A low-delay code-excited linear prediction (LD-CELP) speech coder which is expected to be standardized in 1992 as a CCITT G Series Recommendation for universal applications of speech coding at 16 kb/s is presented. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive and by using a small excitation vector size of five samples. The official CCITT laboratory tests revealed that the speech quality of this 16 kb/s LD-CELP coder is either equivalent to or better than that of the CCITT G.721 standard 32-kb/s ADPCM coder for almost all conditions tested. A description of the LD-CELP algorithm, its implementation on the DSP32C for CCITT testing, and performance results from these tests are presented. >

A low-delay CELP coder for the CCITT 16 kb/s speech coding standard

A low-bitrate (typically 8 kbit/s or less), low-delay digital coder and decoder based on Code Excited Linear Prediction for speech and similar signals features backward adaptive adjustment for codebook gain and short-term synthesis filter parameters and forward adaptive adjustment of long-term (pitch) synthesis filter parameters. A highly efficient, low delay pitch parameter derivation and quantization permits overall delay which is a fraction of prior coding delays for equivalent speech quality at low bitrates.

Method and apparatus for low-delay celp speech coding and decoding

A high-quality 16-kb/s speech coder which has a one-way coding delay of less than 2 ms is presented. The coder is basically a backward-adaptive version of the code-excited linear prediction (CELP) coder. The low coding delay is achieved by using backward-adaptive predictor and gain and by using an excitation vector size as small as five samples. The pitch predictor in conventional CELP coders is eliminated, and the linear predictive coding (LPC) predictor order is increased from 10 to 50. The excitation gain is updated by a tenth-order adaptive logarithmic gain predictor. This log-gain predictor and the LPC predictor are updated by performing LPC analysis on previous log-gain and coded speech, respectively. The excitation codebook is closed-loop optimized and the codebook index is Gray-coded to improve the robustness against channel errors. Formal subjective tests indicate that this coder produces high-quality speech comparable to that of the CCITT G.721 32-kb/s ADPCM standard. >

High-quality 16 kb/s speech coding with a one-way delay less than 2 ms

A candidate algorithm for the new CCITT 16-kb/s speech coding standard is presented. This algorithm is based on backward-adaptive CELP (code-excited linear prediction) where the predictor and the excitation gain are updated by analyzing previously quantized signals. The only information transmitted is the excitation vector with a size as small as five samples so as to achieve a one-way coding delay of less than 2 ms. With a clear channel, this 16-kb/s coder slightly outperformed the CCITT standard 32 kb/s ADPCM (adaptive differential pulse code modulation) (G.721) in speech quality as measured by the mean opinion score (MOS). With noisy channels, the coder scored slightly higher than G.721 fora bit-error rate of 10/sup -2/ and significantly higher (by a margin of 0.5) for bit-error rate of 10/sup -3/. >

A robust low-delay CELP speech coder at 16 kbits/s

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies. >

High-quality coding of telephone speech and wideband audio

The authors study a speech coder using a tree code generated by a stochastic innovations tree and backward adaptive synthesis filters. The synthesis configuration uses a cascade of two all-pole filters-a pitch (long time delay) filter followed by a formant (short time delay) filter. Both filters are updated using backward adaptation. The formant predictor is updated using an adaptive lattice algorithm. The multipath (M,L) search algorithm is used to encode the speech. A frequency weighted error measure is used to reduce the perceptual loudness of the quantization noise. The speech coder has low delay (1 ms) and has been evaluated through formal subjective testing to have speech quality that is equivalent to that for 7-bit log-PCM. >

V. Iyengar

Papers

A low delay 16 kbits/sec speech coder