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Decoder for decoding and reestablishing multi-channel audio signal from audio data code stream

An audio signal and audio data technology, which is applied in the direction of pulse modulation TV signal transmission, code conversion, high-definition TV system, etc., can solve the problems of difficult sound quality, inability to guarantee, complex and expensive encoding and decoding, etc.

Inactive Publication Date: 2010-05-12
广州广晟数码技术有限公司
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  • Summary
  • Abstract
  • Description
  • Claims
  • Application Information

AI Technical Summary

Problems solved by technology

It can be seen from the above that there are several difficulties in this TV station method: 1) The distortion of the audio signal is large: the sampling rate conversion of Dolby E itself introduces distortion, and the code stream from the Dolby E format to the code stream of the AC-3 format is transferred Encoding (Transcoding) introduces new distortion
2) It is difficult to reuse the programs that have been launched: if figure 1 As shown, if the program that has been transmitted is to be reused, it must be decoded into PCM and then encoded by Dolby E before it can be switched with other programs (such as advertisements, etc.)
Since the code rate for the final transmission is generally not high, the sound quality of the transmitted programs after the above series of transfer encoding (AC-3 decoding -> Dolby E encoding -> Dolby E decoding -> AC-3 encoding) is very low. difficult or impossible to guarantee
3) Since the sampling rate of Dolby E's input and output has no simple frequency multiplication relationship, its encoding and decoding are very complicated and expensive

Method used

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  • Decoder for decoding and reestablishing multi-channel audio signal from audio data code stream
  • Decoder for decoding and reestablishing multi-channel audio signal from audio data code stream
  • Decoder for decoding and reestablishing multi-channel audio signal from audio data code stream

Examples

Experimental program
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Embodiment 1

[0047] Such as figure 2 As shown, the encoder of the present invention is composed of a frame length selector 1, a sub-band decomposition filter bank 2, a transient detector 3, a bit allocator 4, a sub-band quantizer 5, and a multiplexer 6. It is used to compress and encode the input audio signal. The audio signal may be an audio signal of a digital television. The following describes the working principle of the encoder through a detailed description of each component.

[0048] Frame length selector 1:

[0049] The function of frame length selector 1 is to divide the input audio signal into frames according to a certain frame length. As for the selection of frame length, it is generally believed that the larger the frame length, the higher the encoding compression efficiency, but the greater the encoding delay, the encoder and decoding Therefore, when the code rate is high, a smaller frame length is generally selected, and when the code rate is low, a larger frame lengt...

Embodiment 2

[0093] The encoder of the second embodiment of the present invention is as Figure 4 shown. Most of its components are the same as those in Embodiment 1, the difference lies in the unique bit allocation method of this embodiment. Specifically, the bit allocation in this embodiment is not as in Embodiment 1 to allocate bits for each subband segment according to the signal-to-noise ratio (SNR) or the signal-to-coverage ratio (SMR), but according to the scale factor quantizer 52 output The quantization index of the scale factor uses the following formula to allocate bits for each subband segment.

[0094] b(c,k,d)=f(α·s(c,k,d))-θ(k)-β (3)

[0095] in:

[0096] 1) b(c, k, d) is the number of bits per sample allocated to the current subband segment.

[0097] 2) f(α·s(c, k, d)) is a strictly monotonically increasing function. It can preferably be set as f(α·s(c,k,d))=[α·s(c,k,d)] q , where 0

[0098] 3) θ...

Embodiment 3

[0115] The encoder of the third embodiment of the present invention is as Figure 6 shown. Most of its components are the same as those of other embodiments, the difference is that this embodiment has one more joint strength encoder 7 than other embodiments. The theoretical basis of joint intensity coding is that the human ear's spatial localization of sound at high frequencies (such as above 7 kHz) is mainly based on the intensity of the sound. Such as Figure 6 As shown, when encoding, the joint intensity encoder 7 can add up the high-frequency subbands output by the transient detector 3 of the left and right (or other joint) channels, and only pass this sum subband (referred to as the source sound The quantization index of each sample of the jointly coded subband of the channel is added to the intensity index of the combined subband of the channel to achieve the purpose of saving bits.

[0116]When joint intensity coding is used, the allocation of bits to jointly coded s...

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Abstract

This invention relates to a device and a method for compressing code and decode to multiple channel digital audio signals, in which, the code device includes: a frame length selector, a sub-band resolving filter set, a transient detector, a proportion factor evaluator, a bit distributor used in distributing the bit resource decided by the target code rate to the sub-band sections adaptively, a sub-band quantizer and a multiple multiplexer, in which, the compressing efficiency of the device and the method is high, the audio signal can be edited synchronously with video signal and endures more than ten times of tandem code so as to meet the needs of distributing and transmitting programs and reduce the processes of program distribution and devices.

Description

[0001] This application is a divisional application submitted on August 21, 2002, with application number 02130245.6. technical field [0002] The present invention relates to a digital audio signal encoding / decoding device and its method, more precisely, to a multi-channel digital audio signal compression encoding / decoding device and method. Background technique [0003] Multi-channel (including stereo) digital audio compression coding technology has been widely used in VCD, SVCD, DVD, satellite TV, digital TV, and Internet (Internet) and other fields. The main problem it needs to solve is that the code rate for expressing multi-channel digital audio signals is very high, but the channel capacity that can be used to propagate or store it is very limited. For example, using PCM to express 5.1-channel surround sound with 48kHz sampling rate and 24 bits per sample requires a code rate of 6912kbps (kilobits per second), and applications with limited channel capacity such as dig...

Claims

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Application Information

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Patent Type & Authority Patents(China)
IPC IPC(8): H03M7/30H04N7/015H04N7/24
Inventor 游余立
Owner 广州广晟数码技术有限公司
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