scispace - formally typeset
Search or ask a question
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.


Papers
More filters
Proceedings ArticleDOI
14 Apr 1991
TL;DR: A single board video codec for ISDN B/2B channel transmission, which depends on a CCITT standardization p*64 video coding algorithm and communication protocol, has been developed.
Abstract: A single board video codec for ISDN B/2B channel transmission, which depends on a CCITT standardization p*64 video coding algorithm and communication protocol, has been developed. The video codec is constructed with newly designed DSPs, four kinds of NTSC-CIF (Common Intermediate Format) mutual conversion LSIs, a transmission codec LSI and an AD/DA hybrid IC. The video codec codes and decodes a full CIF signal at a frame rate of 10 frames/s and communicates over either an ISDN 64 kb/s or 2*6 64 kb/s channel. The codec is fabricated on a single small board with a size of 280 mm*280 mm. Furthermore, a desk-top prototype visual telephone terminal which uses the video codec, voice codec and NCU has been developed. >

4 citations

Proceedings ArticleDOI
08 Sep 2014
TL;DR: Speech compression is used to eliminate the redundancies present in the speech signal and is associated with its efficient transmission using Speech Coding.
Abstract: The development in the field of digital communication has led to drastic improvements in the way human speech is handled for transmission. Voice communication has become a booming field for a few decades. Speech compression is used to eliminate the redundancies present in the speech signal and is associated with its efficient transmission. Speech compression is achieved by using Speech Coding. Speech coding uses speech-specific parameter estimation utilising digital signal processing techniques on audio signals, to model the speech signal. The model obtained is combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.

4 citations

Patent
14 Oct 2005
TL;DR: In this article, a polyphonic audio chip between the controller and a highly integrated audio codec is used for audio control in a dual-processor mobile device, where one processor is a radio processor and the other is an application processor.
Abstract: In a dual processor mobile device one processor is a radio processor and one processor is an application processor, the mobile device further including a polyphonic audio chip, a method and apparatus for audio control, the method having the steps of: connecting the polyphonic audio chip between the controller and a highly integrated audio codec; connecting the radio processor to the highly integrated audio codec; controlling the highly integrated audio codec from the radio processor; and coordinating between the application processor and the radio processor to allow the radio processor to control the highly integrated audio codec through the radio processor

4 citations

Proceedings ArticleDOI
30 Oct 2007
TL;DR: This paper proposes some criteria for users to find the optimal audio codes from energy analyses of them on handheld devices and investigates the effects from choices of codecs and codec parameters.
Abstract: Audio applications on mobile handheld devices are challenging mainly due to the complex computational needs of audio decoding and the limited capacity of batteries of handheld devices. The increasing number of codecs makes the situation even more perplexing for users trying to find the best audio codec that consumes the least electricity for playing on handheld devices. In this paper, we propose some criteria for users to find the optimal audio codes from energy analyses of them on handheld devices. Various experiments have been conducted to investigate the effects from choices of codecs and codec parameters.

4 citations

Proceedings ArticleDOI
Kazunori Ozawa1, T. Nomura, M. Serizawa, H. Ehara, K. Yoshida, N. Tana 
08 Mar 1999
TL;DR: Subjective evaluation results demonstrate that the speech quality for MPEG-4 speech coding at above 8.3 kb/s is higher than that for the ITU-T G.726 ADPCM at 32kb/s in the clean speech condition.
Abstract: This paper evaluates MPEG-4 narrowband (NB) CELP speech coding under various mobile communication conditions, such as clean, background noise and transmission errors In order to make the codec robust against the errors with minimum increase of redundant bits, a CRC error correction code is attached into the codec as well as an error concealment is included in the decoder Subjective evaluation results demonstrate that the speech quality for MPEG-4 speech coding at above 83 kb/s is higher than that for the ITU-T G726 ADPCM at 32 kb/s in the clean speech condition Further, the speech quality degradation is less than 01 in MOS under 10/sup -3/ bit error conditions, and still comparable to or higher than that for G726 at 32 kb/s without error

4 citations


Network Information
Related Topics (5)
Signal processing
73.4K papers, 983.5K citations
79% related
Data compression
43.6K papers, 756.5K citations
78% related
Decoding methods
65.7K papers, 900K citations
78% related
Computational complexity theory
30.8K papers, 711.2K citations
76% related
Hidden Markov model
28.3K papers, 725.3K citations
75% related
Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202310
202214
20201
20193
20183
201721