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 published on a yearly basis
Papers
More filters
09 Sep 1997
TL;DR: A real-time full-duplex videoconferencing system using parallel DSPs is described and effective strategies for H.263 video coding have been developed to improve the efficiency of the system.
Abstract: A real-time full-duplex videoconferencing system using parallel DSPs is described. Effective strategies for H.263 video coding have been developed to improve the efficiency of the system. Further speed-up is achieved by utilising multiple DSP processors. The video codec is integrated with a G.723.1 speech codec to form an H.324 compliant bitstream, using H.223 multiplex and H.245 control protocols.
4 citations
30 Jul 2014-International Journal of Advanced Research in Electrical, Electronics and Instrumentation Energy
TL;DR: A new algorithm for speech signals compression using wavelet transform technique with Discrete Cosine Transform (DCT) technique is implemented and evaluated based on Signal to Noise Ratio (SNR), Root Mean Square Error (MSE) and compression ratio tested on speech signals.
Abstract: Compared to most digital data types, with the exception of digital audio, the data rates associated with uncompressed digital audio are substantial. Digital audio compression enables more efficient storage and transmission of audio data. The many forms of audio compression techniques offer a range of encoder and decoder complexity, compressed audio quality, and differing amounts of data compression. In this paper a new algorithm for speech signals compression using wavelet transform technique with Discrete Cosine Transform (DCT) technique. The performance of the implemented algorithm is evaluated based on Signal to Noise Ratio (SNR), Root Mean Square Error (MSE) and compression ratio tested on speech signals. In this paper a Wavelet & cosine hybrid model, based speech coder is implemented in software using Matlab.
4 citations
TL;DR: A very low-complexity audio codec that provides audio playback quality similar to the MPEG-I/audio level 3 codec at 64 Kbps for a monophonic-channel signal and an adaptive arithmetic coder with multiplication-free adaptation is presented.
4 citations
05 Sep 2007
TL;DR: Experimental results carried out on the real speech signals show that the performance of the proposed VAD algorithm is better than that of AMR-WB VAD.
Abstract: This paper proposed a new voice activity detection (VAD) algorithm using support vector machine (SVM) for improving the VAD performance of AMR-WB speech codec. The SVM is applied to train an optimized non-linear decision rule involving the VAD parameters, e.g., sub-band signal level, pitch gain, background noise level, and etc., defined in AMR-WB standard. Then, by the use of the trained SVM, the proposed algorithm can achieve accurate VAD under various noisy conditions. Experimental results carried out on the real speech signals show that the performance of the proposed VAD algorithm is better than that of AMR-WB VAD.
4 citations
04 Sep 2006
TL;DR: The goal here is to drive the estimation of the echo path by the watermark itself, in order to take advantage of its optimal properties (whiteness and stationarity) and the proposed WAEC exhibits better transient and steady state performance than the classical one.
Abstract: Audio watermarking, or embedding information in a host signal was originally used for digital copyright protection purposes. As audio coding, watermarking is progressively brought in audio processing applications.
4 citations