Speech-Coding Techniques in VS .NET

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Speech-Coding Techniques
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content of Figures 3-8 and 3-9 are reproduced here with the prior authorization of the ITU as copyright holder
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After the conversion from A-law or m-law PCM to uniform PCM, the input signal is partitioned into blocks of five-consecutive input signal samples For each input block, the encoder passes each of 1024 candidate codebook vectors (stored in an excitation codebook) through a gain scaling unit and a synthesis filter From the resulting 1024 candidate quantized signal vectors, the encoder identifies the one that minimizes a frequency-weighted mean-squared error measure with respect to the input signal vector The 10-bit codebook index of the corresponding best codebook vector (or codevector ), which gives rise to that best candidate quantized signal vector, is transmitted to the decoder The decoding operation is also performed on a block-by-block basis Upon receiving each 10-bit index, the decoder performs a table look-up to extract the corresponding codevector from the excitation codebook The extracted codevector is then passed through a gain scaling unit and a synthesis filter to produce the current decoded signal vector The decoded signal vector is then passed through an adaptive postfilter to enhance the perceptual quality The postfilter coefficients are updated periodically using the information available at the decoder The five samples of the postfilter signal vector are next converted to five A-law or m-law PCM output samples
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The reason why the G728 coder is termed LD-CELP is the very small algorithmic delay that it introduces Due to the fact that it operates on 5 PCM samples at a time and given that these samples are taken at a rate of 8,000 Hz, the coder introduces a delay of 0625 milliseconds This delay is so small that it would not be noticeable even to the keenest ear The fact that only a 10-bit pointer value is transmitted for every 5 incoming PCM samples means that the actual bandwidth consumed between encoder and decoder is one-quarter of that used by regular PCM (that is, 16 Kbps) Given that the G728 codec gives an MOS score of about 39, the quality reduction is not enormous compared to the bandwidth saved Figure 3-8 provides a simplified block diagram of the LD-CELP encoder and decoder Although the G728 coder provides high quality at a reasonably low bandwidth, it is quite rare to find the coder used in commercial VoIP products The reason is the fact that the G728 coder is quite processor intensive Consequently, the digital signal processors (DSPs) used for G728 are quite expensive There are other coders that provide good quality at a low bandwidth and a lower cost than G728
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Figure 3-8 A simplified block diagram of the LDCELP coder
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64 Kbit/s A-law or -law PCM input Convert to uniform PCM Vector buffer
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Excitation VQ codebook
Gain
Synthesis filter
Perceptual weighting filter
Min MSE
VQ index
16 Kbit/s output
Backward gain adaptation
Backward predictor adaptation
a) LD-CELP encoder
VQ index 16 Kbit/s input
Excitation VQ codebook
Gain
Synthesis filter
Postfilter
Convert to PCM
64 Kbit/s A-law or -law PCM output
T1506740-92
Backward gain adaptation
Backward predictor adaptation
b) LD-CELP decoder
G7231 Algebraic Code-Excited Linear Prediction (ACELP)
ITU-T Recommendation G7231 specifies a speech coder that can operate at either 63 Kbps or 53 Kbps, with the higher bit rate providing higher speech quality Both rates are mandatory parts of the codec and we can change from one mode to another during a conversation The coder takes a band-limited input speech signal that is sampled at 8,000 Hz and that undergoes uniform PCM quantization, resulting in a 16bit PCM signal The encoder then operates on blocks or frames of 240 samples at a time Thus, each frame corresponds to 30 milliseconds of speech, which means that the coder automatically causes a delay of 30 milliseconds The G7231 coder also utilizes a look-ahead of 75 milliseconds, resulting in
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