公开(公告)号：US20060198511A1

公开(公告)日：2006-09-07

申请号：US11072476

申请日：2005-03-03

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

发明人： Huan-Yu Su ,  Adil Benyassine ,  Nick 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：US06898566B1

公开(公告)日：2005-05-24

申请号：US09640841

申请日：2000-08-16

申请人： Adil Benyassine ,  Huan-Yu Su

发明人： Adil Benyassine ,  Huan-Yu Su

IPC分类号： G10L11/02 ,  G10L11/04 ,  G10L19/14 ,  G10L21/02

CPC分类号： G10L19/22 ,  G10L19/09 ,  G10L2025/783

摘要： There are provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one of a plurality of speech coding algorithms, the plurality of speech parameters includes pitch information, the plurality of speech parameters is calculated using a plurality of thresholds. An example method includes estimating a background noise level in the speech signal to determine a signal to noise ratio (SNR) for the speech signal, adjusting one or more of the plurality of thresholds based on the SNR to generate one or more SNR adjusted thresholds, analyzing the speech signal to extract the pitch information using the one or more SNR adjusted thresholds, and repeating the estimating, the adjusting and the analyzing to code the speech signal using one the plurality of speech coding algorithms.

摘要翻译： 提供了语音编码方法和系统，用于使用多种语音编码算法中的一种来估计用于对语音信号进行编码的语音信号的多个语音参数，所述多个语音参数包括音调信息，所述多个语音参数被计算 使用多个阈值。 示例性方法包括估计语音信号中的背景噪声电平以确定语音信号的信噪比（SNR），基于SNR调整多个阈值中的一个或多个阈值以产生一个或多个SNR调整阈值， 分析语音信号以使用一个或多个SNR调整的阈值提取音调信息，并且使用多个语音编码算法中的一个重复对该语音信号的估计，调整和分析。

公开(公告)号：US06735567B2

公开(公告)日：2004-05-11

申请号：US10409430

申请日：2003-04-08

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1304

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

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。

公开(公告)号：US06581032B1

公开(公告)日：2003-06-17

申请号：US09662828

申请日：2000-09-15

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L1912

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

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。

公开(公告)号：US20090043574A1

公开(公告)日：2009-02-12

申请号：US12284623

申请日：2008-09-23

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L19/12 ,  G10L19/00

CPC分类号： G10L19/00 ,  G10L19/167 ,  G10L19/20 ,  G10L19/22 ,  G10L19/24 ,  G10L2019/0001 ,  H03G3/00

摘要： There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse.

摘要翻译： 提供了一种对从语音信号产生的语音数据进行解码的方法。 该方法包括：接收语音数据的子帧中具有至少一个主脉冲的语音数据; 基于所述至少一个主脉冲在所述语音数据的子帧中的所述主脉冲的一侧产生第一预测脉冲，其中所述第一预测脉冲具有比所述主脉冲更低的增益; 在语音数据的子帧中的主脉冲的另一侧上产生第二预测脉冲作为反时限上的第一预测脉冲的镜像; 使用所述至少一个主脉冲，所述第一预测脉冲和所述第二预测脉冲来重构所述语音信号。

公开(公告)号：US07146309B1

公开(公告)日：2006-12-05

申请号：US10653874

申请日：2003-09-02

申请人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

发明人： Adil Benyassine ,  Eyal Shlomot ,  Huan-Yu Su

IPC分类号： G10L19/00

CPC分类号： G10L19/08

摘要： There are provided methods and devices for generating excitation values for a speech signal. In one aspect, an example method comprises obtaining one or more characteristics of a first speech frame of the speech signal, deriving a first seed value based on the one or more characteristics of the first speech frame, providing the first seed value to a Gaussian time series generator; and using the Gaussian time series generator to generate an excitation values for the first frame. The one or more characteristics may include a spectrum information of the first frame, an energy information of the first frame, or a gain information of the first frame.

摘要翻译： 提供了用于产生语音信号的激励值的方法和装置。 在一个方面，示例性方法包括获得语音信号的第一语音帧的一个或多个特征，基于第一语音帧的一个或多个特征导出第一种子值，将第一种子值提供给高斯时间 串联发电机; 并使用高斯时间序列发生器来产生第一帧的激励值。 一个或多个特征可以包括第一帧的频谱信息，第一帧的能量信息或第一帧的增益信息。

公开(公告)号：US07120578B2

公开(公告)日：2006-10-10

申请号：US09841764

申请日：2001-04-24

申请人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

发明人： Jes Thyssen ,  Huan-yu Su ,  Adil Benyassine ,  Eyal Shlomot

IPC分类号： G10L11/06 ,  G10L19/12

CPC分类号： G10L19/012

摘要： Speech coding systems include multi-rate speech codecs having an encoder and a decoder. Silence description coding for multi-rate speech coding systems that employ discontinued transmission is performed in either the encoder or the decoder of the multi-rate speech codec. It may also be performed in a distributed manner wherein it is performed partially in the encoder and partially in the decoder. The silence description coding is performed on a speech signal having a substantially non-speech-like characteristic. Voice activity detection classifies the speech signal as being either substantially speech-like or substantially non-speech-like. The silence description coding is selected from a plurality of coding modes. In certain embodiments of the invention, the silence description coding is a source coding mode that operates at a bit rate that fits within a bit rate budget as determined by all of the available source coding modes within the plurality of coding modes. The silence description coding is also accompanied with signaling coding and channel coding of the speech signal. Error checking is performed using an unused portion of a bandwidth of the multi-rate speech codec's bit rate. This error checking involves majority voting in certain embodiments of the invention.

摘要翻译： 语音编码系统包括具有编码器和解码器的多速率语音编解码器。 在多速率语音编解码器的编码器或解码器中执行采用中断传输的多速率语音编码系统的静音描述编码。 它也可以以分布式方式执行，其中部分地在编码器中执行，部分地在解码器中执行。 对具有基本上非语音的特征的语音信号执行静音描述编码。 语音活动检测将语音信号分类为基本上是语音的或基本上非语音的。 从多种编码模式中选择静音描述编码。 在本发明的某些实施例中，静默描述编码是以适合在多个编码模式内的所有可用源编码模式所确定的比特率预算中的比特率操作的源编码模式。 静音描述编码也伴随着语音信号的信令编码和信道编码。 使用多速率语音编解码器的比特率的带宽的未使用部分来执行错误检查。 在本发明的某些实施例中，该错误检查涉及多数投票。

公开(公告)号：US06961698B1

公开(公告)日：2005-11-01

申请号：US10420654

申请日：2003-04-21

申请人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

发明人： Yang Gao ,  Adil Benyassine ,  Jes Thyssen ,  Eyal Shlomot ,  Huan-yu Su

IPC分类号： G10L19/00 ,  G10L13/00 ,  G10L13/04 ,  G10L19/02 ,  G10L19/04 ,  G10L19/08 ,  G10L19/10 ,  G10L19/12 ,  G10L19/14 ,  H03M7/30 ,  H03M7/36

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

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 bitstream comprises a type component and a gain component. The type component is representative of a type classification of a frame of speech signal that is transmitted. The type component comprises a first type and second type. The gain component represents an adaptive codebook gain and a fixed codebook gain component comprises a fixed codebook gain component and an adaptive codebook gain component exclusively encoded as separate components of the bitstream as a function of the bit rate when the type classification is the second type.

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。 比特流包括类型分量和增益分量。 类型分量代表传输的语音信号帧的类型分类。 类型组件包括第一类型和第二类型。 增益分量表示自适应码本增益，并且固定码本增益分量包括固定码本增益分量和自适应码本增益分量，该类型分类作为第二类型时，作为比特率的单独分量专门编码。

公开(公告)号：US06463414B1

公开(公告)日：2002-10-08

申请号：US09547832

申请日：2000-04-12

申请人： Huan-Yu Su ,  Eyal Shlomot ,  Jes Thyssen ,  Adil Benyassine ,  Yang Gao

发明人： Huan-Yu Su ,  Eyal Shlomot ,  Jes Thyssen ,  Adil Benyassine ,  Yang Gao

IPC分类号： G10L1102

CPC分类号： G10L19/173

摘要： There is provided a conference bridge or transcoder configured to intelligently handle multiple speech channels in the contest of a packet network, wherein various speech channels may adhere to variety of speech encoding standards. For example, the conference bridge establishes framing and alignment of multiple incoming speech channels associated with multiple participants, extracts parameters from the speech samples, mixes the parameters, and re-encodes the resulting speech samples for transmission to the participants. In one aspect, a speech processing method comprises decoding a first bitstream according to a first coding scheme to generate first speech samples and a first side information; generating second speech samples and a second side information using the first speech samples and the first side information, for use according to a second coding scheme; and creating a second bitstream, encoded based on the second coding scheme, using the second speech samples and the second side information.

摘要翻译： 提供了一种配置成在分组网络的比赛中智能地处理多个语音信道的会议桥或代码转换器，其中各种语音信道可以遵循各种语音编码标准。 例如，会议桥建立与多个参与者相关联的多个输入语音信道的成帧和对准，从语音样本中提取参数，混合参数，并对所得到的语音样本进行重新编码以传输给参与者。 一方面，语音处理方法包括根据第一编码方案对第一比特流进行解码，以产生第一语音样本和第一侧信息; 使用第一语音样本和第一侧信息生成第二语音样本和第二侧信息，以便根据第二编码方案使用; 以及使用所述第二语音样本和所述第二侧信息来创建基于所述第二编码方案编码的第二比特流。

公开(公告)号：US06240386B1

公开(公告)日：2001-05-29

申请号：US09198414

申请日：1998-11-24

申请人： Jes Thyssen ,  Huan-yu Su ,  Yang Gao ,  Adil Benyassine

发明人： Jes Thyssen ,  Huan-yu Su ,  Yang Gao ,  Adil Benyassine

IPC分类号： G10L2100

CPC分类号： G10L19/265 ,  G10L19/002 ,  G10L19/005 ,  G10L19/012 ,  G10L19/08 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/125 ,  G10L19/18 ,  G10L19/20 ,  G10L21/0364 ,  G10L2019/0005 ,  G10L2019/0007 ,  G10L2019/0011

摘要： A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The speech coder distinguishes various voice signals as a function of their voice content. For example, a Voice Activity Detection (VAD) algorithm selects an appropriate coding scheme depending on whether the speech signal comprises active or inactive speech. The encoder may consider varying characteristics of the speech signal including sharpness, a delay correlation, a zero-crossing rate, and a residual energy. In another embodiment of the present invention, code excited linear prediction is used for voice active signals whereas random excitation is used for voice inactive signals; the energy level and spectral content of the voice inactive signal may also be used for noise coding. The multi-rate speech codec may employ distributed detection and compensation processing the speech signal. For high quality perceptual speech reproduction, the speech codec may perform noise detection in both an encoder and a decoder. The noise detection may be coordinated between the encoder and decoder. Similarly, noise compensation may be performed in a distributed manner among both the decoder and the encoder.

摘要翻译： 多速率语音编解码器通过自适应地选择编码比特率模式以匹配通信信道限制来支持多种编码比特率模式。 在较高的比特率编码模式中，通过CELP（码激励线性预测）和其他相关联的建模参数的语音的精确表示被生成用于更高质量的解码和再现。 对于所选择的每个比特率模式，选择多个固定或创新子码本来用于产生创新向量。 语音编码器将各种语音信号区分为其语音内容的函数。 例如，语音活动检测（VAD）算法根据语音信号是否包括有源或非活动语音来选择适当的编码方案。 编码器可以考虑包括锐度，延迟相关性，零交叉速率和剩余能量的语音信号的变化特性。 在本发明的另一实施例中，码激励线性预测用于语音有源信号，而随机激励用于语音无效信号; 语音无效信号的能级和频谱内容也可用于噪声编码。 多速率语音编解码器可以采用语音信号的分布式检测和补偿处理。 对于高质量的感知语音再现，语音编解码器可以在编码器和解码器中执行噪声检测。 可以在编码器和解码器之间协调噪声检测。 类似地，可以在解码器和编码器之间以分布式方式执行噪声补偿。