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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16 Apr 1990TL;DR: Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal, which include the important subclass of wideband speech.
Abstract: Advances in signal processing which have created several new technologies for high-quality digital audio are discussed. For traditional telephony, characterized by a signal bandwidth of about 3.2 kHz, the transmission rate for network quality speech is now down to 16 kb/s, with the prospect of a new standard from the CCITT. Robust communications quality appropriate for cellular radio has been realized at 8 kb/s. The current focus is on 4 kb/s, with the aim of improving naturalness and speaker identification. In the coding of wideband audio, an important point of reference is the CCITT standard for 7 kHz speech at a rate of 64 kb/s. Results of recent research are pointing to better capabilities-higher signal bandwidth at 64 kb/s, and 7 kHz bandwidth at lower bit rates such as 32 kb/s. The coding of audio with a signal bandwidth of 20 kHz is receiving significant attention due to recent activity in the ISO (International Standards Organization), with a goal of storing a CD-grade monophonic audio channel at a bit rate not exceeding 128 kb/s. Prospects for accomplishing this are found to be very good. As a side result, emerging algorithms will offer very attractive options at lower rates such as 96 and 64 kb/s. >
28 citations
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19 Apr 2009TL;DR: The speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec.
Abstract: This paper describes the speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec. G.729.1 and G.718. The core codec in the current implementation is G.729.1 and the encoded frequency range is increased from 7 kHz to 14 kHz. Therefore, the maximum bit rate is raised from 32 kbit/s to 64 kbit/s by adding five bitstream layers. A comprehensive overview of the codec is presented with a focus on the mono coding components. The results of the listening tests that have been conducted during the ITU-T qualification phase are summarized. The proposed codec passes all quality requirements for mono input signals.
28 citations
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21 Aug 2000
TL;DR: An overview on the current status of parametric audio coding developments is given and advantages and challenges of this approach are demonstrated, which indicate possible directions of further improvements.
Abstract: For very low bit rate audio coding applications in mobile communications or on the Internet, parametric audio coding has evolved as a technique complementing the more traditional approaches. These are transform codecs originally designed for achieving CD-like quality on one hand, and specialized speech codecs on the other hand. Both of these techniques usually represent the audio signal waveform in a way such that the decoder output signal gives an approximation of the encoder input signal, while taking into account perceptual criteria. Compared to this approach, in parametric audio coding the models of the signal source and of human perception are extended. The source model is now based on the assumption that the audio signal is the sum of "components," each of which can be approximated by a relatively simple signal model with a small number of parameters. The perception model is based on the assumption that the sound of the decoder output signal should be as similar as possible to that of the encoder input signal. Therefore, the approximation of waveforms is no longer necessary. This approach can lead to a very efficient representation. However, a suitable set of models for signal components, a good decomposition, and a good parameter estimation are all vital for achieving maximum audio quality. We give an overview on the current status of parametric audio coding developments and demonstrate advantages and challenges of this approach. Finally, we indicate possible directions of further improvements.
28 citations
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TL;DR: The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels.
Abstract: Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications
28 citations
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01 Jun 1992TL;DR: As the authors address new challenges in wideband speech technology, several strides in coding research are likely to occur, including refinements of existing models for auditory noise-masking, and a unification of linear prediction and frequency-domain coding.
Abstract: The technologies of ISDN teleconferencing, CD-ROM multimedia services, and High Definition Television are creating new opportunities and challenges for the digital coding of wideband audio signals, wideband speech in particular. In the coding of wideband speech, an important point of reference is the CCITT standard for 7 kHz speech at a rate of 64 kbit/s. Results of recent research are pointing to better capabilities — higher signal bandwidth at 64 kbit/s, and 7 kHz bandwidth at lower bit-rates such as 32 and 16 kbit/s. The coding of audio with a signal bandwidth of 20 kHz is receiving significant attention due to recent activity in the ISO (International Standards Organization), with a goal of storing a CD-grade monophonic audio channel at a bit-rate not exceeding 128 kbit/s. Prospects for accomplishing this are very good. As a side result, emerging algorithms will offer very attractive options at lower rates such as 96 and 64 kbit/s. As we address new challenges in wideband speech technology, several strides in coding research are likely to occur. Among these are refinements of existing models for auditory noise-masking, and a unification of linear prediction and frequency-domain coding.
27 citations