Speech-Coding Techniques. Chapter 3

Transcription

1 Speech-Coding Techniques Chapter 3

2 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 VoIP 2-52

3 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 VoIP 2-53

4 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 VoIP 2-54

5 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 VoIP 2-55

6 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 VoIP 2-56

7 Speech sounds Voiced sound The vocal cords vibrate open and close Interrupt the air flow Quasi-periodic pulses of air The rate of the opening and closing the pitch A high degree of periodicity at the pitch period 2-20 ms VoIP 2-57

8 Voiced speech Power spectrum density VoIP 2-58

9 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 VoIP 2-59

10 unvoiced speech Power spectrum density VoIP 2-60

11 Plosive sounds A complete closure in the vocal tract Air pressure is built up and released suddenly As in the sound p in pit or d in dog A vast array of sounds The speech signal is relatively predictable over time The reduction of transmission bandwidth can be significant VoIP 2-61

12 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 Hz 8000 samples per second VoIP 2-62

13 Quantization How many bits are used to represent a sample Quantization noise The difference between the actual level of the input analog signal VoIP 2-63

14 More bits to reduce noise Diminishing returns Uniform quantization levels Louder talkers sound better 11.2/11 v.s. 2.2/2 Non-uniform quantization Smaller quantization steps at smaller signal levels Spread signal-to-noise ratio more evenly VoIP 2-64

15 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 VoIP 2-65

16 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 VoIP 2-66

17 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 A-law North America Other countries, a little friendlier to lower signal levels An MOS of about 4.3 VoIP 2-67

18 ADPCM DPCM, Differential PCM Only transmit the difference between the predicted value and the actual value Voice changes relatively slowly It is possible to predict the value of a sample based on the values of previous samples The receiver performs the same prediction The simplest form No prediction No algorithmic delay VoIP 2-68

19 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 kbps G.726 A-law/mu-law PCM -> 16, 24, 32, 40 kbps An MOS of about 4.0 at 32 kbps VoIP 2-69

20 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 VoIP 2-70

21 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 characteristics 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 < 1 ms Only the pointer is transmitted VoIP 2-71

22 1024 vectors in the code book 10-bit pointer (index) 16 kbps LD-CELP encoder Minimize a frequency-weighted mean-square error VoIP 2-72

23 LD-CELP decoder An MOS score of about 3.9 One-quarter of G.711 bandwidth VoIP 2-73

24 G 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 VoIP 2-74

25 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 prediction coefficients Gain parameters Excitation codebook index 24-octet frames at 6.3 kbps, 20-octet frames at 5.3 kbps VoIP 2-75

26 G Annex A Silence Insertion Description (SID) frames of size four octets The two lsbs of the first octet kbps 24 octets/frame kbps SID frame 4 An MOS of about 3.8 At least 37.5 ms delay VoIP 2-76

27 G 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, interchangeable G.729/G.729.A Slightly lower quality VoIP 2-77

28 G.729, an MOS of about 4.0 G.729.A an MOS of about 3.7 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 VoIP 2-78

29 G.729 Annex D a lower-rate extension 6.4 kbps; 10 ms speech samples, 64 bits/frame MOS 6.3 kbps G 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 VoIP 2-79

30 Other Codecs CDMA QCELP defined in IS-733 Qualcom Code-Excited Linear Predictor 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 VoIP 2-80

31 GSM Enhanced Full-Rate (EFR) GSM 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 VoIP 2-81

32 GSM Adaptive Multi-Rate (AMR) codec GSM 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 coding choice of many 3G wireless networks VoIP 2-82

33 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 MIPS VoIP 2-83

34 Cascaded Codecs E.g., G.711 stream -> G.729 encoder/decoder Quality might not even come close to G.729 Each coder only generate an approximate of the incoming signal VoIP 2-84

35 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 voic G.711 is OK G and G.729 can be unintelligible The ingress gateway needs to intercept The tones and DTMF digits Use an external signaling system VoIP 2-85

36 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 VoIP 2-86

37 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 VoIP 2-87

38 Payload format VoIP 2-88