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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.


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Journal ArticleDOI
TL;DR: This document describes the development of and implementation of the AMR (adaptive multi-rate) method, and some of the components that went into this process.
Abstract: (접수일자: 2011년 2월 28일; 수정일자: 2011년 4월 27일; 채택일자: 2011년 5월 9일)대역폭 확장 (Bandwidth Extension) 은 300-3400 Hz 대역의 협대역 음성 신호를 50-7000 Hz 대역의 광대역 음성신호로 확장하여 협대역 음성신호의 음질과 명료도를 높이는 기술이다. 본 논문에서는 협대역 음성 정보만을 이용해서 광대역 음성신호를 추정하는 인공 대역폭 확장 기술을 설계하여, ITU-T 협대역 표준 음성 코덱인 AMR (adaptive multi-rate) 복호화기에 내장시킴 (embedded)으로써, 대역폭 확장 모듈에서의 LPC 분석 및 LSP 해석과 관련된 계산량을 감소시켰고, 알고리즘 지연도 줄였다. 그리고 SDS (single distance search) 고속 탐색 방식을 대역폭 확장 시스템의 코드북 매핑에 적용하여, 최종적으로 저 전력 대역 확장 AMR 복호화기를 설계하였다 . 제안된 대역폭 확장 방법은 AMR 복호화기 후단에 독립적으로 설치되는 기존 DTE (decode then extend) 방식에 비해 28 % 정도의 계산량을 줄이고 알고리즘 지연도 20 msec 줄였다. 또한 제안방식은 피치정보를 이용한 classified 코드북 매핑 방식을 사용하여 스펙트럼 포락선을 확장 하였고 , 코드 벡터 탐색 시 가중치를 적용하여 광대역 합성 음성의 성능을 향상시켰다 .핵심용어: 대역폭 확장, 음성코덱, 음질 개선, AMR 부호화기투고분야: 음성처리 분야 (2.3)Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper des igns an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the c omputational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.Keywords: Bandwidth Extension, Speech Codec, Speech Enhancement, AMR Code cASK subject classification: Speech Signal Processing (2.3)
Patent
02 Jul 2008
TL;DR: In this paper, the transmission power is set in a power control state in response to a current signal quality measure and in a rate control state wherein speech transmission is controlled by switching the speech rate.
Abstract: In a method and mobile station for transmitting speech data using a speech codec enabled to code coding speech data at varying speech rates, the transmission power is set in a power controlstate in response to a current signal quality measure and in a rate control state wherein speech transmission is controlled by switching the speech rate in response to the signalqualitymeasure and the transmission power is kept essentially constant Using the method will prevent power control based on the signal quality such as the SIR target when instead it would be better to reduce the used transport block size, i.e. reduce the speech codec rate. As a result of the improved use of radio resources, the voice capacity of a cellular system can be increased, while maintaining speech quality.
Proceedings ArticleDOI
05 Jun 2000
TL;DR: An overview of the requirements of digital covert audio acquisition is provided and a scalable hybrid coder tuned for the coding of intelligible speech in a covert environment is proposed and the results of informal intelligibility testing and an analysis of the algorithm's complexity are presented.
Abstract: An overview of the requirements of digital covert audio acquisition (DCAA) is provided and a scalable hybrid coder tuned for the coding of intelligible speech in a covert environment is proposed. The codec incorporates linear predictive coding (LPC) and an M-band discrete wavelet transform (DWT) to offer effective intelligible speech coding in the presence of multiple signal sources at bit rates between 8 kbps and 32 kbps. The results of informal intelligibility testing and an analysis of the algorithm's complexity are presented to demonstrate the performance of the proposed coder.
Patent
28 Sep 2001
TL;DR: In this article, a speech extender is proposed for the estimation of a broadband speech signal from a narrowband speech signal, which is adaptive such to adjust to the coding of a communication terminal and/or to a speaker.
Abstract: The invention relates to a speech extender, embodied for the estimation of a broadband speech signal from a narrowband speech signal. Furthermore, the above is adaptive such as to adjust to the coding of a communication terminal and/or to a speaker.
Proceedings ArticleDOI
04 Jun 2014
TL;DR: A scalable audio coding system based on joint speech/audio coding method and an auditory perceptual importance model based on bit-plane are proposed here and experimental results indicate that with the same bit rates the subjective quality of proposed method is slightly better than G.729.1.
Abstract: With the technical evolution of global mobile communications, various heterogeneous communication environments, frequently fluctuant bandwidth and multiform signals put new challenges to coding technology of multimedia signals. Scalable Audio Coding (SAC) can provide smooth transition between different coding qualities, which is an optimal choice for coding audio signals of different types and can produce more reliable and consistent service quality in multimedia communications. A scalable audio coding system based on joint speech/audio coding method and an auditory perceptual importance model based on bit-plane are proposed here. Both the audio content and the network bandwidth fluctuation will be considered in the system to obtain stable service qualities in mobile multimedia services. Experimental results indicate that with the same bit rates the subjective quality of proposed method is slightly better than G.729.1 and the SNR is improved by 0.3dB.

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