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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Papers
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03 May 2005TL;DR: In this article, a method of codec mode adaptation of a speech codec in dependency of the prevailing channel condition for transmission of speech frames in a telecommunication system comprising the steps of determining a bit error rate (BER) from a estimated carrier to interferer ratio (C/I) per speech burst, generating a frame BER value of speech frame from a plurality of consecutive bursts and determining a critical BER level for a plurality OF speech frames by a maximum operation of the frame BERT values of the plurality of OFC frames.
Abstract: Method of codec mode adaptation of a speech codec in dependency of the prevailing channel condition for transmission of speech frames in a telecommunication system comprising the steps of determining a bit error rate (BER) from a estimated carrier to interferer ratio (C/I) per speech burst, generating a frame BER value of speech frame from a plurality of consecutive bursts and determining a critical BER level for a plurality of speech frames by a maximum operation of the frame BER values of the plurality of speech frames.
8 citations
01 Oct 2006
TL;DR: This document specifies a Real-time Transport Protocol (RTP) payload format to be used for the International Telecommunication Union (ITU-T) G.729.1 audio codec.
Abstract: This document specifies a Real-time Transport Protocol (RTP) payload
format to be used for the International Telecommunication Union
(ITU-T) G.729.1 audio codec. A media type registration is included for
this payload format. [STANDARDS-TRACK]
8 citations
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8 citations
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08 Apr 2013TL;DR: In this paper, a method, system, and computer-usable non-transitory storage device for dynamic voice codec adaptation are disclosed, where the voice codec adapts in real time to devote more bits to audio quality when it is most needed, and fewer bits to less important parts of utterances.
Abstract: A method, system, and computer-usable non-transitory storage device for dynamic voice codec adaptation are disclosed. The voice codec adapts in real time to devote more bits to audio quality when it is most needed, and fewer bits to less important parts of utterances are disclosed. Dialog knowledge is utilized for compression opportunities to adjust the bitrate moment-by-moment, based on the inferred value of each frame. Frame importance and appropriate transmission fidelity is predicted based on prosodic features and models of dialog dynamics. This technique provides the same communications quality with less spectrum needs, fewer antennas, and less battery drain.
8 citations
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15 Oct 2003TL;DR: The results indicate that stereo MP3 at 44 kHz and 128 kbps can be decoded using 27 MIPS on the ARM9TDMI, and the output of the decoder is fully bit-compliant with the standard on the ISO test vectors.
Abstract: MPEG-1/2 audio layer-3 (MP3) is the must popular format for playback of high quality compressed audio for portable devices such as audio players and mobile phones. Typically these devices are based on either DSP or RISC processors. While the DSP architecture is more efficient for implementing the MP3 algorithm, the challenges a RISC implementation are lesser understood. This paper describes the challenges and optimization techniques useful for implementing the MP3 decoder algorithm on the RISC-based ARM9TDMI processor. Some of these techniques are generic and hence applicable to the any audio codec implementation on RISC-based platforms. Our results, which are among the best in the industry, indicate that stereo MP3 at 44 kHz and 128 kbps can be decoded using 27 MIPS on the ARM9TDMI. In addition, the output of our decoder is fully bit-compliant with the standard on the ISO test vectors.
8 citations