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.

Perceptually-Based Joint-Program Audio Coding

Abstract: Some digital audio broadcasting systems, such as Satellite Digital Audio Radio Services (SDARS), transmit many audio programs over the same transmission channel. Instead of splitting up the channel into fixed bitrate subchannels, each carrying one audio program, one can dynamically distribute the channel capacity among the audio programs. We describe an algorithm which implements this concept taking into account statistics of the bitrate variation of audio coders and perception. The result is a dynamic distribution of the channel capacity among the coders depending on the perceptual entropy of the individual programs. This solution provides improved audio quality compared with fixed bitrate subchannels for the same total transmission capacity. The proposed scheme is non-iterative and has a low computational complexity.

A Real-time Wavelet-based Video Decoder using SIMD Technology

Abstract: This paper presents a fast implementation of a wavelet-based video codec. The codec consists of motion-compensated temporal filtering (MCTF), 2-D spatial wavelet transform, and SPIHT for wavelet coefficient coding. It offers compression efficiency that is competitive to H.264. The codec is implemented in software running on a general purpose PC, using C programming language and streaming SIMD extensions intrinsics, without assembly language. This high-level software implementation allows the codec to be portable to other general-purpose computing platforms. Testing with a Pentium 4 HT at 3.6GHz (running under Linux and using the GCC compiler, version 4), shows that the software decoder is able to decode 4CIF video in real-time, over 2 times faster than software written only in C language. This paper describes the structure of the codec, the fast algorithms chosen for the most computationally intensive elements in the codec, and the use of SIMD to implement these algorithms.© (2008) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

On the implementation of software-only videoconferencing codec on PC

Abstract: Intensive recent progress in computer technologies has stimulated new solutions for transmission of audio and video information in digital form, The transmission of digital video and audio signals through channels of restricted capacity, such as switched public telephone lines, mobile radio systems, local computer networks, Internet and satellite telecommunications attract a constantly growing interest. The bit rate in such lines available for transmission of audio and video information is limited dependent upon particular application in the range from 30 to 512 Kbit/s. In spite of the low bit rate digital coding itself requires significant computational resources and implies severe limitations on the performance of the real-time encoder. It should be also taken into account that for channels with low bit rate a codec must implement compression algorithms allowing us to achieve a large compression ratio and acceptable quality. Necessary compression ratios can be achieved only by means of intraframe coding as well as interframe motion compensation.

Wideband speech coding based on the MBE structure.

Abstract: This paper deals with the adaptation to wideband of the MBE coder which was initially developed for the telephone band. As the constraints of quality and bit rate for a wideband and a telephone band coder are different, and as the signal characteristics on these two bands are different too, we must reconsider the coder structure. Several improvements are proposed, some of which were already proposed for the telephone band such as the phonetic classification of the frames or the multiharmonic modelling of the spectrum. We also propose in order to reach a good quality, especially for high frequency voices, to model and synthesize, as part of the signal, the initial error between the synthetic and original spectra.