Showing papers on "Adaptive Multi-Rate audio codec published in 1987"

15/30 Mbit/s Universal Digital TV Codec Using a Median Adaptive Predictive Coding Method

Abstract: This paper describes a 15/30 Mbit/s TV codec with a new approach to high-efficiency coding for TV signals, i.e., median adaptive prediction. The 15/30 Mbit/s codec, commonly applicable to NTSC, PAL, and SECAM (525/60 and 625/50) systems, uses adaptive prediction incorporating a motion-compensated interframe, an interfield, and an intrafield predictor. Its performance for digital transmission is presented. This universal codec is designed, based on CCIR recommendations concerning digital TV coding parameters for studios (Rec. 601) and general principles on long-distance digital TV transmission (Rec. 604). A field trial of 15 Mbit/s digital TV transmission using this codec between earth stations with a 30 m diameter antenna and a 5 m diameter antenna is reported.

A new speech coding model based on a least-squares sinusoidal representation

Abstract: In recent years the concept of analysis-by-synthesis has been applied very successfully to improving the performance of LPC based models At the same time, new speech models have been introduced based on representing speech by a sum of amplitude and frequency-modulated sinusoids which have been shown to successfully represent the non-linear, time-varying and quasi-periodic nature of speech In this paper we present an approach to applying the analysis-by-synthesis technique to sinusoidal speech modelling in an attempt to increase the ability of the model to accurately represent the speech waveform

A 16 kbps ADPCM with multi-quantizer (ADPCM-MQ) codec and its implementation by digital signal processor

Abstract: This paper describes an implementation of a new 16 kbps speech codec using commercially available DSPs and its performance. The coding algorithm chosen here is ADPCM with Multi-Quantizer (ADPCM-MQ) which selects the optimum ADPCM coder frame by frame and switches to it dynamically. To implement this coding algorithm, we used two Fujitsu DSPs (MB8764), 1.5 chips for the encoder and 0.5 chip for the decoder. Reconstructed speech with a 21 dB segmental SNR was obtained. With error correction, this codec provides good speech quality even with a bit-error rate of 10^-2 to 10^-3. To improve the subjective quality of the reconstructed speech, adaptive postfiltering was also applied. Since the processing delay of this codec is less than 10 ms, no echo-canceller is needed. Moreover, 2400 bps voice band data (CCITT Rec.V. 26) could be transmitted with a data error rate from 10^-7 to 5×10^-6, and G.III facsimiles were successfully transmitted using this codec.

Implementation of a multi-pulse speech codec with pitch prediction on a single chip floating-point signal processor

Abstract: The multi-pulse speech coding with pitch prediction has been known as an efficient speech coding technique for coding speech at a bit rate of 8 to 9.6 kbps. To implement this coding method on a signal processor for real time applications, problems exist concerning the amount of computations, speech quality, and the data RAM size on a signal processor. As countermeasures against these problems, we propose: 1) Correlation computation amount reduction method by utilizing pitch periodicity of the impulse response, 2) Pulse search method to modify pulse amplitude by only a small amount of computations in order to improve speech quality, 3) Efficient use of the memory space of a signal processor. These methods made it possible to implement a 8 to 9.6 kbps multi-pulse speech codec with pitch prediction with an optimum analysis frame length of 20 ms on a single chip 32-bit floating point signal processor (µPD77230).

Hybrid codec for video signals

Abstract: Motion compensation can be used for improving the prediction within a hybrid codec. As a rule, great circuit complexity is required for determining the displacement vectors. If the data block to be coded or decoded in the hybrid codec, and a particular search area accommodated in a frame buffer of the hybrid codec are subjected to a coarse adaptive quantization, the mathematical effort for the motion compensation can be considerably reduced in a simple manner.