公开(公告)号：US06188981B1

公开(公告)日：2001-02-13

申请号：US09156416

申请日：1998-09-18

申请人： Adil Benyassine ,  Eyal Shlomot

发明人： Adil Benyassine ,  Eyal Shlomot

IPC分类号： G10L1500

CPC分类号： G10L25/78

摘要： A method and apparatus for generating frame voicing decisions for an incoming speech signal having periods of active voice and non-active voice for a speech encoder in a speech communications system. A predetermined set of parameters is extracted from the incoming speech signal, including a pitch gain and a pitch lag. A frame voicing decision is made for each frame of the incoming speech signal according to values calculated from the extracted parameters. The predetermined set of parameters further includes a frame full band energy, and a set of spectral parameters called Line Spectral Frequencies (LSF).

摘要翻译： 一种用于在语音通信系统中为语音编码器的有效语音和非有效语音周期的输入语音信号生成帧语音决定的方法和装置。 从输入语音信号中提取预定的一组参数，包括音调增益和音调滞后。 根据从提取的参数计算的值，对输入语音信号的每个帧进行帧发声决定。 预定的参数集还包括帧全频带能量和称为线谱频率（LSF）的一组频谱参数。

公开(公告)号：US07711108B2

公开(公告)日：2010-05-04

申请号：US11072476

申请日：2005-03-03

申请人： Huan-Yu Su ,  Adil Benyassine ,  Nick J. Lavrov

发明人： Huan-Yu Su ,  Adil Benyassine ,  Nick J. Lavrov

IPC分类号： H04M9/08

CPC分类号： H04B3/234

摘要： A method of adjusting an echo canceller comprises obtaining a first cross-correlation between a far-end signal and an error signal, wherein the error signal is generated by subtracting an output signal of an adaptive filter from a local-end signal; determining whether the first cross-correlation is above a pre-determined threshold; relocating the adaptive filter by a few samples if the determining determines that the first cross-correlation is above a pre-determined threshold; calculating a first improvement indicator parameter, wherein the first improvement indicator parameter is calculated after the relocating the adaptive filter by the few samples; determining whether the first improvement indicator parameter indicates a performance improvement by the adaptive filter after the relocating the adaptive filter by the few samples; calculating a gain based on the local-end signal and the error signal if the determining does not determine the performance improvement; and multiplying the adaptive filter by the gain.

摘要翻译： 一种调整回波消除器的方法包括获得远端信号和误差信号之间的第一互相关，其中通过从本地端信号中减去自适应滤波器的输出信号来产生误差信号; 确定所述第一互相关是否高于预定阈值; 如果确定确定第一互相关高于预定阈值，则将自适应滤波器重定位几个样本; 计算第一改进指标参数，其中在通过所述少数样本重定位所述自适应滤波器之后计算所述第一改进指标参数; 在由所述少数样本重新定位所述自适应滤波器之后，确定所述第一改进指示符参数是否指示所述自适应滤波器的性能改善; 如果确定不确定性能改进，则基于本地端信号和误差信号计算增益; 并将自适应滤波器乘以增益。

公开(公告)号：US06694293B2

公开(公告)日：2004-02-17

申请号：US09782883

申请日：2001-02-13

申请人： Adil Benyassine ,  Huan-Yu Su

发明人： Adil Benyassine ,  Huan-Yu Su

IPC分类号： G10L1902

CPC分类号： G10L19/18 ,  G10L25/78 ,  G10L2025/783

摘要： The invention provides a speech coding system with a music classifier. An encoder is disposed to receive an input signal and provides a bitstream based upon a speech coding of a portion of the input signal. The encoder provides a classification of the input signal as one of noise, speech, and music. The music classifier analyzes or determines signal properties of the input signal. The music classifier compares the signal properties to thresholds to determine the classification of the input signal.

摘要翻译： 本发明提供一种具有音乐分类器的语音编码系统。 编码器被设置为接收输入信号并且基于输入信号的一部分的语音编码来提供比特流。 编码器将输入信号分类为噪声，语音和音乐之一。 音乐分类器分析或确定输入信号的信号特性。 音乐分类器将信号特性与阈值进行比较，以确定输入信号的分类。

公开(公告)号：US07613291B1

公开(公告)日：2009-11-03

申请号：US11201637

申请日：2005-08-10

申请人： Adil Benyassine ,  Carlo Murgia

发明人： Adil Benyassine ,  Carlo Murgia

IPC分类号： H04M9/08

CPC分类号： H04B3/237 ,  H04B3/234

摘要： There is provided a method for use by an echo canceller to detect an echo path change and adjust to the echo path change. The method comprises determining a first bulk delay using a SPARSE foreground adaptive filter; configuring the foreground adaptive filter to an open-loop mode; canceling the echo signal based on the first bulk delay using the foreground adaptive filter; determining a second bulk delay of the echo signal using a SPARSE background adaptive filter; configuring the foreground adaptive filter to a closed-loop mode and continuing to cancel the echo signal based on the first bulk delay; configuring the background adaptive filter to the open-loop mode; measuring echo cancellation performance of the foreground adaptive filter and the background adaptive filter; and changing parameters of the foreground adaptive filter if the echo cancellation performance of the background adaptive filter is better than the foreground adaptive filter.

摘要翻译： 提供了一种由回波消除器使用的方法来检测回波路径变化并适应回波路径变化。 该方法包括使用SPARSE前景自适应滤波器来确定第一批量延迟; 将前景自适应滤波器配置为开环模式; 使用前景自适应滤波器基于第一批量延迟来消除回波信号; 使用SPARSE背景自适应滤波器确定回波信号的第二批量延迟; 将前景自适应滤波器配置为闭环模式，并且基于第一批量延迟继续抵消回波信号; 将背景自适应滤波器配置为开环模式; 测量前景自适应滤波器和背景自适应滤波器的回波消除性能; 如果背景自适应滤波器的回波消除性能优于前景自适应滤波器，则改变前景自适应滤波器的参数。

公开(公告)号：US08620647B2

公开(公告)日：2013-12-31

申请号：US12321935

申请日：2009-01-26

申请人： Yang Gao ,  Adil Benyassine

发明人： Yang Gao ,  Adil Benyassine

IPC分类号： G10L11/06

CPC分类号： G10L19/12 ,  G10L19/0204 ,  G10L19/09 ,  G10L19/18 ,  G10L19/20 ,  G10L25/90 ,  G10L2019/0002 ,  G10L2019/0016

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.

公开(公告)号：US06782360B1

公开(公告)日：2004-08-24

申请号：US09574396

申请日：2000-05-19

申请人： Yang Gao ,  Adil Benyassine

发明人： Yang Gao ,  Adil Benyassine

IPC分类号： G10L1904

CPC分类号： G10L19/00 ,  G10L19/083 ,  G10L19/167 ,  G10L19/18 ,  G10L19/24 ,  H03G3/00

摘要： A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization. The result of quantizing gains of periodic speech in this manner results in a reduction of the number of bits required to represent the quantized gain information and for periodic speech, the ability to use the quantized pitch gain for the current subframe to search the fixed codebook for the fixed codebook excitation vector for the current subframe. Alternatively, the new gain quantization process which was used only for periodic signals may be extended to non-periodic signals as well. This second strategy results in a slightly higher bit rate than that for periodic signals that use the new gain quantization strategy, but is still lower than the prior art's bit rate. Yet another alternative is to use the new gain quantization process for all speech signals without distinguishing between periodic and non-periodic signals.

摘要翻译： 语音编码器，其将每个语音帧分析和分类为周期性语音或非周期性类似语音，其中语音编码器根据语音是否周期性执行不同的增益量化处理。 如果语音是周期性的，则改进的语音编码器从非量化加权语音信号获得音调增益，并且在子帧处理开始之前针对帧的每个子帧执行自适应码本增益GP的前向量化，并且执行闭环延迟判定 固定码本增益GC的矢量量化。 如果语音是非周期性的，语音编码器可以使用任何已知的增益量化方法。 以这种方式量化周期性语音的增益的结果导致表示量化增益信息所需的位数减少，对于周期性语音，使用当前子帧的量化音调增益来搜索固定码本的能力 用于当前子帧的固定码本激励矢量。 或者，仅用于周期性信号的新增益量化处理也可以扩展到非周期信号。 该第二策略比使用新的增益量化策略的周期信号的比特率稍高，但是仍低于现有技术的比特率。 另一个替代方案是对所有语音信号使用新的增益量化处理，而不区分周期性和非周期性信号。

公开(公告)号：US06633841B1

公开(公告)日：2003-10-14

申请号：US09526017

申请日：2000-03-15

申请人： Jes Thyssen ,  Adil Benyassine

发明人： Jes Thyssen ,  Adil Benyassine

IPC分类号： G10L1520

CPC分类号： G10L19/18 ,  G10H2210/046 ,  G10L25/78

摘要： An extended signal coding system that accommodates substantially music-like signals within a signal while maintaining a high perceptual quality in a reproduced signal during discontinued transmission (DTX) operation. The extended signal coding system contains internal circuitry that performs detection and classification of the speech signal, depending on numerous characteristics of the signal, to ensure the high perceptual quality in the reproduced signal. In certain embodiments of the invention, the signal is a speech signal, and the speech signal has a substantially music-like signal contained therein, and the extended signal coding system overrides any voice activity detection (VAD) decision that is used to determine which among a plurality of source coding modes are to be employed using a voice activity detection (VAD) correction/supervision circuitry. This is particularly relevant for discontinued transmission (DTX) operation. In certain embodiments of the invention, a signal coding circuitry maintains an improved perceptual quality in a coded signal having a substantially music-like component. This assurance of an improved perceptual quality is very desirable when there is a presence of a music-like signal in an un-coded signal.

摘要翻译： 一种扩展信号编码系统，其在信号中容纳实质上类似音乐的信号，同时在中断传输（DTX）操作期间保持重放信号中的高感知质量。 扩展信号编码系统包含根据信号的许多特征执行语音信号的检测和分类的内部电路，以确保再现信号中的高感知质量。 在本发明的某些实施例中，信号是语音信号，并且语音信号中包含基本类似音乐的信号，并且扩展信号编码系统覆盖任何语音活动检测（VAD）决定，其用于确定哪个 将使用语音活动检测（VAD）校正/监视电路来采用多种源编码模式。 这对于停止传输（DTX）操作特别有用。 在本发明的某些实施例中，信号编码电路在具有基本上类似音乐的分量的编码信号中保持改善的感知质量。 当在未编码信号中存在类似音乐的信号时，这种对感知质量改善的保证是非常理想的。

公开(公告)号：US20090177464A1

公开(公告)日：2009-07-09

申请号：US12381036

申请日：2009-03-06

申请人： Yang Gao ,  Adil Benyassine

发明人： Yang Gao ,  Adil Benyassine

IPC分类号： G10L11/04

CPC分类号： G10L19/00 ,  G10L19/083 ,  G10L19/167 ,  G10L19/18 ,  G10L19/24 ,  H03G3/00

摘要： A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization. The result of quantizing gains of periodic speech in this manner results in a reduction of the number of bits required to represent the quantized gain information and for periodic speech, the ability to use the quantized pitch gain for the current subframe to search the fixed codebook for the fixed codebook excitation vector for the current subframe. Alternatively, the new gain quantization process which was used only for periodic signals may be extended to non-periodic signals as well. This second strategy results in a slightly higher bit rate than that for periodic signals that use the new gain quantization strategy, but is still lower than the prior art's bit rate. Yet another alternative is to use the new gain quantization process for all speech signals without distinguishing between periodic and non-periodic signals.

摘要翻译： 语音编码器，其将每个语音帧分析和分类为周期性语音或非周期性类似语音，其中语音编码器根据语音是否周期性执行不同的增益量化处理。 如果语音是周期性的，则改进的语音编码器从非量化加权语音信号获得音调增益，并且在子帧处理开始之前针对帧的每个子帧执行自适应码本增益GP的前向量化，并且执行闭环延迟判定 固定码本增益GC的矢量量化。 如果语音是非周期性的，语音编码器可以使用任何已知的增益量化方法。 以这种方式量化周期性语音的增益的结果导致表示量化增益信息所需的位数减少，对于周期性语音，使用当前子帧的量化音调增益来搜索固定码本的能力 用于当前子帧的固定码本激励矢量。 或者，仅用于周期性信号的新增益量化处理也可以扩展到非周期信号。 该第二策略比使用新的增益量化策略的周期信号的比特率稍高，但是仍低于现有技术的比特率。 另一个替代方案是对所有语音信号使用新的增益量化处理，而不区分周期性和非周期性信号。

公开(公告)号：US07539300B1

公开(公告)日：2009-05-26

申请号：US11150696

申请日：2005-06-11

申请人： Adil Benyassine ,  Nick J. Lavrov

发明人： Adil Benyassine ,  Nick J. Lavrov

IPC分类号： H04M9/08

CPC分类号： H04B3/234

摘要： There is provided a method of detecting an infinite echo return loss (ERL) in an echo cancellation system while in a finite ERL mode. The method comprises determining a running mean attenuation by the echo cancellation system, determining a ratio of an echo signal to a near-end noise ratio (ENR), defining an infinite ERL threshold (THinfinite) as a function of the ENR, and switching to an infinite ERL mode as a function of the running mean attenuation and the THinfinite. The running mean attenuation can be enhanced echo return loss (ERLE), and the higher the ENR the higher the THinfinite and the lower the ENR the lower the THinfinite. The switching can further be a function of an energy distribution, where the switching switches to the infinite ERL mode based on a non-localized energy distribution over an echo path delay for a predetermined period of time.

摘要翻译： 提供了一种在有限ERL模式下在回声消除系统中检测无限回波回波损耗（ERL）的方法。 该方法包括通过回波消除系统确定运行的平均衰减，确定回波信号与近端噪声比（ENR）的比率，定义作为ENR的函数的无限ERL阈值（THinfinite），并切换到 作为运行平均衰减和无限次的函数的无限ERL模式。 运行平均衰减可以增强回波回波损耗（ERLE），ENR越高，THR越高，ENR越低，THnfinite越低。 切换还可以是能量分布的函数，其中切换基于在回波路径上的非局部能量分布延迟预定时间段切换到无限ERL模式。

公开(公告)号：US07260522B2

公开(公告)日：2007-08-21

申请号：US10888420

申请日：2004-07-10

申请人： Yang Gao ,  Adil Benyassine

发明人： Yang Gao ,  Adil Benyassine

IPC分类号： G10L19/00

CPC分类号： G10L19/00 ,  G10L19/083 ,  G10L19/167 ,  G10L19/18 ,  G10L19/24 ,  H03G3/00

摘要： There is provided a speech encoding system that receives a speech signal. The speech encoding system comprises a frame processor for processing a frame of the speech signal, where the frame processor includes a pitch gain generator that derives unquantized pitch gains, and a first vector guantizer that receives the unquantized pitch gains and generates quantized pitch gains. The speech encoding system also comprises a subframe processor that begins subframe processing after the pitch gain generator has derived the unquantized pitch gains and the first vector quantizer has generated the quantized pitch gains.

摘要翻译： 提供了一种接收语音信号的语音编码系统。 语音编码系统包括用于处理语音信号的帧的帧处理器，其中帧处理器包括导出非量化音调增益的音调增益发生器，以及接收未量化音调增益并产生量化音调增益的第一矢量指示器。 语音编码系统还包括在音调增益发生器已经导出无量子化音调增益并且第一矢量量化器已经产生量化音调增益之后开始子帧处理的子帧处理器。