公开(公告)号：US06873956B2

公开(公告)日：2005-03-29

申请号：US10464307

申请日：2003-06-17

申请人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

发明人： Carlo Murgia ,  Jeffrey D. Klein ,  Huan-Yu Su

IPC分类号： G01L21/04 ,  G10L19/00 ,  G10L19/14 ,  G10L21/04 ,  H04L12/56

CPC分类号： G10L19/008

摘要： An exemplary multi-channel speech processor comprises a controller capable of interfacing with a plurality of channels, and at least one signal processing unit (SPU) coupled to the controller, where the multi-channel speech processor has a maximum execution time for processing all frames, one channel at a time, by processing a single frame from each of the plurality of channels. The signal processing unit encodes each of the single frames from each of the plurality of channels, one channel at a time, to generate encoded frames until the maximum execution time elapses or is about to elapse. The controller also transmits a pre-determined frame for each of the plurality of channels not processed during the encoding step, due to the maximum execution time elapsing or being about to elapse, such that the predetermined frame causes a decoder which receives the predetermined frame to generate a frame erase frame.

摘要翻译： 示例性多声道语音处理器包括能够与多个通道接口的控制器以及耦合到控制器的至少一个信号处理单元（SPU），其中多声道语音处理器具有用于处理所有帧的最大执行时间 通过处理来自多个信道中的每一个的单个帧，一次一个信道。 信号处理单元从多个信道中的每一个，每次一个信道中对每个单个帧进行编码，以生成经编码的帧，直到经过或将要经过最大执行时间。 由于最大执行时间经过或将要经过，控制器还为编码步骤期间未被处理的多个通道中的每一个发送预定帧，使得预定帧使接收预定帧的解码器 生成帧擦除帧。

公开(公告)号：US06850884B2

公开(公告)日：2005-02-01

申请号：US09783822

申请日：2001-02-14

申请人： Yang Gao ,  Huan Yu-Su

发明人： Yang Gao ,  Huan Yu-Su

IPC分类号： G10L19/14 ,  G10L21/02

CPC分类号： G10L19/18 ,  G10L19/265 ,  G10L21/0364

摘要： In a coding procedure, coding parameters are selected for coding the speech signal to achieve enhanced perceptual quality of reproduced speech. At least one coding parameter value or preferential coding parameter value is selected to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

摘要翻译： 在编码过程中，选择编码参数来编码语音信号，以实现增强的再现语音的感知质量。 选择至少一个编码参数值或优先编码参数值以使语音信号的频谱响应更均匀，以补偿由信号处理系统相关联的通信网络否则可能赋予语音信号的频谱变化。

公开(公告)号：US06842733B1

公开(公告)日：2005-01-11

申请号：US09781735

申请日：2001-02-12

申请人： Yang Gao ,  Huan-Yu Su

发明人： Yang Gao ,  Huan-Yu Su

IPC分类号： G10L19/14 ,  G10L21/02

CPC分类号： G10L19/18 ,  G10L19/265 ,  G10L21/0364

摘要： A signal processing system is well suited for conditioning a speech signal prior to coding the speech signal to achieve enhanced perceptual quality of reproduced speech. The signal processing system may be incorporated into mobile or portable wireless communications devices, wireless infrastructure equipment, or both. The signal processing system includes a filtering arrangement for filtering an input speech signal to make a spectral response of the speech signal more uniform to compensate for spectral variations that might otherwise be imparted into the speech signal by a communications network associated with the signal processing system.

摘要翻译： 信号处理系统非常适合在对语音信号进行编码之前对语音信号进行调理，以实现增强的再现语音的感知质量。 信号处理系统可以并入到移动或便携式无线通信设备，无线基础设施设备或两者中。 信号处理系统包括用于对输入语音信号进行滤波以使语音信号的频谱响应更均匀的滤波装置，以补偿由与信号处理系统相关联的通信网络可能赋予语音信号的频谱变化。

公开(公告)号：US06823303B1

公开(公告)日：2004-11-23

申请号：US09156832

申请日：1998-09-18

申请人： Huan-Yu Su ,  Adil Benyassine ,  Jes Thyssen

发明人： Huan-Yu Su ,  Adil Benyassine ,  Jes Thyssen

IPC分类号： G10L1904

CPC分类号： G10L19/005 ,  G10L19/002 ,  G10L19/012 ,  G10L19/08 ,  G10L19/083 ,  G10L19/09 ,  G10L19/10 ,  G10L19/12 ,  G10L19/125 ,  G10L19/18 ,  G10L19/265 ,  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.

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。

公开(公告)号：US06804203B1

公开(公告)日：2004-10-12

申请号：US09663246

申请日：2000-09-15

申请人： Adil Benyassine ,  Patrick D. Ryan ,  Huan-Yu Su

发明人： Adil Benyassine ,  Patrick D. Ryan ,  Huan-Yu Su

IPC分类号： G01R3108

CPC分类号： H04M9/082

摘要： A speech communication system is provided that uses pitch information, pitch lags, pitch gains, energy and/or other speech characteristics about the outgoing speech and the unknown signal on a frame basis to determine if the unknown signal is an echo signal of the outgoing speech or if the unknown signal also contains speech from a second talker (double talk). Additionally, a plurality of frames of these characteristics of the outgoing speech signal and the unknown incoming signal may be buffered so that the analysis and comparison can be made more efficiently and quickly in the frame domain as opposed to a time domain. Multiple characteristics may be optionally weighted and then analyzed. The system and method may further determine a level of confidence, based on any criterion, in the determination that may then be used to adjust the gain of a filter.

摘要翻译： 提供一种语音通信系统，其基于帧基于使用音调信息，音调延迟，音调增益，能量和/或关于输出语音和未知信号的其他语音特性，以确定未知信号是否是出局语音的回波信号 或者如果未知信号还包含来自第二讲话者的语音（双语）。 此外，输出语音信号和未知输入信号的这些特性的多个帧可以被缓冲，使得可以在帧域中比时域更加有效和快速地进行分析和比较。 可以可选地对多个特征加权并分析。 系统和方法可以基于任何标准来确定可能随后用于调整滤波器的增益的确定中的置信水平。

公开(公告)号：US06604070B1

公开(公告)日：2003-08-05

申请号：US09663734

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.

公开(公告)号：US06493665B1

公开(公告)日：2002-12-10

申请号：US09154662

申请日：1998-09-18

申请人： Huan-Yu Su ,  Yang Gao

发明人： Huan-Yu Su ,  Yang Gao

IPC分类号： G10L2102

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

摘要： A multi-rate speech coded 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. To support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. For each bit rate mode selected, pluralities of fixed or innovation subcodebooks are selected for use in generating innovation vectors. The fixed codebook contains pulse subcodebooks and noise-like subcodebooks. To assist in selection of one of the subcodebooks, an adaptive weighting approach is applied in a searching procedure wherein residual classification and various parameters are used to generate a weighting function that is used to favor one subcodebook over another. The pulse subcodebooks are favored to code pulse-like residuals, while the noise-like subcodebooks are favored to code noise-like residuals. The classification may involve identification of noise-like residuals, while the various parameters may comprise pitch correlation, signal to noise ratio, and average to peak ratio. Favoring involves an adjustment to a weighting factor applied to the subcodebooks.

摘要翻译： 通过自适应地选择编码比特率模式以匹配通信信道限制，多速率语音编码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。 为了支持较低比特率编码模式，应用了许多技术，其中许多技术涉及输入信号的分类。 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。 固定码本包含脉冲子码本和类噪声子码。 为了帮助选择一个子码本，在搜索过程中应用自适应加权方法，其中使用残差分类和各种参数来产生用于有利于一个子码本的加权函数。 脉冲子码本有利于编码类似脉冲的残差，而类似噪声的子代码则有利于编码类似噪声的残差。 该分类可以涉及噪声类残差的识别，而各种参数可以包括音调相关性，信噪比和平均峰值比。 喜好涉及到适用于子码本的加权因子的调整。

公开(公告)号：US06256606B1

公开(公告)日：2001-07-03

申请号：US09200624

申请日：1998-11-30

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： G10L2100

CPC分类号： G10L19/012

摘要： Silence description coding for multi-rate speech coding systems that employ discontinued transmission. Speech coding systems include multi-rate speech codecs having an encoder and a decoder. The silence description coding 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： 使用中断传输的多速率语音编码系统的静音描述编码。 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：US6122611A

公开(公告)日：2000-09-19

申请号：US75365

申请日：1998-05-11

申请人： Huan-yu Su ,  Adil Benyassine

发明人： Huan-yu Su ,  Adil Benyassine

IPC分类号： G10L19/04 ,  G10L19/00 ,  G10L19/12 ,  G10L21/02 ,  H03M7/30 ,  G10L19/14

CPC分类号： G10L19/012 ,  G10L21/0364

摘要： A system and method to improve the quality of coded speech coexisting with background noise. For instance, the present invention receives a coded speech signal via a communication network and then decodes and synthesizes the different parameters contained within it to produce a synthesized speech signal. The present invention determines the non-speech periods that are represented within the synthesized speech signal. The determined non-speech periods are then utilized to determine and code LPC parameters needed for background noise synthesis. Because medium or low bit rate LPC-coded speech during voice activity periods has the coexisting background noise attenuated, the decoded signal has audible abrupt changes in the level of the background noise. To improve decoded speech quality, the present invention adds simulated background noise to decoded noisy speech when synthesizing the noisy speech signal during voice activity periods. The resulting output signal sounds more natural and realistic to the human ear because of the continuous presence of background noise during speech and non-speech periods.

摘要翻译： 一种改善与背景噪声共存的编码语音质量的系统和方法。 例如，本发明经由通信网络接收编码的语音信号，然后解码并合成包含在其中的不同参数，以产生合成语音信号。 本发明确定在合成语音信号内表示的非语音周期。 然后使用所确定的非语音周期来确定和编码背景噪声合成所需的LPC参数。 由于语音活动期间的中等或低比特率LPC编码的语音具有衰减的共存背景噪声，所以解码信号在背景噪声的电平中具有可听到的突然变化。 为了提高解码的语音质量，本发明在语音活动期间合成噪声语音信号时，将解码噪声语音的模拟背景噪声增加。 由于在语音和非语音周期期间持续存在背景噪声，所得到的输出信号对于人耳来说更加自然和现实。

公开(公告)号：US6104992A

公开(公告)日：2000-08-15

申请号：US154663

申请日：1998-09-18

申请人： Yang Gao ,  Huan-Yu Su

发明人： Yang Gao ,  Huan-Yu Su

IPC分类号： G10L11/04 ,  G10L19/00 ,  G10L19/08 ,  G10L19/10 ,  G10L19/12 ,  G10L19/14 ,  G10L21/02 ,  G01L19/12 ,  G01L19/14

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

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. The encoder applies adaptive gain reduction to optimize selection of appropriate gain contributions from the adaptive and fixed codebooks. Specifically, the encoder uses a first target signal to identify a contribution (a best code vector and a gain) from the adaptive codebook. Thereafter, a contribution from the fixed codebook is selected. The gain associated with the adaptive codebook contribution is then reduced by a factor, and the gain contribution from the fixed codebook is searched a second time, permitting fine tuning of the overall contribution. The gain reduction factor applied is adapted by considering both the encoding bit rate and a normalized correlation between the original target signal and the filtered signal from the adaptive codebook.

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。 编码器采用自适应增益减小来优化来自自适应和固定码本的适当增益贡献的选择。 具体地，编码器使用第一目标信号来识别来自自适应码本的贡献（最佳码矢量和增益）。 此后，选择固定码本的贡献。 然后，与自适应码本贡献相关联的增益被减少一个因素，并且第二次搜索来自固定码本的增益贡献，允许对整个贡献进行微调。 通过考虑编码比特率和原始目标信号与来自自适应码本的滤波信号之间的归一化相关性来适应所应用的增益减小因子。