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    METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS
    6.
    发明公开
    METHODS AND APPARATUSES FOR ENCODING AND DECODING OBJECT-BASED AUDIO SIGNALS 审中-公开
    用于编码和解码基于对象的音频信号的方法和设备

    公开(公告)号：EP2070080A1

    公开(公告)日：2009-06-17

    申请号：EP07833112.1

    申请日：2007-10-01

    申请人： LG Electronics, Inc.

    发明人： YOON, Sung Yong ,  PANG, Hee Suk ,  LEE, Hyun Kook ,  KIM, Dong Soo ,  LIM, Jae Hyun

    IPC分类号： G10L19/00

    CPC分类号： G10L19/008 ,  G10L19/087 ,  G10L19/20 ,  G10L21/04 ,  H04S7/302 ,  H04S2400/03 ,  H04S2400/11 ,  H04S2420/01

    摘要： Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating channel-based side information based on object-based side information and control information for rendering the downmix signal; processing the downmix signal using a decorrelated channel signal; and generating a multi-channel audio signal using the processed downmix signal and the channel-based side information.

    摘要翻译： 提供了一种音频编码方法和装置以及音频解码方法和装置，其中音频信号可以被编码或解码，使得声音图像可以被定位在每个对象音频信号的任何期望的位置。 该音频解码方法通过组合从第一音频信号提取的第一缩混信号和从第二音频信号提取的第二缩混信号来生成第三缩混信号; 通过组合从所述第一音频信号提取的第一基于对象的辅助信息和从所述第二音频信号提取的第二基于对象的辅助信息来产生第三基于对象的辅助信息; 将第三基于对象的辅助信息转换为基于信道的辅助信息; 以及使用第三下混合信号和基于信道的辅助信息来生成多声道音频信号。

    METHOD FOR MODELING SPEECH HARMONIC MAGNITUDES
    7.
    发明授权
    METHOD FOR MODELING SPEECH HARMONIC MAGNITUDES 有权
    方法建模谐波在语言数理科学

    公开(公告)号：EP1495465B1

    公开(公告)日：2006-06-07

    申请号：EP03745516.9

    申请日：2003-02-14

    申请人： Motorola, Inc., A Corporation of the State of Delaware

    发明人： RAMABADRAN, Tenkasi V., ,  SMITH, Aaron M., ,  JASIUK, Mark A.,

    IPC分类号： G10L19/02 ,  G10L19/04

    CPC分类号： G10L19/06 ,  G10L19/087

    摘要： A system or method for modeling a signal, such as a speech signal, wherein harmonic frequencies and amplitudes are identified (106) and the harmonic magnitudes are interpolated (110) to obtain spectral magnitudes at a set of fixed frequencies. An inverse transform is applied (112) to the spectral magnitudes to obtain a pseudo auto-correlation sequence, from which linear prediction coefficients are calculated (114). From the linear prediction coefficients, model harmonic magnitudes are generated by sampling the spectral envelope (118) defined by the linear prediction coefficients. A set of scale factors are then calculated (120) as the ratio of the harmonic magnitudes to the model harmonic magnitudes and interpolated to obtain a second set of scale factors (122) at the set of fixed frequencies. The spectral envelope magnitudes at the set of fixed frequencies (124) are multiplied by the second set of scale factors (126) to obtain new spectral magnitudes and the process is iterated to obtain final linear prediction coefficients.

    QUANTIZATION OF VARIABLE-DIMENSION SPEECH SPECTRAL AMPLITUDES USING SPECTRAL INTERPOLATION BETWEEN PREVIOUS AND SUBSEQUENT FRAMES
    9.
    发明公开
    QUANTIZATION OF VARIABLE-DIMENSION SPEECH SPECTRAL AMPLITUDES USING SPECTRAL INTERPOLATION BETWEEN PREVIOUS AND SUBSEQUENT FRAMES 审中-公开
    谱声THE QUANTIZE幅度变维BY之间的之前和之后的帧插值谱

    公开(公告)号：EP1183682A1

    公开(公告)日：2002-03-06

    申请号：EP00917636.3

    申请日：2000-03-13

    申请人： COMSAT Corporation

    发明人： YELDENER, Suat

    IPC分类号： G10L19/02 ,  G10L11/04

    CPC分类号： G10L19/06 ,  G10L19/087 ,  G10L25/12

    摘要： A speech coding algorithm interpolates groups speech frames into speech frame pairs, and quantizes each frame of the pair according to a different algorithm. The spectral amplitudes of the second frame are quantized by dividing them into two portions and quantizing one portion and then quantizing a difference between the two portions. The spectral amplitudes of the first frame of the pair are quantized by first converting to a fixed dimension, then interpolating between previous and subsequent frames, then selecting interpolated values in accordance with a mean squared error approach.

    Estimation of excitation parameters
    10.
    发明公开
    Estimation of excitation parameters 失效
    Abschätzungvon Anregungsparametern。

    公开(公告)号：EP0676744A1

    公开(公告)日：1995-10-11

    申请号：EP95302290.2

    申请日：1995-04-04

    申请人： DIGITAL VOICE SYSTEMS, INC.

    发明人： Griffin, Daniel Wayne ,  Lim, Jae S.

    IPC分类号： G10L9/14 ,  G10L9/06

    CPC分类号： G10L25/93 ,  G10L19/087 ,  G10L25/18 ,  G10L25/21

    摘要： Speech is encoded by analyzing a digitized speech signal to determine excitation parameters for the digitized speech signal. The digitized speech signal is divided into at least two frequency bands. A nonlinear operation is performed on at least one of the frequency bands to produce a modified frequency band. A determination is made as to whether the modified frequency band is voiced or unvoiced.

    摘要翻译： 通过分析数字化语音信号来确定语音，以确定数字化语音信号的激励参数。 数字化语音信号被分成至少两个频带。 对至少一个频带执行非线性运算以产生修改的频带。 确定修改的频带是有声还是无声。