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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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Proceedings ArticleDOI
Wen Xu1, M. Marke
07 Oct 2001
TL;DR: Investigations are reported on joint source-channel (JSC) decoding applied in the adaptive multirate (AMR) speech codec for typical real speech, and it is shown that for the AMR codec, although highly optimized in exploiting signal redundancy, some redundancy still remains in the encoded speech bits.
Abstract: Investigations are reported on joint source-channel (JSC) decoding applied in the adaptive multirate (AMR) speech codec for typical real speech. The studied approaches include the "HUK" method, two-step decoding, repeated decoding, and error concealment based on the so-called soft bit decoding. A method based on a Kalman filter for a priori information estimation is described, where both inter- and intra-frame correlation are taken into account. It is shown that for the AMR codec, although highly optimized in exploiting signal redundancy, some redundancy still remains in the encoded speech bits. By using an optimized JSC decoding algorithm like the repeated decoding, a gain of up to 0.2 dB in channel signal-to-noise ratio is achievable, compared to decoding without exploiting the residual redundancy. Under bad channel conditions, the error concealment algorithm can deliver an even better subjective speech quality.

1 citations

Book ChapterDOI
01 Jan 2001
TL;DR: This chapter contains sections titled: Introduction Motivation and Background backward-adaptive G728 Codec Schematic Backward- adapting G728 Coding Algorithm Reduced-rate G728-like Codec: Variable-length Excitation Vector The Effects of Long-term Prediction Closed-loop Codebook Training
Abstract: This chapter contains sections titled: Introduction Motivation and Background Backward-adaptive G728 Codec Schematic Backward-adaptive G728 Coding Algorithm Reduced-rate G728-like Codec: Variable-length Excitation Vector The Effects of Long-term Prediction Closed-loop Codebook Training Reduced-rate G728-like Codec: Constant-length Excitation Vector Programmable-rate 8-4 kbps Low-delay CELP Codecs Backward-adaptive Error Sensitivity Issues A Low-delay Multimode Speech Transceiver Chapter Summary

1 citations

Proceedings ArticleDOI
21 Mar 2015
TL;DR: Introducing the basic concepts of speech coding related to signal processing modules through speech and audio coding in the Speech and Audio Processing Laboratory.
Abstract: This research paper reports in the Speech and Audio Processing Laboratory (SAP). Introducing the basic concepts of speech coding related to signal processing modules. The SAP Laboratory enhances the student's education through multimedia signal processing learning through speech and audio coding.

1 citations

Patent
12 Feb 2002
TL;DR: A speech coding apparatus includes a comparator and a selecting unit as discussed by the authors, where the comparator compares at least one of the coding distortions involved in the encoding with a fixed threshold value or with a threshold value that is determined in response to signal power of the input speech, and the selecting unit selects a more favorable excitation that will provide better speech quality.
Abstract: A speech coding apparatus includes driving excitation coding units, a comparator and a selecting unit. The driving excitation coding units encode in respective excitation modes a target signal to be encoded that is obtained from the input speech, and output coding distortions involved in the encoding. The comparator compares at least one of the coding distortions involved in the encoding with a fixed threshold value or with a threshold value that is determined in response to signal power of the input speech or with a threshold value that is determined in response to signal power of the target signal to be encoded. The selecting unit selects the excitation mode in response to the coding distortions and a compared result of the comparator. The speech coding apparatus can select a more favorable excitation that will provide better speech quality, thereby being able to improve the subjective quality of the speech it outputs by decoding resultant speech code.

1 citations

Journal ArticleDOI
TL;DR: A simple and fast motion estimation algorithm to reduce searching time is proposed in a real‐time software‐based H.261 video codec, which can compress video in CIF format over 30 fps and in QCIF at 105 fps, and can achieve a very high decoding rate.
Abstract: ITU-T H.261 is a video coding standard for videophone and video-conferencing applications on LAN and ISDN, which requires a great amount of computing power for DCT and motion estimation, traditionally provided by hardware. Since motion estimation is a major problem in developing real-time video codec, in this paper we propose a simple and fast motion estimation algorithm to reduce searching time. Mainly, a real-time software-based H.261 video codec is investigated and implemented, in which several fast methods such as programming technique and Intel MMX™ instruction are used to improve computing speed. The experimental results have demonstrated that our H.261 codec can compress video in CIF format over 30 fps and in QCIF at 105 fps, and can achieve a very high decoding rate. © 2002 Wiley Periodicals, Inc. Int J Imaging Syst Technol 12, 73–83, 2002; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ima.10013

1 citations


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