Multi-channel audio frequency coder

Abstract

A subband audio coder employs perfect / non-perfect reconstruction filters, predictive / non-predictive subband encoding, transient analysis, and psycho-acoustic / minimum mean-square-error (mmse) bit allocation over time, frequency and the multiple audio channels to encode / decode a data stream to generate high fidelity reconstructed audio. The audio coder windows the multi-channel audio signal such that the frame size, i.e. number of bytes, is constrained to lie in a desired range, and formats the encoded data so that the individual subframes can be played back as they are received thereby reducing latency. Furthermore, the audio coder processes the baseband portion (0-24 kHz) of the audio band-width for sampling frequencies of 48 kHz and higher with the same encoding / decoding algorithm so that audio coder architecture is future compatible.

Description

[0001] This application is a divisional application of a Chinese patent application with the application number 03156927.7, the filing date being November 21, 1996, and the invention title being "multi-channel vocoder". technical field [0002] The present invention relates to high-quality encoding and decoding of multi-channel audio signals, and more particularly to a sub-band encoder using perfect / non-perfect Filter bank construction, predictive / non-predictive subband coding, transient analysis, and psychoacoustic / minimum mean square error (MMSE) bit-rate allocation are used to generate data streams whose corresponding decoding computations are constrained. Background technique [0003] Known high-quality audio and music encoders can be divided into two broad categories of schemes. The first category is a subband / transform coder with medium to high frequency resolution, which adaptively quantizes subband or coefficient samples within its analysis window based on psychoacou...

AI Technical Summary

Problems solved by technology

The second type is a sub-band encoder with low frequency resolution, which processes sub-band sampling data through ADPCM (adaptive differential pulse code modulation) to compensate for the lack of frequency resolution.

This method only provides some detection capability, and the errors in the audio data still cannot be found

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