A real-time implementation of a source coding method allowing bit rate reduction from 16 bits to 3 bits per sample at sampling rates up to 48 kHz is described. Using an adaptive transform coding technique optimized for low computing complexity and a system based on the AT&T DSP 32 single-chip signal processor, a very good compromise between sound quality and hardware complexity has been achieved. Distortions were audible only by trained listeners for critical pieces of music. One signal-processor module is needed for each of the two coders and decoders. Each one fits on a single-height Eurocard
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...
International audiencea new method for coding generic audio signals at 64 kbit/s in the bandwidth 20...
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...
Low bit rate coding schemes for coding high quality digital audio at bit rates of 64 to 128 kbit/sec...
An Analysis/Synthesis Audio Codec (ASAC) is presented, which allows the coding of audio signals at v...
The speed of current PCs enables them to decode and play an MPEG bitstream in real time. The encodin...
The Low Complexity Adaptive Transform Coding (LC-ATC)- and the Optimum Coding in the Frequency Domai...
This project focused on the creation of a series of audio processing functions that could run in rea...
The main purpose of this project is to implement a speech compression algorithm using a digital sign...
Abtract-AC-3 audio coding technology is a kind of Perceptual Audio Coder (PAC) developed by the Dolb...
This paper proposes a VQ-based transform coding scheme for audio signals ( sampled at 8 kHz) at very...
For real-time as well as off-line transmission of high quality audio signals over the Internet, the ...
In recent years, audio coding has proved to be of great importance for applications such as mobile c...
International audienceThis paper describes a fast digital signal processor working on 16 bits data w...
A subband coding system for high quality digital audio signals is described. To achieve low bit rate...
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...
International audiencea new method for coding generic audio signals at 64 kbit/s in the bandwidth 20...
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...
Low bit rate coding schemes for coding high quality digital audio at bit rates of 64 to 128 kbit/sec...
An Analysis/Synthesis Audio Codec (ASAC) is presented, which allows the coding of audio signals at v...
The speed of current PCs enables them to decode and play an MPEG bitstream in real time. The encodin...
The Low Complexity Adaptive Transform Coding (LC-ATC)- and the Optimum Coding in the Frequency Domai...
This project focused on the creation of a series of audio processing functions that could run in rea...
The main purpose of this project is to implement a speech compression algorithm using a digital sign...
Abtract-AC-3 audio coding technology is a kind of Perceptual Audio Coder (PAC) developed by the Dolb...
This paper proposes a VQ-based transform coding scheme for audio signals ( sampled at 8 kHz) at very...
For real-time as well as off-line transmission of high quality audio signals over the Internet, the ...
In recent years, audio coding has proved to be of great importance for applications such as mobile c...
International audienceThis paper describes a fast digital signal processor working on 16 bits data w...
A subband coding system for high quality digital audio signals is described. To achieve low bit rate...
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...
International audiencea new method for coding generic audio signals at 64 kbit/s in the bandwidth 20...
Speech coding at 64 and 32 Kb/s is well developed and standardized. The next bit rate of interest is...