Digital Speech Coding
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1 Digital Speech Processing David Tipper Associate Professor Graduate Program of Telecommunications and Networking University of Pittsburgh Telcom 2720 Slides 7 Digital Speech Coding Digital Speech Convert analog speech to digital form and transmit digitally Applications Telephony: (cellular, wired and Internet- VoIP) Speech Storage (Automated call-centers) High-Fidelity recordings/voice Text-to-speech (machine generated speech) Issues Efficient use of bandwidth Compress to lower bit rate per user => more users Speech Quality Want tollgrade or better quality in a specific transmission environment Environment ( BER, packet lost, packet out of order, delay, etc.) Hardware complexity Speed (coding/decoding delay), computation requirement and power consumption Telcom
2 Digital Speech Processing Speech coding in wireless systems All 1G systems have analog speech transmission 2G and 3G systems have digital speech Type of source coding Motivation for digital speech Increase system capacity Compression possible Quality/bandwidth tradeoffs can be made Improve quality of speech Error control coding possible, equalization, etc. Improve security as encryption possible for privacy Reduce Cost and Operations and Maintenance (OAM) Telcom Typical Wireless Communication System Source Source Encoder Channel Encoder Modulator Channel Destination Source Decoder Channel Decoder Demod -ulator Telcom
3 Characteristics of Speech Bandwidth Most of energy between 20 Hz to about 7KHz, Human ear sensitive to energy between 50 Hz and 4KHz Time Signal High correlation Short term stationary Classified into four categories Voiced : created by air passed through vocal cords (e.g., ah, v) Unvoiced : created by air through mouth and lips (e.g., s, f ) Mixed or transitional Silence Telcom Characteristics of Speech Typical Voiced speech Typical Unvoiced speech Telcom
4 Digital Speech Speech Coder: device that converts speech to digital Types of speech coders Waveform coders Convert any analog signal to digital form Vocoders (Parametric coders) Try to exploit special properties of speech signal to reduce bit rate Build model of speech transmit parameters of model Hybrid Coders Combine features of waveform and vocoders Telcom Speech Quality of Various Coders Mean Opinion Score is a subjective measure of quality Tradeoff in quality vs. data rate vs. complexity Telcom
5 Waveform Coders (e.g.,pcm) Waveform Coders Convert any analog signal to digital - basically A/D converter signal sampled > twice highest frequency- then quantized into ` n bit samples Uniform quantization Example Pulse Code Modulation band limit speech < 4000 Hz pass speech through μ law compander sample 8000 Hz, 8 bit samples 64 Kbps DS0 rate Characteristics Quality High Complexity Low Bit rate High Delay -Low Robustness - High Telcom PCM Speech Coding input Bandpass filter compressor Sampleand-hold circuit PAM -to- Digital converter PCM μ law compander Transmission medium output Bandpass filter expander Hold circuit PAM Digital-to converter μ law expander Pulse code modulation (PCM) system with analog companding then digital conversion ITU G.700 standard basis for speech coding In PSTN in 60 s Telcom
6 Companding 1 μ-law companding Output μ = (no compression) Compander emphasizes small values, de-emphasizes large values in-order to equalize SNR across samples. Reverse the mapping at the receiver with an expandor Input Telcom PCM Speech Coding Digitally companded PCM system ITU G.711 standard better quality speech than analog companding input Bandpass filter Sampleand-hold circuit PAM -to- Digital converter Linear PCM Digital compressor Compressed PCM PCM transmitter Transmission medium output Bandpass filter Hold circuit PAM Digital-to converter Linear PCM Digital expander PCM receiver Differential PCM (DPCM) : reduce bit rate from 64 Kbps to 32 Kbps) since change is small between sample transmit 1 sample then on transmit difference between samples use 4 bits to quantize adaptively adjust range of quantizer improves quality (ADPCM ITU G.726 ) Telcom
7 DPCM Speech Coding input Low-pass filter Differentiator (summer) -to- Digital converter Encoded difference samples Accumulated signal level Integrator Digital-to converter DPCM transmitter DPCM input Digital-to converter Integrator Hold circuit Low-pass filter output DPCM receiver Telcom Subband Speech Coding Bandpass Filter 1 A/D 1 speech Bandpass Filter 2 A/D 2 Mux Channel encoder Bandpass Filter 3 A/D 3 Partition signal into non-overlapping frequency bands use different A/D quantizer for each band Example: 3 subbands = 31.2 Kbps band Range encoding Hz 4 bits Hz 3 bits Hz 2 bits Telcom
8 Vocoders Vocoders (Parametric Coders) Models the vocalization of speech Speech sampled and broken into frames (~25 msec) Instead of transmitting digitized speech 1. Build model of speech 2. Transmit parameters of model 3. Synthesize approximation of speech Linear Predictive Coders (LPC) basic Vocoder model Models vocal tract as a filter Filter excitation periodic pulse (voiced speech) or noise (unvoiced speech) Transmitted parameters: gain, voiced/unvoiced decision, pitch (if voiced), LPC parameters Telcom Vocoders Linear Predictive Coders (LPC) Excitation periodic pulse (voiced speech) or noise (unvoiced speech) Transmitted parameters: gain, voiced/unvoiced decision, pitch (if voiced), LPC parameters Telcom
9 Vocoders Example Tenth Order Linear Predictive Coder Samples Voice at 8000 Hz buffer 240 samples => 30 msec Filter Model (M=10 is order, G is gain, z -1 unit delay, b k are filter coefficients) H ( z ) = 1 + M G k = 1 b k z G = 5 bits, b k = 8 bits each, voiced/unvoiced decision = 1 bit, pitch = 6 bits => 92 bits/30 msec = 3067 bps k Telcom Vocoders LPC coders can achieve low bit rates Kbps Characteristics of LPC Quality Low Complexity Moderate Bit Rate Low Delay Moderate Robustness Low Quality of pure LPC vocoder to low for cellular telephony - try to improve quality by using hybrid coders Try to improve the quality by refining model of speech, improve accuracy of model improve input to speech coder Telcom
10 Vocoders Hybrid Coders Combine Vocoder and Waveform Coder concept Residual LPC (RELP) Codebook excited LPC (CELP) Telcom RELP Vocoder Residual Excited LPC improve quality of LPC by transmitting error (residue) along with LPC parameters s(n) Buffer/ Window residue v/u decision gain, pitch ST-LP Analysis LP parameters LPC Synthesis ENCODER Encoded output Block diagram of a RELP encoder Telcom
11 GSM Speech Coding speech Low-pass filter A/D 104 kbps 13 kbps RPE-LTP Channel speech encoder encoder 8000 samples/s, 13 bits/sample GSM uses Regular Pulse Excited -- Linear Predictive Coder (RPE-- LPC) for speech Basically combine DPCM concept with LPC Information from previous samples used to predict the current sample. The LPC coefficients, plus an encoded form of the residual (predicted - actual sample = error), represent the signal. Telcom GSM Speech Coding (cont) Regular pulse excited - long term prediction (RPE-LRP) speech encoder (RELP speech coder 160 samples/ 20 ms from A/D (= 2080 bits) RPE-LTP speech encoder 36 LPC bits/20 ms 9 LTP bits/5 ms 47 RPE bits/5 ms 260 bits/20 ms to channel encoder LPC: linear prediction coding filter LTP: long term prediction pitch + input RPE: Residual Prediction Error: Telcom
12 GSM Speech Coding (cont) Channel encoder 260 bits/ 20 ms = 13 kb/s 50 class 1a bits 3-bit CRC 182 class 1b bits 78 class 2 bits 53 bits 4 tail bits* (2,1,5) convolution coder 470 bits Bit interleaver Class 1a: CRC (3-bit error detection) and convolutional coding (error correction) Class 1b: convolutional coding Class 2: no error protection *tail bits to periodically reset convolutional coder 456 bits/ 20 ms = 22.8 kb/s Telcom Hybrid Vocoders Codebook Excited LPC Problem with simple LPC is U/V decision and pitch estimation doesn t model transitional speech well, and not always accurate Codebook approach pass speech through an analyzer to find closest match to a set of possible excitations (codebook) Transmit codebook pointer + LPC parameters NA-TDMA standard, IS-95, 3G, ITU G.729 standard Telcom
13 Typical CELP Encoder Telcom CELP Speech Coders General CELP architecture Telcom
14 CELP Speech Coders i Code Book #1 γ 1 γ 1 l Code Book #2 γ 2 LPC Synthesis Filter Output γ 2 h Code Book #3 β β Filter Coefficients Block diagram of the NA-TDMA (IS-54) speech coder subband codebook approach termed vector sum excited LPC (VSELPC) Telcom Evaluating Speech Coders Qualitative Comparison based on subjective procedures in ITU-T Rec. P. 830 Major Procedures Absolute Category Rating Subjects listen to samples and rank them on an absolute scale - result is a mean opinion score (MOS) Comparison Category Rating Subjects listen to coded samples and original uncoded sample (PCM or analog), the two are compared on a relative scale result is a comparison mean opinion score (CMOS) Mean Opinion Score (MOS) Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 Comparison MOS (CMOS) Much Better 3 Better 2 Slightly Better 1 About the Same 0 Slightly Worse -1 Worse -2 Much Worse -3 Telcom
15 Evaluating Speech Coders MOS for clear channel environment no errors Result vary a little with language and speaker gender Standard Speech coder Bit rate MOS PCM Waveform 64 Kbps 4.3 CT2 ADPCM 32 Kbps 4.1 DECT ADPCM 32 Kbps 4.1 NA-TDMA Hybrid VSELPC 8Kbps 3.0 GSM Hybrid RELPC 13 kbps 3.54 QCELP Hybrid CELP 14.4 Kbps QCELP Hybrid CELP 9.6 Kbps 3.4 LPC Vocoder 2.4 Kbps 2.5 ITU G.729 Hybrid CELP 8.Kbps 3.9 Telcom Evaluating Speech Coders Types of environments recommended for testing coder quality Clean Channel no background noise Vehicle : emulate car background noise Street : emulate pedestrian environment Hoth : emulate background noise in office environment (voice band interference) Consider environments above for cases of Perfect Channel no transmission errors Random channel errors Bursty channel errors May consider repeated encoding/decoding (e.g., mobile to mobile call) Telcom
16 Evaluating Speech Coders Background noise and errors degrade quality Repeated coding degrades quality Telcom Codec Selection For cellular need to consider Quality, Complexity, Delay, Compression Rate ITU Coder Bit Rate Coding Delay Decoding Delay Complexity G Kbps 0 0 Low G Kbps 15 ms 7.5 ms Medium G.723.a,b 6.4/5.3 Kbps 35.5 ms ms High Telcom
17 3G Standards Two competing 3G standards Both standards use multi-mode CELP vocoders 1. 3GPP/cdma GPP/UMTS (SMV Multimode rate set 1) Variable bit rate vocoder Source Control of bit rate Channel coding treats all bits equally (AMR-NB Multi-rate) Fixed rate vocoder Voice Activity Detection Discontinuous Transmission network control of coder rate Tailors Channel coding to speech coder Telcom Silence Compression Much of a conversation is Silence (~40%) no need to transmit Voice Activity Detector (VAD) Hardware to detect silence period quickly Variable Bit Rate coders reduce bit rate when silence Discontinuous transmission (DTX) Stop transmitting frames Send minimal # of frames to keep connection up Comfort Noise Generator (CNG) Synthesize background noise avoids: Did you hang up? Random noise or reproduce speaker s ambient background For example GSM codec and popular VoIP G codec has VAD/DTX/CNG Cdmaone and CDMA2000 codec use variable bit rate approach Telcom
18 Silence Compression Telcom Voice Coding Basic Voice Coding Approaches Waveform Vocoders Hybrid Vocoders Evaluation of Vocoder Quality Codebook based vocoders use in new technology 3GPP and ITU recently standardized a AMR wideband CELP input HZ rather than Hz of current systems more natural quality speech slightly higher bit rate Telcom
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