Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

Progressive Multi-Stage Neural Audio Coding with Guided References

Abstract: In this paper, we propose an effective multi-stage neural audio coding algorithm that encodes full-band audio signals (up to 20 kHz) using an end-to-end training criterion. By predefining several dyadic subband signals as training targets, we progressively encode input audio signals in each stage such that deeper stages of the network encode the residual error terms from the previous encoding stage. Our proposed audio codec successfully decodes full-band audio signals by using an effective multi-stage vector quantization scheme to represent key encoding features extracted in the latent space. Subjective listening tests show that the decoded outputs of the proposed audio codec achieve almost transparent quality at an average bitrate of 132 kbps.

Implementation of H.324 audiovisual codec for mobile computing

Abstract: A VLSI implementation of the H.324 audiovisual codec is described. A number of sophisticated low-power architectures have been devised dedicatedly for the mobile use. A set of specific functional units, each corresponding to a process of H.263 video codec, is employed to lighten different performance bottlenecks. A compact DSP core composed of two MAC units is used for both ACELP and MP-MLQ coding schemes of the G.723.1 speech codec. The proposed audiovisual codec core has been implemented by using 0.35 /spl mu/m CMOS 4LM technology, which contains totally 420 k transistors with a dissipation of 224.32 mW from single 3.3 V supply.

New post-processing techniques for low bit rate celp codecs

Abstract: This paper presents two new post-processing techniques to address limitations of the deployed low bit rate speech codecs in case of unvoiced speech and background noise, and in case of music. Both post-processing techniques enhance the spectrum of the decoded excitation signal without increasing the codec algorithmic delay. The paper discusses how to integrate the enhancement procedure of unvoiced speech and background noise and of generic audio signals coded by low bit rate ACELP codecs. The proposed post-processing procedures are part of the AMR-WB interoperable modes of the recently standardized 3GPP EVS codec [1].

CCITT Standardizing activities on speech coding

Abstract: CCITT (International Telegraph and Telephone Consultative Committee) Study Group XVIII is in charge of studies on speech processing, such as speech coding techniques. During its last study period (1981-84), a recommendation for 32 kbit/s speech coding was made. Afterwards, Study Group XVIII began standardization studies on wideband speech coding (i.e., 7 kHz audio coding) at 64 kbit/s or less. This paper reports on the progress of these standardization studies. The 32 kbit/s Adaptive Differential PCM (ADPCM) standard is briefly reviewed. Also, the 7 kHz audio coding at 64 kbit/s or less is mainly discussed focussing on its applications, design requirements and coding algorithm.

Method for streaming through a data service over a radio link subsystem

Abstract: An apparatus for controlling a data rate in a data client for a digital audio broadcasting system includes a buffer for storing data, a codec for coding data, and a control module for controlling a bit rate of the codec in response to a level of the data in the buffer. A method performed by the apparatus is also included.