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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.


Papers
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Proceedings ArticleDOI
05 Dec 2005
TL;DR: This paper proposes and evaluates power-based congestion control mechanisms in the WCDMA downlink air interface based on the functionalities of the call admission control, conversational service class rate adaptation and load control algorithms.
Abstract: This paper proposes and evaluates power-based congestion control mechanisms in the WCDMA downlink air interface based on the functionalities of the call admission control (CAC), conversational service class rate adaptation (RA) via AMR codec and load control (LC) algorithms. It is verified by means of dynamic system level simulations that the congestion control technique is capable of guaranteeing the users' QoS requirements even for high offered loads.

3 citations

Proceedings ArticleDOI
08 Oct 2007
TL;DR: A candidate perceptual weighting scheme is proposed and testified on AVS-M audio coder, and subjective results of the optimized scheme provide hybrid audio quality comparable to the 3 GPP AMR-WB+.
Abstract: Audio video coding standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) audio coding tools target mobility applications where perceptual weighting is of great importance for subjective performance. This paper first briefly introduces the general concept of perceptual weighting. Then a candidate perceptual weighting scheme is proposed and testified on AVS-M audio coder. Further optimization on the candidate is illustrated thereafter. Subjective results of the optimized scheme provide hybrid audio quality comparable to the 3 GPP AMR-WB+.

3 citations

Patent
13 Oct 2009
TL;DR: In this paper, the authors present an audio playback device that decodes and plays back a stream containing basic CODEC, whereby noise generation can be avoided without markedly increasing the quantity of calculations even with multi-channel situations.
Abstract: Disclosed is an audio playback device (100), which decodes and plays back a stream containing basic CODEC, whereby noise generation can be avoided without markedly increasing the quantity of calculations even with multi-channel situations, and which is equipped with a stream separator (101), which separates a stream into basic CODEC and band expansion data, a basic CODEC analyzer (102), which analyzes the basic CODEC, a basic CODEC decoder (103), which decodes the basic CODEC according to basic CODEC analysis data, a band expansion data analyzer (104), which analyzes the band expansion data, a first band expansion processor (106), which uses the band expansion data to spread the decoded basic CODEC signal, a second band expansion processor (107), which uses the band expansion data to spread the decoded basic CODEC signal with higher precision than the first band expansion processor (106), and a switch (109), which switches between the first band expansion processor (106) and the second band expansion processor (107) based on the basic CODEC analysis data.

3 citations

Proceedings ArticleDOI
11 Apr 1988
TL;DR: A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed, based on time scale modification and sub-band coding with the aid ofADPCM with a multiquantizer that was confirmed by computer simulations.
Abstract: A new 8 kbps speech coding algorithm called TC-MQ (time domain compression ADPCM-MQ) is proposed. It is based on time scale modification and sub-band coding with the aid of ADPCM with a multiquantizer. For time scale modification, the decimation/interpolation technique is introduced for the unvoiced period. Furthermore, in order to get computational accuracy of the pitch extraction for the voiced period, a method using the normalized autocovariance function is proposed. The algorithm was confirmed by computer simulations. The segmental SNR was about 13-16 dB for Japanese short sentences. A good mean opinion score value was also obtained by means of a subjective evaluation test. >

3 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202310
202214
20201
20193
20183
201721