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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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Patent
Kazuyoshi Kuwahara1, Koichi Kaji1
05 Feb 1997
TL;DR: In this paper, a sound codec has an encoded signal input/output terminal, which is selectively connected to a modem codec, and a microphone and a speaker are used as a transceiver of the speaker phone.
Abstract: A sound codec has an encoded signal input/output terminal. This terminal is selectively connected to a modem codec. Due to this selective connection, a microphone and a speaker, both connected to a sound signal input/output terminal of the sound codec, are used as a transceiver of the speaker phone, and the sound codec is controlled to function as a speaker phone codec.

27 citations

Proceedings ArticleDOI
23 May 1989
TL;DR: A novel approach to narrow- and medium-band speech coding that can dynamically balance the transmission rate between the excitation and the spectral parameters is introduced, improving the subjective speech quality.
Abstract: The authors introduce a novel approach to narrow- and medium-band speech coding that can dynamically balance the transmission rate between the excitation and the spectral parameters. The coding algorithm, called multimode coding, operates several coding blocks, each of which has a different bit assignment in parallel, and selects the optimum coding block frame by frame based on an evaluation of the reproduced speech quality. This coding algorithm is applied to 4.8 and 8.0 kb/s CELP coders, and 2.0-2.4 dB of SNRseg improvement is achieved over conventional CELP coders. The spectral distortion measure is added as an evaluation function, improving the subjective speech quality. >

27 citations

Proceedings ArticleDOI
01 Apr 1987
TL;DR: This paper presents an approach to applying the analysis-by-synthesis technique to sinusoidal speech modelling in an attempt to increase the ability of the model to accurately represent the speech waveform.
Abstract: In recent years the concept of analysis-by-synthesis has been applied very successfully to improving the performance of LPC based models At the same time, new speech models have been introduced based on representing speech by a sum of amplitude and frequency-modulated sinusoids which have been shown to successfully represent the non-linear, time-varying and quasi-periodic nature of speech In this paper we present an approach to applying the analysis-by-synthesis technique to sinusoidal speech modelling in an attempt to increase the ability of the model to accurately represent the speech waveform

27 citations

Journal ArticleDOI
TL;DR: The results show that the MFCC-based codec exceeds the state-of-the-art MELPe codec across the entire range of 600-2400 bps, when evaluated with the perceptual evaluation of speech quality (PESQ) (ITU-T recommendation P.862).
Abstract: In this paper, we propose a low bit-rate speech codec based on vector quantization (VQ) of the mel-frequency cepstral coefficients (MFCCs) We begin by showing that if a high-resolution mel-frequency cepstrum (MFC) is computed, good-quality speech reconstruction is possible from the MFCCs despite the lack of phase information By evaluating the contribution toward speech quality that individual MFCCs make and applying appropriate quantization, our results show that the MFCC-based codec exceeds the state-of-the-art MELPe codec across the entire range of 600-2400 bps, when evaluated with the perceptual evaluation of speech quality (PESQ) (ITU-T recommendation P862) The main advantage of the proposed codec is in distributed speech recognition (DSR) since the MFCCs can be directly applied thus eliminating additional decode and feature extract stages; furthermore, the proposed codec better preserves the fidelity of MFCCs and better word accuracy rates as compared to CELP and MELPe codecs

27 citations

Proceedings ArticleDOI
Yang Gao1, Adil Benyassine2, Jes Thyssen2, Huan-Yu Su2, Eyal Shlomot2 
07 May 2001
TL;DR: This paper presents the core technology of novel enhancements to achieve toll quality at 4 kbps, traditional CELP coding, coined eXtended CelsP (eX-CELP), and showed that this technology is also successful and suitable for centered on a combined and selective usage of closed-loop/open-both high and medium bit rates.
Abstract: This paper presents the core technology of novel enhancements to achieve toll quality at 4 kbps, our experiments and test results traditional CELP coding, coined eXtended CELP (eX-CELP). It is showed that this technology is also successful and suitable for centered on a combined and selective usage of closed-loop/open-both high and medium bit rates. Fig. I and Fig.2 illustrate the basic loop approach, and variant algorithm structure concept. The above structure of the eX-CELP encoder and decoder. two concepts are complemented by new features and refined One of the main themes of the eX-CELP technology is the existing technologies. The eX-CELP paradigm was used in judicious combination of the closed-loop approach and the open-several speech coding systems. It is the core technology of the loop approach, together with a careful selective usage of them. recently chosen candidate for the 3G-CDMA speech codec This mechanism is coined COLA, and its main objective is to standard. It was the best candidate for ITU-T 4 kbps codec intelligently employ the most appropriate approach for different qualification test, and became the basis technology for a types of input signals in order to preserve the perceptually consortium candidate to the ITU-T 4 kbps speech coding important contents. Another important feature in the eX-CELP competition.

27 citations


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