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Audio Compression
Techniques
  Lecture 8


              Prepared by
              Razia Nisar Noorani

                                    1
Introduction
   Digital Audio Compression
     Removal   of redundant or otherwise irrelevant
      information from audio signal
     Audio compression algorithms are often referred to as
      “audio encoders”
   Applications
     Reduces required storage space
     Reduces required transmission bandwidth




                                                          2
Audio Compression
   Audio signal – overview
     Sampling   rate (# of samples per second)
     Bit rate (# of bits per second). Typically,
      uncompressed stereo 16-bit 44.1KHz signal has a
      1.4MBps bit rate
     Number of channels (mono / stereo / multichannel)
   Reduction by lowering those values or by data
    compression / encoding



                                                          3
Audio Data Compression
   Redundant information
     Implicit
             in the remaining information
     Ex. oversampled audio signal
          oversampling is the process of sampling a signal with a
           sampling frequency significantly higher than twice the
           bandwidth or highest frequency of the signal being sampled
   Irrelevant information
     Perceptuallyinsignificant
     Cannot be recovered from remaining information



                                                                        4
Audio Data Compression
   Lossless Audio Compression
     Removes   redundant data
     Resulting signal is same as original – perfect
      reconstruction
   Lossy Audio Encoding
     Removes   irrelevant data
     Resulting signal is similar to original


                                                       5
Audio Data Compression
   Audio vs. Speech Compression
    Techniques
     Speech  Compression uses a human vocal
      tract model to compress signals
     Audio Compression does not use this
      technique due to larger variety of possible
      signal variations


                                                    6
Generic Audio Encoder
   Psychoacoustic Model
     Psychoacoustics – study of how sounds are
      perceived by humans
     Uses perceptual coding
         eliminate information from audio signal that is
          inaudible to the ear
     Detectsconditions under which different audio
     signal components mask each other

                                                            7
Psychoacoustic Model
   Signal Masking
     Threshold  cut-off
     Spectral (Frequency / Simultaneous) Masking
     Temporal Masking
   Threshold cut-off and spectral masking
    occur in frequency domain, temporal
    masking occurs in time domain

                                                8
Signal Masking
   Threshold cut-off
     Hearing  threshold
      level – a function of
      frequency
     Any frequency
      components below the
      threshold will not be
      perceived by human
      ear


                              9
Signal Masking
   Spectral Masking
    A   frequency
      component can be
      partly or fully masked
      by another component
      that is close to it in
      frequency
     This shifts the hearing
      threshold


                                10
Signal Masking
   Temporal Masking
    A  quieter sound can
      be masked by a louder
      sound if they are
      temporally close
     Sounds that occur
      both (shortly) before
      and after volume
      increase can be
      masked


                              11
Spectral Analysis
   a device or algorithm that identifies a
    frequency domain representation of a
    time domain signal.
   Tasks of Spectral Analysis
     To derive masking thresholds to determine which
      signal components can be eliminated
     To generate a representation of the signal to which
      masking thresholds can be applied
   Spectral Analysis is done through transforms or
    filter banks
                                                            12
Spectral Analysis
   Transforms
     Fast Fourier Transform (FFT)
     Discrete Cosine Transform (DCT) - similar to
      FFT but uses cosine values only
     Modified Discrete Cosine Transform (MDCT)
      [used by MPEG-1 Layer-III, MPEG-2 AAC,
      Dolby AC-3] – overlapped and windowed
      version of DCT


                                                     13
Spectral Analysis
   Filter Banks
   a filter bank is an array of band-pass filters that
    separates the input signal into multiple
    components, each one carrying a single
    frequency subband of the original signal
     Time  sample blocks are passed through a set of
      bandpass filters
     Masking thresholds are applied to resulting frequency
      subband signals
     Poly-phase and wavelet banks are most popular filter
      structures                                          14
Filter Bank Structures
   Polyphase Filter Bank
    [used in all of the MPEG-1 encoders]
     Signal is separated into subbands, the widths
      of which are equal over the entire frequency
      range
     The resulting subband signals are
      downsampled to create shorter signals (which
      are later reconstructed during decoding
      process)

                                                  15
Filter Bank Structures
   Wavelet Filter Bank
    [used by Enhanced Perceptual Audio
    Coder (EPAC) by Lucent]
     Unlike  polyphase filter, the widths of the
      subbands are not evenly spaced (narrower for
      higher frequencies)
     This allows for better time resolution (ex. short
      attacks), but at expense of frequency
      resolution

                                                     16
Noise Allocation
   System Task: derive and apply shifted hearing
    threshold to the input signal
     Anything  below the threshold doesn’t need to be
      transmitted
     Any noise below the threshold is irrelevant
   Frequency component quantization
     Tradeoff between space and noise
     Encoder saves on space by using just enough bits for
      each frequency component to keep noise under the
      threshold - this is known as noise allocation

                                                         17
Noise Allocation
   Pre-echo
     In case a single audio block contains silence followed
      by a loud attack, pre-echo error occurs - there will be
      audible noise in the silent part of the block after
      decoding
     This is avoided by pre-monitoring audio data at
      encoding stage and separating audio into shorter
      blocks in potential pre-echo case
     This does not completely eliminate pre-echo, but can
      make it short enough to be masked by the attack
      (temporal masking)

                                                            18
Additional Encoding Techniques
   Other encoding techniques techniques are
    available (alternative or in combination)
     Predictive Coding
     Coupling / Delta Encoding
     Huffman Encoding




                                            19
Additional Encoding Techniques
   Predictive Coding
     Often used in speech and image compression
     Estimates the expected value for each sample based
      on previous sample values
     Transmits/stores the difference between the expected
      and received value
     Generates an estimate for the next sample and then
      adjusts it by the difference stored for the current
      sample
     Used for additional compression in MPEG2 AAC
      (Advance audio Coding)
                                                        20
Additional Encoding Techniques
   Coupling / Delta encoding
     Used  in cases where audio signal consists of two or
      more channels (stereo or surround sound)
     Similarities between channels are used for
      compression
     A sum and difference between two channels are
      derived; difference is usually some value close to zero
      and therefore requires less space to encode
     This is a case of lossless encoding process



                                                           21
Additional Encoding Techniques
   Huffman Coding
     Information-theory-based   technique
     An element of a signal that often reoccurs in the
      signal is represented by a simpler symbol, and its
      value is stored in a look-up table
     Implemented using a look-up tables in encoder and in
      decoder
     Provides substantial lossless compression, but
      requires high computational power and therefore is
      not very popular
     Used by MPEG1 and MPEG2 AAC

                                                         22
Encoding - Final Stages
 Audio data packed into frames
 Frames stored or transmitted




                                  23
Questions



            24

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Lecture 8 audio compression

  • 1. Audio Compression Techniques Lecture 8 Prepared by Razia Nisar Noorani 1
  • 2. Introduction  Digital Audio Compression  Removal of redundant or otherwise irrelevant information from audio signal  Audio compression algorithms are often referred to as “audio encoders”  Applications  Reduces required storage space  Reduces required transmission bandwidth 2
  • 3. Audio Compression  Audio signal – overview  Sampling rate (# of samples per second)  Bit rate (# of bits per second). Typically, uncompressed stereo 16-bit 44.1KHz signal has a 1.4MBps bit rate  Number of channels (mono / stereo / multichannel)  Reduction by lowering those values or by data compression / encoding 3
  • 4. Audio Data Compression  Redundant information  Implicit in the remaining information  Ex. oversampled audio signal  oversampling is the process of sampling a signal with a sampling frequency significantly higher than twice the bandwidth or highest frequency of the signal being sampled  Irrelevant information  Perceptuallyinsignificant  Cannot be recovered from remaining information 4
  • 5. Audio Data Compression  Lossless Audio Compression  Removes redundant data  Resulting signal is same as original – perfect reconstruction  Lossy Audio Encoding  Removes irrelevant data  Resulting signal is similar to original 5
  • 6. Audio Data Compression  Audio vs. Speech Compression Techniques  Speech Compression uses a human vocal tract model to compress signals  Audio Compression does not use this technique due to larger variety of possible signal variations 6
  • 7. Generic Audio Encoder  Psychoacoustic Model  Psychoacoustics – study of how sounds are perceived by humans  Uses perceptual coding  eliminate information from audio signal that is inaudible to the ear  Detectsconditions under which different audio signal components mask each other 7
  • 8. Psychoacoustic Model  Signal Masking  Threshold cut-off  Spectral (Frequency / Simultaneous) Masking  Temporal Masking  Threshold cut-off and spectral masking occur in frequency domain, temporal masking occurs in time domain 8
  • 9. Signal Masking  Threshold cut-off  Hearing threshold level – a function of frequency  Any frequency components below the threshold will not be perceived by human ear 9
  • 10. Signal Masking  Spectral Masking A frequency component can be partly or fully masked by another component that is close to it in frequency  This shifts the hearing threshold 10
  • 11. Signal Masking  Temporal Masking A quieter sound can be masked by a louder sound if they are temporally close  Sounds that occur both (shortly) before and after volume increase can be masked 11
  • 12. Spectral Analysis  a device or algorithm that identifies a frequency domain representation of a time domain signal.  Tasks of Spectral Analysis  To derive masking thresholds to determine which signal components can be eliminated  To generate a representation of the signal to which masking thresholds can be applied  Spectral Analysis is done through transforms or filter banks 12
  • 13. Spectral Analysis  Transforms  Fast Fourier Transform (FFT)  Discrete Cosine Transform (DCT) - similar to FFT but uses cosine values only  Modified Discrete Cosine Transform (MDCT) [used by MPEG-1 Layer-III, MPEG-2 AAC, Dolby AC-3] – overlapped and windowed version of DCT 13
  • 14. Spectral Analysis  Filter Banks  a filter bank is an array of band-pass filters that separates the input signal into multiple components, each one carrying a single frequency subband of the original signal  Time sample blocks are passed through a set of bandpass filters  Masking thresholds are applied to resulting frequency subband signals  Poly-phase and wavelet banks are most popular filter structures 14
  • 15. Filter Bank Structures  Polyphase Filter Bank [used in all of the MPEG-1 encoders]  Signal is separated into subbands, the widths of which are equal over the entire frequency range  The resulting subband signals are downsampled to create shorter signals (which are later reconstructed during decoding process) 15
  • 16. Filter Bank Structures  Wavelet Filter Bank [used by Enhanced Perceptual Audio Coder (EPAC) by Lucent]  Unlike polyphase filter, the widths of the subbands are not evenly spaced (narrower for higher frequencies)  This allows for better time resolution (ex. short attacks), but at expense of frequency resolution 16
  • 17. Noise Allocation  System Task: derive and apply shifted hearing threshold to the input signal  Anything below the threshold doesn’t need to be transmitted  Any noise below the threshold is irrelevant  Frequency component quantization  Tradeoff between space and noise  Encoder saves on space by using just enough bits for each frequency component to keep noise under the threshold - this is known as noise allocation 17
  • 18. Noise Allocation  Pre-echo  In case a single audio block contains silence followed by a loud attack, pre-echo error occurs - there will be audible noise in the silent part of the block after decoding  This is avoided by pre-monitoring audio data at encoding stage and separating audio into shorter blocks in potential pre-echo case  This does not completely eliminate pre-echo, but can make it short enough to be masked by the attack (temporal masking) 18
  • 19. Additional Encoding Techniques  Other encoding techniques techniques are available (alternative or in combination)  Predictive Coding  Coupling / Delta Encoding  Huffman Encoding 19
  • 20. Additional Encoding Techniques  Predictive Coding  Often used in speech and image compression  Estimates the expected value for each sample based on previous sample values  Transmits/stores the difference between the expected and received value  Generates an estimate for the next sample and then adjusts it by the difference stored for the current sample  Used for additional compression in MPEG2 AAC (Advance audio Coding) 20
  • 21. Additional Encoding Techniques  Coupling / Delta encoding  Used in cases where audio signal consists of two or more channels (stereo or surround sound)  Similarities between channels are used for compression  A sum and difference between two channels are derived; difference is usually some value close to zero and therefore requires less space to encode  This is a case of lossless encoding process 21
  • 22. Additional Encoding Techniques  Huffman Coding  Information-theory-based technique  An element of a signal that often reoccurs in the signal is represented by a simpler symbol, and its value is stored in a look-up table  Implemented using a look-up tables in encoder and in decoder  Provides substantial lossless compression, but requires high computational power and therefore is not very popular  Used by MPEG1 and MPEG2 AAC 22
  • 23. Encoding - Final Stages  Audio data packed into frames  Frames stored or transmitted 23
  • 24. Questions 24

Editor's Notes

  1. Hello, Today I will talk about the common techniques commonly used for digital audio compression of various audio filetype formats.
  2. -I will discuss the difference between redundant and irrelevant further in my presentation. -Depending on storage or transmission, there is an optimization in size