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.

Method and apparatus for transmitting video contents compressed by codec

Abstract: A method and apparatus for transmitting compressed video content are provided. The method includes transmitting a codec selection request frame, the codec selection request frame including an identifier of one or more codecs to be used to compress video content and requesting approval of the use of the codec, receiving a codec selection response frame, the codec selection response frame including result information indicating whether the use of the codec is approved, and transmitting video content frames for the video content compressed by the approved codec based on the codec selection response frame. Each video content frame includes frame type information indicating a type of compression applied to the video content included in the video content frame.

A Multiprocessor Based Low Bitrate Video Codec Architecture and a Well-Adapted Two Phase Software Coding Strategy

Abstract: The first part of the paper describes a specialized multiprocessor environment for hybrid coding of visual communications signals in the range from ISDN basic access to primary rate transmission channels. Most important is a proprietary 4-wide SIMD parallel video processor with 80 MIPS. The second part deals with the software philosophy of the codec. It uses preanalysis and prebuffering in the first phase of coding a frame. In the second phase, limited processing power and available channel bits are distributed optimally over time and over changed areas of one frame. Codec delay is halved with respect to conventional codecs.

Multiband prototype waveform analysis synthesis for very low bit rate speech coding

Abstract: Prototype waveform interpolation is one of the most efficient compression techniques for coding the speech signal at bit rates below 4 kb/s. Most of the PWI coders employ prototype waveforms of the linear predictive residual signal for coding purpose. In the latest PWI systems, decomposition methods are used to separate the voiced and unvoiced components of the prototype waveforms prior to coding. This has resulted in high quality speech at very low bit rates. This paper presents a novel combination of the multiband voicing analysis and PWI coding system in which the multiband analysis is exploited to identify the voiced and unvoiced spectral components of the prototype waveforms of the original speech signal. To produce a high quality synthetic speech, the energy variation of the original signal is recovered by transmitting its energy envelope. This method resulted in a high quality and low complexity coder operating at 2.55 kb/s.

System and Method for a High Performance Audio Codec

Abstract: A system for a high performance audio codec provides higher voice quality and higher recognition accuracy from an ASR engine at an increased data rate and computational power and embodiments include those having a CELP-based codec, an ASR engine, a text comparator, an encoder, a decoder, an LPC Computation and formant analysis module, a dual stage data rate determination module, a VQ of LSP coefficients module, a pitch synthesis and optimal pitch parameter search module, and an excitation codebook parameter search module. A method for high performance audio codec includes three stages and comprises the steps of having an ASR engine yield transcribed text from each of an uncompressed reference signal and a decompressed signal that has passed through an encoder and wherein the transcribed text is compared with original text to determine word error rates in an iterative process whereby both voice quality and recognition accuracy are optimized.

Non-uniform QMF Decomposition for Wide-band Audio Coding based on Frequency Domain Linear Prediction

Abstract: This paper presents a new technique for perfect reconstruction non-uniform QMF decomposition developed to increase efficiency of a generic wide-band audio coding system based on Frequency Domain Linear Prediction (FDLP). The base line FDLP codec, operating at high bit-rates ( 136 kbps), exploits an uniform QMF decomposition into 64 sub-bands followed by sub-band processing based on FDLP. Here, we propose a non-uniform QMF decomposition into 32 frequency sub-bands obtained by merging 64 uniform QMF bands. The merging operation is performed in such a way that bandwidths of the resulting critically sampled sub-bands emulate the characteristics of the critical band filters in the human auditory system. Such frequency decomposition, when employed in the FDLP audio codec, results in a bit-rate reduction of 40% over the base line. We also describe the complete audio codec, which provides high-fidelity audio compression at 66 kbps. In subjective listening tests, the FDLP codec outperforms MPEG-1 Layer 3 (MP3) and achieves similar qualities as MPEG-4 AAC+ standard.