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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TL;DR: Speech coding techniques discussed here are Linear Predictive Coding, waveform coding, Code excited linear predictive coding, etc, which are studied with the help of MATLAB to check their performance measures like compression ratio and speech audible quality.
3 citations
26 Mar 1995
TL;DR: A new digital codec is described, which can transmit 1125/60 HDTV signals at a bit-rate of 15 to 45 Mbps in order to cover a wide range of applications such as SNG, distribution or contribution.
Abstract: A new digital codec is described, which can transmit 1125/60 HDTV signals at a bit-rate of 15 to 45 Mbps in order to cover a wide range of applications such as SNG, distribution or contribution. To achieve satisfactory picture quality at such low bit rates, including high quality stereo sounds, a data channel, a forward error correction code, and new advanced coding techniques are introduced into a conventional motion compensated interframe and intrafield adaptive DCT coding scheme. By using these key techniques, this system provides a significantly better coding performance than MPEG-2. In order to verify the transmission performance over an actual transmission link, field trials were carried out between Japan and the USA, which demonstrates that this codec is suitable for practical use in a wide range of HDTV applications. >
3 citations
11 Apr 2007
TL;DR: An improved efficiency perceptual audio codec is presented which analyzes each block of input signal and selects a suitable time/frequency mapping transform for it, compared with the widely famous MPEG-1 Layer-III algorithm.
Abstract: An improved efficiency perceptual audio codec is presented which analyzes each block of input signal and selects a suitable time/frequency mapping transform for it. The selection is based on statistics of the input signal vis-a-vis energy compaction and resolution power properties of the transforms employed which include the DFT (uniform subbands), DFT (critical subbands), DCT and CELP( for speech only blocks). The performance of the codec is compared with the widely famous MPEG-1 Layer-III algorithm. Efficiency enhancement is indicated by improved grades of subjective listening test for the proposed codec compared to those for MPEG-1 Layer-III at similar bit rates.. The paper concludes with a discussion of future research implications of the work.
3 citations
28 May 2000
TL;DR: This paper describes implementation of an AAC decoder on a novel dual-DSP architecture utilizing cost-effective 24-bit fixed-point arithmetic.
Abstract: MPEG-4 audio coding is a recent audio standard that integrates a wide range of different types of audio coding, and, thus, applies to almost every audio application today. A very important subpart of MPEG-4 is the Advanced Audio Coding (AAC) algorithm, which has already distinguished itself in industry due to its high audio quality. This paper describes implementation of an AAC decoder on a novel dual-DSP architecture utilizing cost-effective 24-bit fixed-point arithmetic.
3 citations