Speech-Coding Techniques

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Speech-Coding Techniques

• Chapter 3

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Introduction

• Efficient speech-coding techniques
• Advantages for VoIP
• Digital streams of ones and zeros
• The lower the bandwidth, the lower the quality
• RTP payload types
• Processing power
• The better quality (for a given bandwidth) uses a
  more complex algorithm
• A balance between quality and cost

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Voice Quality

• Bandwidth is easily quantified
• Voice quality is subjective
• MOS, Mean Opinion Score
• ITU-T Recommendation P.800
• Excellent 5
• Good 4
• Fair 3
• Poor 2
• Bad 1
• A minimum of 30 people
• Listen to voice samples or in conversations

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• P.800 recommendations
• The selection of participants
• The test environment
• Explanations to listeners
• Analysis of results
• Toll quality
• A MOS of 4.0 or higher

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• Subjective and objective quality-testing
  techniques
• PSQM Perceptual Speech Quality Measurement
• ITU-T P.861
• faithfully represent human judgement and
  perception
• algorithmic comparison between the output signal
  and a know input
• type of speaker, loudness, delay, active/silence
  frames, clipping, environmental noise

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A Little About Speech

• Speech
• Air pushed from the lungs past the vocal cords
  and along the vocal tract
• The basic vibrations vocal cords
• The sound is altered by the disposition of the
  vocal tract ( tongue and mouth)
• Model the vocal tract as a filter
• The shape changes relatively slowly
• The vibrations at the vocal cords
• The excitation signal

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Speech sounds

• Voiced sound
• The vocal cords vibrate open and close
• Interrupt the air flow
• Quasi-periodic pluses of air
• The rate of the opening and closing the pitch
• A high degree of periodicity at the pitch period
• 2-20 ms

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• Voiced speech

• Power spectrum density

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• Unvoiced sounds
• Forcing air at high velocities through a
  constriction
• The glottis is held open
• Noise-like turbulence
• Show little long-term periodicity
• Short-term correlations still present

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• unvoiced speech

• Power spectrum density

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• Plosive sounds
• A complete closure in the vocal tract
• Air pressure is built up and released suddenly
• A vast array of sounds
• The speech signal is relatively predictable over
  time
• The reduction of transmission bandwidth can be
  significant

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Voice Sampling

• A-to-D
• discrete samples of the waveform and represent
  each sample by some number of bits
• A signal can be reconstructed if it is sampled at
  a minimum of twice the maximum freq.
• Human speech
• 300-3800 Hz
• 8000 samples per second

Each sample is encoded into an 8-bit PCM code
word (e.g. 01100101)
time
gt 8000 x 8 bit/s
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Quantization

• How many bits is used to represent
• Quantization noise
• The difference between the actual level of the
  input analog signal
• More bits to reduce
• Diminishing returns
• Uniform quantization levels
• Louder talkers sound better
• 11.2/11 v.s. 2.2/2

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• Non-uniform quantization
• Smaller quantization steps at smaller signal
  levels
• Spread signal-to-noise ratio more evenly

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DTX and Comfort Noise

• DTX is Discontinuous Transmission
• Voice activity detector (VAD) detects if there is
  active speech or not.
• When there is no active speech different DTX
  procedures can be used
• No Transmission at all
• Comfort Noise (CN) using RFC 3389
• Codec built CN in like AMR SID (Silence
  Descriptor)
• Frequency of Comfort Noise packets varies but is
  usually some fraction of normal packet rate

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Type of Speech Coders

• Waveform codecs
• Sample and code
• High-quality and not complex
• Large amount of bandwidth
• source codecs (vocoders)
• Match the incoming signal to a math model
• Linear-predictive filter model of the vocal tract
• A voiced/unvoiced flag for the excitation
• The information is sent rather than the signal
• Low bit rates, but sounds synthetic
• Higher bit rates do not improve much

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• Hybrid codecs
• Attempt to provide the best of both
• Perform a degree of waveform matching
• Utilize the sound production model
• Quite good quality at low bit rate

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G.711

• The most commonplace codec
• Used in circuit-switched telephone network
• PCM, Pulse-Code Modulation
• If uniform quantization
• 12 bits 8 k/sec 96 kbps
• Non-uniform quantization
• 64 kbps DS0 rate
• mu-law
• North America
• A-law
• Other countries, a little friendlier to lower
  signal levels
• An MOS of about 4.3

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DPCM

• DPCM, Differential PCM
• Only transmit the difference between the
  predicated value and the actual value
• Voice changes relatively slowly
• It is possible to predict the value of a sample
  base on the values of previous samples
• The receiver perform the same prediction
• The simplest form
• No prediction
• No algorithmic delay

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ADPCM

• ADPCM, Adaptive DPCM
• Predicts sample values based on
• Past samples
• Factoring in some knowledge of how speech varies
  over time
• The error is quantized and transmitted
• Fewer bits required
• G.721
• 32 kbps
• G.726
• A-law/mu-law PCM -gt 16, 24, 32, 40 kbps
• An MOS of about 4.0 at 32 kbps

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Analysis-by-Synthesis (AbS) Codecs

• Hybrid codec
• Fill the gap between waveform and source codecs
• The most successful and commonly used
• Time-domain AbS codecs
• Not a simple two-state, voiced/unvoiced
• Different excitation signals are attempted
• Closest to the original waveform is selected
• MPE, Multi-Pulse Excited
• RPE, Regular-Pulse Excited
• CELP, Code-Excited Linear Predictive

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G.728 LD-CELP

• CELP codecs
• A filter its characteristics change over time
• A codebook of acoustic vectors
• A vector a set of elements representing various
  char. of the excitation
• Transmit
• Filter coefficients, gain, a pointer to the
  vector chosen
• Low Delay CELP
• Backward-adaptive coder
• Use previous samples to determine filter
  coefficients
• Operates on five samples at a time
• Delay lt 1 ms
• Only the pointer is transmitted

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• 1024 vectors in the code book
• 10-bit pointer (index)
• 16 kbps
• LD-CELP encoder
• Minimize a frequency-weighted mean-square error

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• LD-CELP decoder
• An MOS score of about 3.9
• One-quarter of G.711 bandwidth

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G.723.1 ACELP

• 6.3 or 5.3 kbps
• Both mandatory
• Can change from one to another during a
  conversation
• The coder
• A band-limited input speech signal
• Sampled at 8 KHz, 16-bit uniform PCM quantization
• Operate on blocks of 240 samples at a time
• A look-ahead of 7.5 ms
• A total algorithmic delay of 37.5 ms other
  delays
• A high-pass filter to remove any DC component

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• Various operations to determine the appropriate
  filter coefficients
• 5.3 kbps, Algebraic Code-Excited Linear
  Prediction
• 6.3 kbps, Multi-pulse Maximum Likelihood
  Quantization
• The transmission
• Linear predication coefficients
• Gain parameters
• Excitation codebook index
• 24-octet frames at 6.3 kbps, 20-octet frames at
  5.3 kbps

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• G.723.1 Annex A
• Silence Insertion Description (SID) frames of
  size four octets
• The two lsbs of the first octet
• 00 6.3kbps 24 octets/frame
• 01 5.3kbps 20
• 10 SID frame 4
• An MOS of about 3.8
• At least 27.5 ms delay

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G.729

• 8 kbps
• Input frames of 10 ms, 80 samples for 8 KHz
  sampling rate
• 5 ms look-ahead
• Algorithmic delay of 15 ms
• An 80-bit frame for 10 ms of speech
• A complex codec
• G.729.A (Annex A), a number of simplifications
• Same frame structure
• Encoder/decoder, G.729/G.729.A
• Slightly lower quality

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• G.729.B
• VAD, Voice Activity Detection
• Based on analysis of several parameters of the
  input
• The current frames plus two preceding frames
• DTX, Discontinuous Transmission
• Send nothing or send an SID frame
• SID frame contains information to generate
  comfort noise
• CNG, Comfort Noise Generation
• G.729, an MOS of about 4.0
• G.729A an MOS of about 3.7

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• G.729 Annex D
• a lower-rate extension
• 6.4 kbps 10 ms speech samples, 64 bits/frame
• MOS ? 6.3 kbps G.723.1
• G.729 Annex E
• a higher bit rate enhancement
• the linear prediction filter of G.729 has 10
  coef.
• that of G.729 Annex E has 30 coef.
• the codebook of G.729 has 35 bits
• that of G.729 Annex E has 44 bits
• 118 bits/frame 11.8 kbps

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Other Codecs

• CDMA QCELP defined in IS-733
• Variable-rate coder
• Two most common rates
• The high rate, 13.3 kbps
• A lower rate, 6.2 kbps
• Silence suppression
• For use with RTP, RFC 2658

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• GSM Enhanced Full-Rate (EFR)
• GSM 06.60
• An enhanced version of GSM Full-Rate
• ACELP-based codec
• The same bit rate and the same overall packing
  structure
• 12.2 kbps
• Support discontinuous transmission
• For use with RTP, RFC 1890

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• GSM Adaptive Multi-Rate (AMR) codec
• 20 ms coding delay
• Eight different modes
• 4.75 kbps to 12.2 kbps
• 12.2 kbps, GSM EFR
• 7.4 kbps, IS-641 (TDMA cellular systems)
• Change the mode at any time
• Offer discontinuous transmission
• The SID (Silence Descriptor) is sent in every 8th
  frame and is 5 bytes in size
• The coding choice of many 3G wireless networks

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• The MOS values are for laboratory conditions
• G.711 does not deal with lost packets
• G.729 can accommodate a lost frame by
  interpolating from previous frames
• But cause errors in subsequent speech frames
• Processing Power
• G.728 or G.729, 40 MIPS
• G.726 10 MIPS

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iLBC

• a FREE codec for robust VoIP
• 13.33 kbit/s with an encoding frame length of 30
  ms and 15.20 kbps of 20 ms
• Computational complexity in a range of G.729A

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Speex

• Open-source patent-free speech codec
• CELP (code-excited linear prediction) codec
• operating modes
• narrowband (8 kHz sampling rate)
• 2.15 24.6 kb/s
• delay of 30 ms
• wideband (16 kHz sampling rate)
• 4-44.2 kb/s
• delay of 34 ms
• ultra-wideband (32 kHz sampling rate)
• intensity stereo encoding
• variable bit rate (VBR) possible
• voice activity detection (VAD)

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• Cascaded Codecs
• E.g., G.711 stream -gt G.729 encoder/decoder
• Might not even come close to G.729
• Each coder only generate an approximate of the
  incoming signal
• Audio samples
• http//www.cs.columbia.edu/hgs/audio/codecs.html

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Effects of packetization
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Tones, Signal, and DTMF Digits

• The hybrid codecs are optimized for human speech
• Other data may need to be transmitted
• Tones fax tones, dialing tone, busy tone
• DTMF digits for two-stage dialing or voice-mail
• G.711 is OK
• G.723.1 and G.729 can be unintelligible
• The ingress gateway needs to intercept
• The tones and DTMF digits
• Use an external signaling system

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• Easy at the start of a call
• Difficult in the middle of a call
• Encode the tones differently from the speech
• Send them along the same media path
• An RTP packet provides the name of the tone and
  the duration
• Or, a dynamic RTP profile an RTP packet
  containing the frequency, volume and the duration
• RFC 2198
• An RTP payload format for redundant audio data
• Sending both types of RTP payload

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• RTP Payload Format for DTMF Digits
• An Internet Draft
• Both methods described before
• A large number of tones and events
• DTMF digits, a busy tone, a congestion tone, a
  ringing tone, etc.
• The named events
• E the end of the tone, R reserved

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• Payload format