Device, method, and program for encoding/decoding of speech with function of encoding silent period
    1.
    发明授权
    Device, method, and program for encoding/decoding of speech with function of encoding silent period 有权
    具有编码静音期功能的语音编码/解码的装置,方法和程序

    公开(公告)号:US08195469B1

    公开(公告)日:2012-06-05

    申请号:US09980275

    申请日:2000-05-31

    IPC分类号: G10L19/00 G10L21/04

    CPC分类号: G10L19/012 G10L2019/0012

    摘要: A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.

    摘要翻译: 本发明的语音解码装置在解密无语音周期的语音信号中,对不连续发送的RMS和滤波器系数进行平滑化,并提供给合成滤波器。 由此,能够防止由过滤器系数的间歇传递引起的过滤器系数的不连续变化。 结果,可以提高解码质量。 另外,为了消除由平滑处理引起的效果,根据在过去的帧中发送的滤波器系数或RMS,调整平滑因子而不在一定时间段(或一定数量的帧)下进行平滑化, 从当从语音周期的语音周期转变时,或者当解码的特征参数满足预定条件时。

    Encoder for multiplexing blocks of error protected bits with blocks of unprotected bits
    2.
    发明授权
    Encoder for multiplexing blocks of error protected bits with blocks of unprotected bits 失效
    用于将错误保护位的块与未保护位的块复用的编码器

    公开(公告)号:US06532564B1

    公开(公告)日:2003-03-11

    申请号:US09499218

    申请日:2000-02-07

    IPC分类号: H03M1300

    CPC分类号: H04L1/0057 H04L1/007

    摘要: An error detection encoder comprises separation circuitry for separating an input signal into a first sequence of error protected bits and a second sequence of error unprotected bits. Calculation circuitry produces an error check sequence from the first sequence and concatenates the error check sequence to the first sequence to produce a third sequence. The second sequence may be further separated into a first sub-sequence of higher significant bits and a second sub-sequence of lower significant bits. A multiplexer is provided for segmenting the third sequence into a plurality of first blocks and segmenting the first sub-sequence into a plurality of second blocks corresponding to the first blocks and multiplexing each of the first blocks with a corresponding one of the second blocks to produce a fourth sequence in which the first and the second blocks are arranged in an alternating order. The second sub-sequence is concatenated to the fourth sequence to produce an output sequence for transmission.

    摘要翻译: 误差检测编码器包括分离电路,用于将输入信号分离成第一错误保护位序列和第二错误未保护位序列。 计算电路从第一个序列产生错误检查序列,并将错误检查序列连接到第一个序列以产生第三个序列。 第二序列可以进一步分为较高有效位的第一子序列和低有效位的第二子序列。 提供了多路复用器,用于将第三序列分割成多个第一块,并将第一子序列分割成与第一块相对应的多个第二块,并且将第一块中的每一个与第二块中的相应一个进行多路复用以产生 其中第一和第二块以交替顺序排列的第四序列。 第二子序列连接到第四序列以产生用于传输的输出序列。

    Speech encoding method and speech encoding system
    3.
    发明授权
    Speech encoding method and speech encoding system 有权
    语音编码方法和语音编码系统

    公开(公告)号:US06581031B1

    公开(公告)日:2003-06-17

    申请号:US09450305

    申请日:1999-11-29

    IPC分类号: G10L1912

    CPC分类号: G10L19/08 G10L19/09

    摘要: In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal Xw(n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle Top, then outputting at least one pitch cycle Top to the adaptive codebook circuit. The adaptive codebook circuit is input with the perceptual weighting signal x′w(n), the past excitation signal v(n) output from the gain quantization circuit, the perceptual weighting impulse response hw(n) output from the impulse response calculation circuit, and the pitch cycle Top from the pitch calculation circuit, searching near the pitch cycle, calculating the delay of adaptive codebook. With the above composition, the delay of adaptive codebook obtained for each subframe can be prevented from being discontinuous in the process of time.

    摘要翻译: 在该语音编码系统中,限制电路以前一子帧获得的自适应码本的延迟输入,并且音调周期搜索范围被限制,使得对于前一个子帧获得的自适应码本的延迟不延迟 对于当前子帧获得的自适应码本,并且将音调周期搜索范围限制输出到音调计算电路。 音调计算电路输入感知加权电路的输出信号Xw(n)和从限制器输出的音调周期搜索范围,计算音调周期Top,然后将至少一个音调周期Top输出到自适应码本电路。 自适应码本电路输入感知加权信号x'w(n),从增益量化电路输出的过去激励信号v(n),从脉冲响应计算电路输出的感知加权脉冲响应hw(n) 和音调周期Top从音调计算电路,在音调周期附近搜索,计算自适应码本的延迟。 利用上述构成,可以防止在每个子帧中获得的自适应码本的延迟在时间过程中不连续。

    Transmission channel error detection code addition apparatus and error detection apparatus
    4.
    发明授权
    Transmission channel error detection code addition apparatus and error detection apparatus 失效
    传输通道错误检测码附加装置和错误检测装置

    公开(公告)号:US06625779B1

    公开(公告)日:2003-09-23

    申请号:US09510900

    申请日:2000-02-23

    IPC分类号: H03M1300

    摘要: A transmission channel error detection code addition apparatus includes a division section, calculation section, and transmission section. The division section divides a code bitstream obtained from input information having a coding unit time length which is an N integer multiple (N>=2) of an error detection unit time length into N bitstreams. The calculation section calculates an error detection code of each divided bitstream output from the division section. The transmission section sequentially transmits each divided bitstream output from the division section and the error detection code output from the calculation section in correspondence with each divided bitstream.

    摘要翻译: 传输通道错误检测码附加装置包括分割部分,计算部分和传输部分。 分割部将从具有错误检测单位时间长度的N个整数倍(N> = 2)的编码单位时间长度的输入信息获得的码位流分割为N个比特流。 计算部分计算从分割部分输出的每个划分的比特流的错误检测码。 发送部分根据每个划分的比特流顺序地发送从分割部分输出的每个划分的比特流和从计算部分输出的错误检测码。

    Voice mixing device, noise suppression method and program therefor
    6.
    发明授权
    Voice mixing device, noise suppression method and program therefor 有权
    语音混音装置,噪声抑制方法及程序

    公开(公告)号:US08428939B2

    公开(公告)日:2013-04-23

    申请号:US12670843

    申请日:2008-07-28

    IPC分类号: G10L21/00

    CPC分类号: G10L21/0208 H04M3/56

    摘要: A voice mixing device for mixing a plurality of voice signals, comprises: a speaker selection unit selecting at least one voice signal among said plurality of voice signals; a full signal adder unit adding all of at least one voice signal selected by said speaker selection unit; respective subtractor unit subtracting only one of said selected voice signals from an addition result of said full signal adder unit; a common noise suppression unit suppressing noise of a common voice signal, being an addition result of said full signal adder unit; individual noise suppression unit suppressing noise of respective individual voice signals, being subtraction results of said subtractor unit; and memory switching unit copying information of noise suppression obtained in said common noise suppression unit based on a selection result of said speaker selection unit, to information of noise suppression in said individual noise suppression unit.

    摘要翻译: 一种用于混合多个语音信号的语音混合装置,包括:扬声器选择单元,在所述多个语音信号中选择至少一个语音信号; 全信号加法器单元,添加由所述扬声器选择单元选择的所有至少一个语音信号; 各个减法器单元从所述全信号加法器单元的相加结果中减去所述所选语音信号中的一个; 公共噪声抑制单元,抑制公共语音信号的噪声,作为所述全信号加法单元的相加结果; 个别噪声抑制单元,抑制各个声音信号的噪声,作为所述减法器单元的减法结果; 以及存储器切换单元,基于所述扬声器选择单元的选择结果将在所述公共噪声抑制单元中获得的噪声抑制的信息复制到所述各个噪声抑制单元中的噪声抑制信息。

    SOUND MIXING APPARATUS AND METHOD AND MULTIPOINT CONFERENCE SERVER
    7.
    发明申请
    SOUND MIXING APPARATUS AND METHOD AND MULTIPOINT CONFERENCE SERVER 有权
    声音混合设备和方法与多点会议服务器

    公开(公告)号:US20100290645A1

    公开(公告)日:2010-11-18

    申请号:US12812135

    申请日:2009-01-28

    IPC分类号: H04B1/00

    CPC分类号: H04M3/56 G10L19/00

    摘要: A sound mixing apparatus includes mixing processing units 11 to 1k provided according to sampling frequencies. Each of mixing processing units 11 to 1k adds up input sound signals of the same sampling frequency to generate a first added-up sound signal, converts the sampling frequency of the first added-up sound signal into a sampling frequency processable by the other mixing processing units, and supplies sound signals that are converted to the sampling frequency, to the other mixing processing units, adds up, to generate a second added-up sound signal, the first added-up sound signal generated by itself and the first added-up sound signals that are converted to a sampling frequency processable by itself and that are supplied from the other mixing processing units.

    摘要翻译: 声音混合装置包括根据采样频率提供的混合处理单元11至1k。 混合处理单元11至1k中的每一个将具有相同采样频率的输入声音信号相加以产生第一相加声音信号,将第一相加声音信号的采样频率转换为可由另一混合处理处理的采样频率 单元,并将被转换为采样频率的声音信号提供给其他混合处理单元,相加,以产生第二相加声音信号,由其自身产生的第一相加声音信号和第一相加 声音信号被转换成可以自己处理并从其他混合处理单元提供的采样频率。

    SOUND DATA DECODING APPARATUS
    8.
    发明申请
    SOUND DATA DECODING APPARATUS 有权
    声音数据解码设备

    公开(公告)号:US20100005362A1

    公开(公告)日:2010-01-07

    申请号:US12309597

    申请日:2007-07-23

    IPC分类号: H03M13/05 G06F11/10

    CPC分类号: G10L19/005

    摘要: A sound data decoding apparatus based on a waveform coding method includes a loss detector, sound data decoder, sound data analyzer, parameter modifying section and sound synthesizing section. The loss detector detects whether a loss exists in a sound data. The sound data decoder decodes the sound data to generate a first decoded sound signal. The sound data analyzer extracts a first parameter from the first decoded sound signal. The parameter modifying section modifies the first parameter based on a result of the detection of loss. The sound synthesizing section generates a first synthesized sound signal by using the modified first parameter. Thus, a deterioration of sound quality is prevented in the error compensation of sound data.

    摘要翻译: 一种基于波形编码方法的声音数据解码装置,包括损失检测器,声音数据解码器,声音数据分析器,参数修正部分和声音合成部分。 损失检测器检测声音数据中是否存在损失。 声音数据解码器对声音数据进行解码以产生第一解码声音信号。 声音数据分析器从第一解码声音信号中提取第一参数。 参数修改单元根据损失检测的结果修改第一个参数。 声音合成部通过使用修改的第一参数来生成第一合成声音信号。 因此,在声音数据的误差补偿中防止声音质量的恶化。

    VOICE MIXING METHOD AND MULTIPOINT CONFERENCE SERVER AND PROGRAM USING THE SAME METHOD
    9.
    发明申请
    VOICE MIXING METHOD AND MULTIPOINT CONFERENCE SERVER AND PROGRAM USING THE SAME METHOD 有权
    语音混合方法和多点会议服务器和程序使用相同的方法

    公开(公告)号:US20090248402A1

    公开(公告)日:2009-10-01

    申请号:US12438659

    申请日:2007-08-28

    IPC分类号: G10L19/00

    摘要: The voice mixing method includes a first step for selecting voice information from a plurality of voice information, a second step for adding up all the selected voice information, a third step for obtaining a voice signal totaling the voice signals other than one voice signal, of the selected voice signals, a fourth step for encoding the voice information obtained in the second step, a fifth step for encoding the voice signal obtained in the third step, and a sixth step for copying the encoded information obtained in the fourth step into the encoded information in the fifth step.

    摘要翻译: 语音混合方法包括:从多个语音信息中选择语音信息的第一步骤,将所有所选语音信息相加的第二步骤,用于获得合成除了一个语音信号之外的语音信号的语音信号的第三步骤; 所选择的语音信号,用于编码在第二步骤中获得的语​​音信息的第四步骤,用于编码在第三步骤中获得的语​​音信号的第五步骤,以及第六步骤,用于将在第四步骤中获得的编码信息复制到编码的 信息在第五步。