quadrature mirror filter is a filter most commonly used to
implement a filter bank that splits an input signal into two
bands. The resulting high-pass and low-pass signals are often
reduced by a factor of 2, giving a critically sampled two-
channel representation of the original signal.
DECIMATION




 Downsampling (or "subsampling") is the process of redusing
 the sampling rate of asignal. This is usually done to reduce
 the data rate or the size of the data.
INTERPOLATOR



 Upsampling is the process of increesing the sampling rate
  of a signal.
 The upsampling factor (commonly denoted by L) is
  usually an integer or a rational fraction greater than unity.
2.Quantization and Coding
Selection of the compression scheme
Allocation of bits between the subbands
Allocate the available bits among the subbands
 according to measure of the information content in
 each subband.
Bit Allocation
Minimizing the distortion i.e. minimizing the
 reconstruction error drives the bit allocation
 procedure.
Bit allocation procedure can have a significant
 impact on the quality of the final reconstruction
3.Synthesis
 Quantized and Coded coefficients are used to reconstruct a
  representation of the original signal at the decoder.
 Encoded samples from each subband decoded
  upsampled  bank of reconstruction filters outputs
  combined  Final reconstructed output
SIMULATION AND RESULTS
CONCLUSION
 Subband coding is another approach to decompose the
  source output into components based on frequency.
 A structure of two channel QMF with lowpass
  filter,highpass filter,decimators and interpolators has
  been proposed to perform subband coding of speech
  signal in the digital domain.
 The general subband encoding procedure can be summarized
  as follows:
• Select a set of filters for decomposing the source.
• Using the filters, obtain the subband signals.
• Decimate the output of the filters.
• Encode the decimated output.

 The decoding procedure is the inverse of the encoding
  procedure
APPLICATIONS
    Speech Coding
    ITU-T G.722

    Encode high quality speech at 64/56/48 kbps

    Audio Coding

    MPEG audio

    Image Compression
REFERENCES
 YUE Dongjian “The Study of Speech Coding Technology
  Based on Code Excited Linear Predictive Coding”
  Ph.D.thesis, Tongji University, 2000.
 B. Carnero and A. Drygajlo. “Perceptual speech coding
  and enhancement using frame synchronized fast wavelet
  packet transform algorithms.” IEEE Trans. Signal
  Processing Vol.47 No.6 ,June 1999.
 P. Philippe, F. Moreau de Saint-Martin and M. Lever.
“Wavelet packet filterbanks for low time delay audio
coding.” IEEE Trans. Speech and Audio Processing. 1999.
 John G. Proakis and Dimitris G. Manolakis, “Digital
    Signal Processing: Principles,Algorithms and
    Applications”, Third Edition.
   Roberts R. A. and Mullis C. T. Digital Signal Processing.
    Addison-Wesley, Reading.
   Mass, 2006.
   [3]. Oppenheim A. V. and Schafer R. W. Discrete-Time
    Signal Processing. Prentice Hall.
   Englewood Cliffs, New Jersey, 2007.
QUESTIONS…???

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Sub band project

  • 2. GUIDE: MIS.AMBIKA SEKHAR GROUP MEMBERS: ATHIRA.P SIRAJ SIDHIK SHAHANA.P.N
  • 3. PROBLEMS  speech coding systems is to transmit speech with the highest possible quality using the least possible channel capacity.  To save bandwidth in telecoms applications and to reduce memory storage requirements.  Maintain certain levels of complexity to reduce the processing delay and cost of implementation.
  • 4. PRESENTATION OUTLINE  Section I  Introduction to speech  Sub-band coding (SBC)  Filter Banks  Section II  Sub band coder implimentation  QMF design  Simulation and result  Section III  Conclusion  Applications
  • 5. Introduction to Speech What is the Speech? o Speech is the primary method of human communication. o To transmit/store a speech waveform using as few bits as possible while retaining high quality
  • 6. Speech Process  Production  Propagation:  Perception: The incoming sounds are deciphered by the listener into a received message, thereby completing the chain of events that culminated in the transfer of information from the speaker to the listener
  • 7. SUB BAND CODING  Divides the speech signal into many smaller sub-bands and encodes each sub-band separately according to some perceptual significance.  Speech is typically divided into 4 or 8 sub-bands by a bank of filters.  Can be used for coding speech at bit rates in the range 9.6 kbps to 32 kbps.
  • 8.  A compression approach where digital filters are used to separate the source output into different bands of frequencies.  Each part then can be encoded separately.
  • 9. FILTERS A system that isolates a constituent part corresponding to certain frequency is called a filter.  If it isolates the low frequency components, it is called a low- pass filter. Similarly, we have high-pass or band –pass filters.  In general, a filter can be called a subband filter if it isolates a number of bands
  • 10. FILTER BANKS  Filter banks are essentially a cascade of stages, where each stage consists of a low-pass filter and a high-pass filter  The source output is passed through a bank of filters.  This filter bank covers the range of frequencies that make up the source output.  The passband of each filter specifies each set of frequencies that can pass through.
  • 12. SUB BAND CODER IMPLIMENTATION
  • 13. MATLAB CODE IMPLIMENTING THE SUBBAND CODER Function y=subband(x,h0,bits) subband decomposition y=subband(x, h0, [bits]) x=input signal vector h0=basic QMF filter bits= a vector of 2 entries giving the number of bits y=output signal vector
  • 14. SUB BAND CODING ALGORITHM
  • 15. 1.ANALYSIS BLOCK DIAGRAM OF A SUB BAND SPEECH ENCODER WITH THREE FREQUENCY SUBDIVISION
  • 16.  The speech signal is to be sampled at a rate fs samples per second.  The first frequency subdivision is splits the signal spectrum into two equal width segments,low pass signal and a high pass signal  The second frequency subdivision split the first lowpass signal into two equal bands ,a lowpass signal ,,,and a highpass signal
  • 17.  Finally, the third frequency subdivision splits the lowpass signal from the second stage into two equal bandwidth signals .  Thus the signal is subdivided into four frequency bands,covering three octaves.
  • 18. BLOCK DIAGRAM OF SUB BAND SPEECH DECODER WITH THREE FREQUENCY SUBDIVISION
  • 19.  The decoding process for the sub band encoded speech signal is basically the reverse of the encoding process. The signal in adjacent lowpass and high pass frequency bands are interpolated, filterd,and combined
  • 20. Quadrature Mirror Filter (QMF) A quadrature mirror filter is a filter most commonly used to implement a filter bank that splits an input signal into two bands. The resulting high-pass and low-pass signals are often reduced by a factor of 2, giving a critically sampled two- channel representation of the original signal.
  • 21.
  • 22.
  • 23. DECIMATION Downsampling (or "subsampling") is the process of redusing the sampling rate of asignal. This is usually done to reduce the data rate or the size of the data.
  • 24. INTERPOLATOR  Upsampling is the process of increesing the sampling rate of a signal.  The upsampling factor (commonly denoted by L) is usually an integer or a rational fraction greater than unity.
  • 25. 2.Quantization and Coding Selection of the compression scheme Allocation of bits between the subbands Allocate the available bits among the subbands according to measure of the information content in each subband.
  • 26. Bit Allocation Minimizing the distortion i.e. minimizing the reconstruction error drives the bit allocation procedure. Bit allocation procedure can have a significant impact on the quality of the final reconstruction
  • 27. 3.Synthesis  Quantized and Coded coefficients are used to reconstruct a representation of the original signal at the decoder.  Encoded samples from each subband decoded upsampled  bank of reconstruction filters outputs combined  Final reconstructed output
  • 29. CONCLUSION  Subband coding is another approach to decompose the source output into components based on frequency.  A structure of two channel QMF with lowpass filter,highpass filter,decimators and interpolators has been proposed to perform subband coding of speech signal in the digital domain.
  • 30.  The general subband encoding procedure can be summarized as follows: • Select a set of filters for decomposing the source. • Using the filters, obtain the subband signals. • Decimate the output of the filters. • Encode the decimated output.  The decoding procedure is the inverse of the encoding procedure
  • 31. APPLICATIONS  Speech Coding  ITU-T G.722  Encode high quality speech at 64/56/48 kbps  Audio Coding  MPEG audio  Image Compression
  • 32. REFERENCES  YUE Dongjian “The Study of Speech Coding Technology Based on Code Excited Linear Predictive Coding” Ph.D.thesis, Tongji University, 2000.  B. Carnero and A. Drygajlo. “Perceptual speech coding and enhancement using frame synchronized fast wavelet packet transform algorithms.” IEEE Trans. Signal Processing Vol.47 No.6 ,June 1999.  P. Philippe, F. Moreau de Saint-Martin and M. Lever. “Wavelet packet filterbanks for low time delay audio coding.” IEEE Trans. Speech and Audio Processing. 1999.
  • 33.  John G. Proakis and Dimitris G. Manolakis, “Digital Signal Processing: Principles,Algorithms and Applications”, Third Edition.  Roberts R. A. and Mullis C. T. Digital Signal Processing. Addison-Wesley, Reading.  Mass, 2006.  [3]. Oppenheim A. V. and Schafer R. W. Discrete-Time Signal Processing. Prentice Hall.  Englewood Cliffs, New Jersey, 2007.
  • 34.