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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14 Sep 1995
TL;DR: The main focus of this paper is to give an overview on the ongoing work towards the NBC (Non Backwards-Compatible) extension to MPEG-2 audio and MPEG-4 audio coding.
Abstract: Since 1988 the audio subgroup is ISO/IEC JTC1/SC29 WG11 (called MPEG) has been working on the standardization of high quality low bit-rate audio coding. Until now the standards 11172-3 (1) (MPEG-1 audio) and 13818-3 (2) (MPEG-2 audio) have been finished. The main focus of this paper is to give an overview on the ongoing work towards the NBC (Non Backwards-Compatible) extension to MPEG-2 audio and MPEG-4 audio coding. >
2 citations
TL;DR: A method of MAP-based audio coding compensation for speaker recognition that firstly sets a standard codec as a reference and trains the speaker models in this codec format, then learns the deviation distributions between the standard codec format and the other ones, and gets the current bias via using a small number adaptive data and theMAP-based adaptive technique.
Abstract: The performance of the speaker recognition system declines when training and testing audio codecs are mismatched. In this paper, based on analyzing the effect of mismatched audio codecs in the linear prediction cepstrum coefficients, a method of MAP-based audio coding compensation for speaker recognition is proposed. The proposed method firstly sets a standard codec as a reference and trains the speaker models in this codec format, then learns the deviation distributions between the standard codec format and the other ones, next gets the current bias via using a small number adaptive data and the MAP-based adaptive technique, and then adjusts the model parameters by the type of coming audio codec format and its related bias. During the test, the features of the coming speaker are used to match with the adjusted model. The experimental result shows that the accuracy reached 82.4% with just one second adaptive data, which is higher 5.5% than that in the baseline system.
2 citations
TL;DR: The result shows that the codec using the mel-LP analysis method results in a good performance at a low bit rate, particularly at 16kbps, and in the cases of pop songs, piano music and male speech, sound quality was improved.
Abstract: In this paper, we propose a low-bit-rate audio codec using a new analysis method named mel-scaled linear predictive analysis (mel-LP analysis). In mel-LP analysis, a spectral envelope is estimated on a mel- or bark-frequency scale, so as to improve the spectral resolution in the low-frequency band. This analysis is accomplished with about a twofold increase in computation over standard LPC analysis. Our codec using mel-LP analysis consists of five key parts: time frequency transformation, flattening of MDCT coefficients using the mel-LP spectral envelope, power normalization, perceptual weighting estimation, and multistage VQ. In subjective experiments, we investigated the performance of our codec using the mel-LP analysis method, through 7-level paired comparison tests. The result shows that the codec using the mel-LP analysis method results in a good performance at a low bit rate, particularly at 16kbps. In the cases of pop songs, piano music and male speech, sound quality was improved.
2 citations
Patent•
03 Feb 2000
TL;DR: In this paper, the authors present a DECT telephone system and a method for transmitting data in a Generic Access Profile (GAP) system using the ETSI DECT standard.
Abstract: The switching center includes a speech codec (20), a data codec, and a control device (16) operating according to the Generic Access Profile (GAP) of the ETSI DECT standard. A switch-over device (18) connects a terminal of the control device to a terminal of the speech codec or data codec in response to a transmission mode being identified. Also claimed are a DECT telephone system and a method for transmitting data in a GAP system.
2 citations
22 May 2016
TL;DR: Experimental simulation results show that the DWT is a promising technique to use for encoding highly non-stationary signals such as the speech coding error and it is shown that the best choice of wavelet depends on many factors including the order and number of levels of the wavelet tree, delay and how well it approximates the human auditory system.
Abstract: We developed scalable narrowband and wideband speech coding schemes based on the internet low bitrate codec (iLBC). Some of these newer codecs used the Discrete Wavelet Transform (DWT) instead of the Modified Discrete Cosine Transform (MDCT). This paper explores the choice of wavelet packet transform (WPT) for an application for a new scalable speech codec for IP networks using the Discrete Wavelet Transform (DWT) to encode the core-layer coding error in the enhancement layer. The issues regarding the design and in particular the choice of wavelet for the wideband codec are discussed. Experimental simulation results show that the DWT is a promising technique to use for encoding highly non-stationary signals such as the speech coding error. The wideband codec achieved speech quality equivalent to ITU-G.718 and similar codecs and is more robust. We also show that the best choice of wavelet depends on many factors including the order and number o f levels of the wavelet tree, delay and how well it approximates the human auditory system.
2 citations