Topic
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.
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20 Sep 1995
TL;DR: The MT-CELP as mentioned in this paper is a carefully optimized anulysis-by-synthesis speech coder with associated channel coding for a future enhanced fill-rate speech mode in GSM based systems.
Abstract: In the near future there will be a need for mobile services which can provide quality close to wire-line. This paper describes the Ericsson candidate codec to meet the requirements for a future enhanced fill-rate speech mode in GSM based systems. It is a carefully optimized anulysisby-synthesis speech coder called MT-CELP with associated channel coding. MOS tests show that all the requirements can be met.
1 citations
TL;DR: AudioPaK as discussed by the authors is a simple lossless audio codec, which uses only a small number of integer arithmetic operations on both the coder and the decoder side for polynomial prediction and Golomb-Rice coding.
1 citations
22 Apr 2008
TL;DR: It is shown from the experiment that the audio conferencing system employing the proposed cross-layer PLC algorithm improves audio quality significantly under packet loss conditions.
Abstract: In this paper, we propose a cross-layer packet loss concealment (PLC) algorithm in order to improve the quality of audio for a real-time audio conferencing system. The proposed algorithm tries to estimate a packet loss rate (PLR) of the network and to request a sender to transmit redundant information for audio decoding when the PLR is assumed to be higher. An audio quality measure implemented on the application layer is used for an estimate of the PLR. In addition, an RTP payload format for the transport layer is defined here to control the degree of redundancy for the PLC algorithm. When lower quality of audio is measured, the packet of the current frame is composed of the bitstream of the current frame and that of the previous frame to help the audio decoder reconstruct the audio signals of the previous frame with better quality than with no redundancy. In order to show the effectiveness of the proposed PLC algorithm, we use MPEG-2 Advanced Audio Coding (AAC) and the ITU-T P.563 as an audio codec and an objective quality measure of audio signals, respectively. It is shown from the experiment that the audio conferencing system employing the proposed cross-layer PLC algorithm improves audio quality significantly under packet loss conditions.
1 citations
TL;DR: Entropy coding principles are applied to the ITU G.728 speech codec and it is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity.
Abstract: Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality.
1 citations