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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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Journal ArticleDOI
TL;DR: This paper gives a brief overview on the complete audio part of the MPEG-4 standard and more detailed information on its parts related to speech coding.
Abstract: While previous MPEG Audio standards mainly were focused on the representation of audio signals close to or equal to CD quality, the new MPEG-4 Audio standard extends the range of applicability towards significantly lower bit rates. Furthermore it offers extended functionalities for the representation of natural and even synthetic audio signals in an object oriented fashion. This paper gives a brief overview on the complete audio part of the MPEG-4 standard and more detailed information on its parts related to speech coding.

12 citations

Proceedings ArticleDOI
17 Sep 2000
TL;DR: An adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services achieves an enhanced performance for background noise while maintaining its clean speech quality.
Abstract: This paper describes an adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services. The coder is a multi rate SB-CELP (subband-code excited linear prediction) with five modes operating at bit rates from 24 kbit/s down to 9.1 kbit/s. Our basic approach consists of an unequal band-splitting of the input signal into two subbands (SB). A variable rate, multi-mode ACELP coder is applied to the lower subband (0-6 kHz). The various bit rates are integrated in a common structure where the scalability is realized by exchanging the fixed excitation codebooks while leaving all other codec parameters invariant. For the GSM related modes (9.1-17.8 kbit/s), the upper subband (6-7 kHz) is coded using a very low bit rate representation based on bandwidth expansion techniques. In case of the 3G application (24 kbit/s) the upper band is coded using a 4 kbit/s ADPCM coding scheme. In addition the analysis by synthesis (AbS) coder of the lower band employs a novel closed loop gain re-quantization technique controlled by the character of the speech signal. Thereby the codec achieves an enhanced performance for background noise while maintaining its clean speech quality.

12 citations

Proceedings ArticleDOI
14 Apr 1991
TL;DR: A speech codec, called TRPE-HLTP (transformed binary regular pulse excitation, high-resolution long-term prediction), has been developed and has been submitted as a candidate for the half-rate codec in the GSM (Group Special Mobile) system.
Abstract: A speech codec, called TRPE-HLTP (transformed binary regular pulse excitation, high-resolution long-term prediction) has been developed. It has been submitted as a candidate for the half-rate codec in the GSM (Group Special Mobile) system and its gross bit rate is 11.4 kb/s. The speech coding algorithm is of the type often called analysis-by-synthesis linear prediction. Its key elements are LSF (line spectral frequency)-coded spectral parameters, a high resolution closed-loop adaptive codebook, and a speech-trained low-complexity transformed binary regular pulse innovation generator, resulting in a net bit rate of 6.9 kb/s. The channel coding scheme consists of forward error correction with convolutional encoding, interleaving, and error detection. >

12 citations

Proceedings ArticleDOI
14 Nov 2005
TL;DR: This paper presents a multiple description (MD) video codec based on the principles side-information coding that uses a bank of sequential LDPC decoders to efficiently decode the transmitted coset information.
Abstract: This paper presents a multiple description (MD) video codec based on the principles side-information coding. In particular, we highlight certain key components of the codec design that contribute significantly to the rate-distortion performance of the proposed codec. These include the use of randomized permutations of the quantization codebook in conjunction with binary LDPC codes for partitioning the available bit-rate among the coefficient bit-planes. Another key component of the proposed codec is the use of pdf estimation for improved decoder reconstruction. Lastly, we use a bank of sequential LDPC decoders to efficiently decode the transmitted coset information. Empirical evaluation demonstrates the superior performance of the proposed codec for the communication of encoded video over packet erasure channels.

11 citations

01 Feb 2008
TL;DR: This document specifies real-time transport protocol (RTP) payload formats to be used for the EVRC wideband codec (EVRC-WB) and updates the media type registrations for EVRC-B codec.
Abstract: This document specifies real-time transport protocol (RTP) payload formats to be used for the EVRC wideband codec (EVRC-WB) and updates the media type registrations for EVRC-B codec. Several media type registrations are included for EVRC-WB RTP payload formats. In addition, a file format is specified for transport of EVRC-WB speech data in storage mode applications such as e-mail.

11 citations


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