Codebook sharing for LSF quantization
    1.
    发明授权
    Codebook sharing for LSF quantization 有权
    LSF量化的码本共享

    公开(公告)号:US08635063B2

    公开(公告)日:2014-01-21

    申请号:US12321950

    申请日:2009-01-26

    申请人: Yang Gao Eyal Shlomot

    发明人: Yang Gao Eyal Shlomot

    IPC分类号: G10L11/06

    摘要: In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure.

    摘要翻译: 根据本发明的一个方面,选择器基于输入语音信号的间隔中的触发特性的检测或不存在,支持选择第一编码方案或第二编码方案。 第一编码方案具有用于处理输入语音信号以形成偏向理想有声和静态特征的修正语音信号的音调预处理过程。 预处理过程允许编码器完全捕获带宽有效的长期预测程序的优点,用于输入语音信号的大量语音分量比否则可能的更多。 根据本发明的另一方面,第二编码方案需要一种长期预测模式,用于以子帧为基础对子帧上的音调进行编码。 长期预测模式被定制为语音的大致周期性分量通常不是静止的或小于完全周期性的,并且需要来自自适应码本的更高频率的更新以在长时间内实现再现语音的期望感知质量, 术语预测程序。

    Decoder with embedded silence and background noise compression
    2.
    发明授权
    Decoder with embedded silence and background noise compression 有权
    解码器具有嵌入式静音和背景噪声压缩

    公开(公告)号:US08195450B2

    公开(公告)日:2012-06-05

    申请号:US13199794

    申请日:2011-09-08

    IPC分类号: G10L21/00 G10L11/06

    摘要: There is provided a method for use by a speech encoder to encode an input speech signal. The method comprises receiving the input speech signal; determining whether the input speech signal includes an active speech signal or an inactive speech signal; low-pass filtering the inactive speech signal to generate a narrowband inactive speech signal; high-pass filtering the inactive speech signal to generate a high-band inactive speech signal; encoding the narrowband inactive speech signal using a narrowband inactive speech encoder to generate an encoded narrowband inactive speech; generating a low-to-high auxiliary signal by the narrowband inactive speech encoder based on the narrowband inactive speech signal; encoding the high-band inactive speech signal using a wideband inactive speech encoder to generate an encoded wideband inactive speech based on the low-to-high auxiliary signal from the narrowband inactive speech encoder; and transmitting the encoded narrowband inactive speech and the encoded wideband inactive speech.

    摘要翻译: 提供了一种由语音编码器用于对输入语音信号进行编码的方法。 该方法包括接收输入语音信号; 确定所述输入语音信号是否包括活动语音信号或无效语音信号; 低通滤波无效语音信号以产生窄带无效语音信号; 高通滤波无效语音信号以产生高频带无效语音信号; 使用窄带无源语音编码器对窄带无源语音信号进行编码,以生成编码窄带无效语音; 基于窄带无效语音信号,由窄带无源语音编码器生成低到高的辅助信号; 使用宽带无源语音编码器对高频带无效语音信号进行编码,以根据来自窄带无源语音编码器的低到高辅助信号产生编码的宽带无效语音; 以及发送编码的窄带无效语音和编码的宽带无效语音。

    Perceptual masking of residual echo
    3.
    发明授权
    Perceptual masking of residual echo 有权
    残余回声的感知掩蔽

    公开(公告)号:US07711107B1

    公开(公告)日:2010-05-04

    申请号:US11129450

    申请日:2005-05-12

    IPC分类号: H04M9/08

    CPC分类号: H04B3/234

    摘要: A method of masking a residual echo signal by an echo canceller is provided. The method comprises receiving a far-end signal, adjusting filter coefficients of an adaptive filter in response to the far-end signal, generating an echo model signal based on the far-end signal using the adaptive filter, receiving a near-end signal, subtracting the echo model signal from the near-end signal to generate an output signal, defining a spectral mask based on the near-end signal, wherein the spectral mask is indicative of near-end spectral peaks and near-end spectral valleys, de-emphasizing the output signal in spectral regions of the near-end spectral peaks, and emphasizing the output signal in spectral regions of the near-end spectral valleys, wherein the de-emphasizing occurs during filter coefficients determination for the adaptive filter. A weighted filter may perform the de-emphasizing and the emphasizing operations, where the weighted filter uses medium term spectral characteristics of the near-end signal.

    摘要翻译: 提供了一种通过回波消除器掩蔽残留回波信号的方法。 该方法包括接收远端信号,响应于远端信号调整自适应滤波器的滤波器系数,使用自适应滤波器基于远端信号生成回波模型信号,接收近端信号, 从近端信号减去回波模型信号以产生输出信号,基于近端信号定义频谱屏蔽,其中频谱掩模表示近端谱峰和近端谱谷, 强调近端光谱峰值的光谱区域中的输出信号,并且强调近端光谱谷的光谱区域中的输出信号,其中在自适应滤波器的滤波器系数确定期间发生去加重。 加权滤波器可以执行去强调和强调操作,其中加权滤波器使用近端信号的中期频谱特性。

    Embedded silence and background noise compression
    4.
    发明申请
    Embedded silence and background noise compression 有权
    嵌入式静音和背景噪声压缩

    公开(公告)号:US20080195383A1

    公开(公告)日:2008-08-14

    申请号:US12002131

    申请日:2007-12-14

    IPC分类号: G10L19/14

    摘要: There is provided a method for use by a speech encoder to encode an input speech signal. The method comprises receiving the input speech signal; determining whether the input speech signal includes an active speech signal or an inactive speech signal; low-pass filtering the inactive speech signal to generate a narrowband inactive speech signal; high-pass filtering the inactive speech signal to generate a high-band inactive speech signal; encoding the narrowband inactive speech signal using a narrowband inactive speech encoder to generate an encoded narrowband inactive speech; generating a low-to-high auxiliary signal by the narrowband inactive speech encoder based on the narrowband inactive speech signal; encoding the high-band inactive speech signal using a wideband inactive speech encoder to generate an encoded wideband inactive speech based on the low-to-high auxiliary signal from the narrowband inactive speech encoder; and transmitting the encoded narrowband inactive speech and the encoded wideband inactive speech.

    摘要翻译: 提供了一种由语音编码器用于对输入语音信号进行编码的方法。 该方法包括接收输入语音信号; 确定所述输入语音信号是否包括活动语音信号或无效语音信号; 低通滤波无效语音信号以产生窄带无效语音信号; 高通滤波无效语音信号以产生高频带无效语音信号; 使用窄带无源语音编码器对窄带无源语音信号进行编码,以生成编码窄带无效语音; 基于窄带无效语音信号,由窄带无源语音编码器生成低到高的辅助信号; 使用宽带无源语音编码器对高频带无效语音信号进行编码,以根据来自窄带无源语音编码器的低到高辅助信号产生编码的宽带无效语音; 以及发送编码的窄带无效语音和编码的宽带无效语音。

    Adaptive voice mode extension for a voice activity detector
    5.
    发明申请
    Adaptive voice mode extension for a voice activity detector 有权
    语音活动检测器的自适应语音模式扩展

    公开(公告)号:US20060217973A1

    公开(公告)日:2006-09-28

    申请号:US11342104

    申请日:2006-01-26

    IPC分类号: G10L19/12

    CPC分类号: G10L25/78 G10L2025/786

    摘要: There is provided a voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving a first portion of an input signal; determining that the first portion of the input signal includes an active voice signal; indicating the active voice mode in response to the determining that the first portion of the input signal includes the active voice signal; receiving a second portion of the input signal immediately following the first portion of the input signal; determining that the second portion of the input signal includes an inactive voice signal; extending the indicating the active voice mode for a period of time after determining that the second portion of the input signal includes the inactive voice signal, wherein the period of time varies based on one or more conditions; and indicating the inactive voice mode after expiration of the period of time.

    摘要翻译: 提供了一种用于指示主动语音模式和无效语音模式的语音活动检测方法。 该方法包括接收输入信号的第一部分; 确定输入信号的第一部分包括有效语音信号; 响应于确定输入信号的第一部分包括有效语音信号,指示主动语音模式; 接收紧接在输入信号的第一部分之后的输入信号的第二部分; 确定输入信号的第二部分包括不活动的语音信号; 在确定所述输入信号的第二部分包括所述不活动语音信号之后,将所述主动语音模式指示一段时间,其中所述时间段基于一个或多个条件而变化; 并且在该时间段期满之后指示不活动的语音模式。

    Codebook tables for multi-rate encoding and decoding with pre-gain and delayed-gain quantization tables
    6.
    发明授权
    Codebook tables for multi-rate encoding and decoding with pre-gain and delayed-gain quantization tables 有权
    用于具有预增益和延迟增益量化表的多速率编码和解码的码表

    公开(公告)号:US06757649B1

    公开(公告)日:2004-06-29

    申请号:US10409404

    申请日:2003-04-08

    IPC分类号: G10L1912

    摘要: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

    摘要翻译: 公开了能够将语音信号编码为比特流以进行后续解码以产生合成语音的语音压缩系统。 语音压缩系统通过将期望的平均比特率与重构语音的感知质量进行平衡来优化比特流消耗的带宽。 语音压缩系统包括全速率编解码器,半速率编解码器,四分之一速率编解码器和八速率编解码器。 基于速率选择来选择性地激活编解码器。 此外,基于类型分类,全速率和半速率编解码器被选择性地激活。 选择性地激活每个编解码器以以强调语音信号的不同方面的不同比特率对语音信号进行编码和解码,以增强合成语音的整体质量。

    Embedded silence and background noise compression
    7.
    发明授权
    Embedded silence and background noise compression 有权
    嵌入式静音和背景噪声压缩

    公开(公告)号:US08032359B2

    公开(公告)日:2011-10-04

    申请号:US12002131

    申请日:2007-12-14

    IPC分类号: G10L21/00 G10L11/06

    摘要: There is provided a method for use by a speech encoder to encode an input speech signal. The method comprises receiving the input speech signal; determining whether the input speech signal includes an active speech signal or an inactive speech signal; low-pass filtering the inactive speech signal to generate a narrowband inactive speech signal; high-pass filtering the inactive speech signal to generate a high-band inactive speech signal; encoding the narrowband inactive speech signal using a narrowband inactive speech encoder to generate an encoded narrowband inactive speech; generating a low-to-high auxiliary signal by the narrowband inactive speech encoder based on the narrowband inactive speech signal; encoding the high-band inactive speech signal using a wideband inactive speech encoder to generate an encoded wideband inactive speech based on the low-to-high auxiliary signal from the narrowband inactive speech encoder; and transmitting the encoded narrowband inactive speech and the encoded wideband inactive speech.

    摘要翻译: 提供了一种由语音编码器用于对输入语音信号进行编码的方法。 该方法包括接收输入语音信号; 确定所述输入语音信号是否包括活动语音信号或无效语音信号; 低通滤波无效语音信号以产生窄带无效语音信号; 高通滤波无效语音信号以产生高频带无效语音信号; 使用窄带无源语音编码器对窄带无源语音信号进行编码,以生成编码窄带无效语音; 基于窄带无效语音信号,由窄带无源语音编码器生成低到高的辅助信号; 使用宽带无源语音编码器对高频带无效语音信号进行编码,以根据来自窄带无源语音编码器的低到高辅助信号产生编码的宽带无效语音; 以及发送编码的窄带无效语音和编码的宽带无效语音。

    Speech compression system and method
    8.
    发明授权
    Speech compression system and method 有权
    语音压缩系统及方法

    公开(公告)号:US07593852B2

    公开(公告)日:2009-09-22

    申请号:US11700481

    申请日:2007-01-30

    IPC分类号: G10L15/20

    摘要: The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

    摘要翻译: 本发明通过将编码聚焦在语音的重要特征上来改进语音的编码和解码。 该系统分析输入语音信号的所选特征,并且首先对输入语音信号进行基于公共帧的语音编码。 然后,该系统基于第一语音编码模式或第二语音编码模式执行语音编码。 模式的选择基于输入语音信号的特性。 第一语音编码模式使用第一成帧结构,第二语音编码模式使用第二帧结构。

    Method and system for reducing effects of noise producing artifacts in a voice codec
    9.
    发明授权
    Method and system for reducing effects of noise producing artifacts in a voice codec 有权
    用于减少语音编解码器中噪声产生伪像的影响的方法和系统

    公开(公告)号:US07454335B2

    公开(公告)日:2008-11-18

    申请号:US11385553

    申请日:2006-03-20

    申请人: Yang Gao Eyal Shlomot

    发明人: Yang Gao Eyal Shlomot

    IPC分类号: G10L21/02 G10L19/06

    摘要: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1−k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

    摘要翻译: 提供了一种减少由语音解码系统使用的语音信号的静音区域中产生噪声的噪声的影响的方法。 该方法包括获得语音子帧的多个输入样本; 将多个输入样本中的每一个的能级的绝对值求和以产生总输入电平(gain_in); 平滑总输入电平以产生平滑电平(Level_in_sm); 基于总输入电平,平滑电平和频谱倾斜参数,确定语音子帧在静音区域中; 使用k1 *(Level_in_sm / 1024)+(1-k1)定义增益,其中K1是频谱倾斜参数的函数; 以及使用所述增益来修改所述语音子帧的能级。

    Method and system for reducing effects of noise producing artifacts in a voice codec
    10.
    发明申请
    Method and system for reducing effects of noise producing artifacts in a voice codec 有权
    用于减少语音编解码器中噪声产生伪像的影响的方法和系统

    公开(公告)号:US20070219791A1

    公开(公告)日:2007-09-20

    申请号:US11385553

    申请日:2006-03-20

    申请人: Yang Gao Eyal Shlomot

    发明人: Yang Gao Eyal Shlomot

    IPC分类号: G10L15/20

    摘要: There is provided a method of reducing effect of noise producing artifacts in silence areas of a speech signal for use by a speech decoding system. The method comprises obtaining a plurality of incoming samples of a speech subframe; summing an absolute value of an energy level for each of the plurality of incoming samples to generate a total input level (gain_in); smoothing the total input level to generate a smoothed level (Level_in_sm); determining that the speech subframe is in a silence area based on the total input level, the smoothed level and a spectral tilt parameter; defining a gain using k1*(Level_in_sm/1024)+(1-k1), where K1 is a function of the spectral tilt parameter; and modifying an energy level of the speech subframe using the gain.

    摘要翻译: 提供了一种减少由语音解码系统使用的语音信号的静音区域中产生噪声的噪声的影响的方法。 该方法包括获得语音子帧的多个输入样本; 将多个输入样本中的每一个的能级的绝对值求和以产生总输入电平(gain_in); 平滑总输入电平以产生平滑电平(Level_in_sm); 基于总输入电平,平滑电平和频谱倾斜参数,确定语音子帧在静音区域中; 使用k1 *(Level_in_sm / 1024)+(1-k1)定义增益,其中K1是频谱倾斜参数的函数; 以及使用所述增益来修改所述语音子帧的能级。