Showing papers on "Adaptive Multi-Rate audio codec published in 1978"
TL;DR: The Parcor analysis‐synthesis method is being applied to a wide range of speech coding from 1200 bps variable frame‐rate coding to high quality 16 kbps adaptive, predictive coding.
Abstract: Since the introduction of speech analysis—synthesis based on the maximum likelihood spectrum estimation—in 1966, we have been conducting research activities on low bit rate speech coding techniques, and their aplication to audio response and low bit rate digital speech transmission. Parcor analysis‐synthesis, demonstrated in 1969, was one of the most fundamental methods, and it has formed the basis of the present development of linear predictive coding. Recently, various kinds of techniques have been proposed to improve speech quality, such as interpolation and nonlinear quantization of parameters, spectral smoothing, etc. They have been applied in the hardware realization of a 4 CH multiplexed 2400 bps Vocoder. At present, the Parcor method is being applied to a wide range of speech coding from 1200 bps variable frame‐rate coding to high quality 16 kbps adaptive, predictive coding.
7 citations
TL;DR: In this paper, a shared \mu \cong 255 PCM codec has been developed for the D4 channel bank which was introduced in 1976, consisting of two hybrid IC's that utilize a common LSI D/A converter chip.
Abstract: A shared \mu \cong 255 PCM codec has been developed for the D4 channel bank which was introduced in 1976. The codec consists of two hybrid IC's that utilize a common LSI D/A converter chip. Operating over the temperature range of 0-75°C, the codec economically provides near theoretical S/D perfornance, gain tracking of better than \pm 0.25 dB, gain stability better than \pm 0.15 dB, and average system performance of 14.5-dBrnCO idle channel noise and 83-dB cross-talk coupling loss. A description of this codec and the derivation of its design objectives are presented.
4 citations
01 Apr 1978
TL;DR: This paper describes an audio response method, where audio messages are stored in a movable-head magnetic disk memory in the form of speech feature parameters through the PARCOR speech analysis, suitable to interactive CAI services for individually paced training.
Abstract: This paper describes an audio response method, where audio messages are stored in a movable-head magnetic disk memory in the form of speech feature parameters through the PARCOR speech analysis. Desired message data is asynchronously read out of the disk memory according to the readout demand queue into a buffer memory to drive a speech synthesizer. The achievable number of output channels is related with the average message duration and the disk memory characteristics. This method realizes an audio filing unit for maximally 128 output channels, of 0.1-second response time, and with 5000 messages of 15-second duration per 100 MB disk. It is suitable to interactive CAI services for individually paced training.