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1. CN102265336 - Adaptive differential pulse code modulation encoding apparatus and decoding apparatus

Office China
Application Number 200980152966.4
Application Date 25.12.2009
Publication Number 102265336
Publication Date 30.11.2011
Publication Kind A
IPC
G PHYSICS
10
MUSICAL INSTRUMENTS; ACOUSTICS
L
SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
19
Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
G10L 19/00
CPC
G10L 19/04
H03M 7/3046
Applicants Kyushu Inst of Technology
国立大学法人九州工业大学
Inventors Sato Yasushi
佐藤宁
Ryu Atsuko
龙敦子
Agents liu xinyu
北京林达刘知识产权代理事务所(普通合伙) 11277
Priority Data 2008-333060 26.12.2008 JP
Title
(EN) Adaptive differential pulse code modulation encoding apparatus and decoding apparatus
(ZH) 自适应差分脉冲编码调制编码设备和解码设备
Abstract
(EN)
Provided are an ADPCM encoding apparatus and an ADPCM decoding apparatus wherein the compressibility can be enhanced and the degradation of sound quality can be prevented. Signals corresponding to the short-period and long-period changes of a sound signal are detected and an adaptive quantization characteristic is varied based on a combination of those two detected signals, thereby performing an appropriate quantization. In the ADPCM encoding apparatus (100), a subtracter (102) is used to calculate a differential value (dn) between a 16-bit input signal (Xn) and a decoded signal (Yn-1) of one sample before. Then, an adaptive quantizing unit (103) is used to adaptively quantize and convert the 16-bit differential value (dn) to length-variable ADPCM values (Dn) of one to eight bits. Then, a compression encoding unit (108) is used to compression encode the ADPCM values (Dn) to generate signals (D'n), which are then framed and outputted by use of a framing unit (130). In the ADPCM decoding apparatus, the framed input signals are decoded by performing the reverse of the foregoing processings.

(ZH)

提供了一种可以提高压缩率并且防止声音质量劣化的ADPCM编码设备和ADPCM解码设备。检测与声音信号的短周期变化和长周期变化相对应的信号,并且基于检测到的这两个信号的组合来改变自适应量化特性,由此进行适当的量化。在ADPCM编码设备(100)中,使用减法器(102)计算16位的输入信号(Xn)与一个样本之前的解码信号(Yn-1)之间的差分值(dn)。然后,使用自适应量化单元(103)进行自适应量化并且将16位的差分值(dn)转换成1~8位的可变长度的ADPCM值(Dn)。然后,使用压缩编码单元(108)对ADPCM值(Dn)进行压缩编码以生成信号(D’n),然后利用成帧单元(130)对信号(D’n)进行成帧并且输出。在ADPCM解码设备中,通过进行前述处理的逆处理来对成帧的输入信号进行解码。