公开(公告)号：US07593852B2

公开(公告)日：2009-09-22

申请号：US11700481

申请日：2007-01-30

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

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

IPC分类号： G10L15/20

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

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。

公开(公告)号：US20070136052A1

公开(公告)日：2007-06-14

申请号：US11700481

申请日：2007-01-30

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： G10L11/04

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

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.

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：US06813602B2

公开(公告)日：2004-11-02

申请号：US10105120

申请日：2002-03-22

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L1912

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. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

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。 为了在低比特率编码模式下实现高质量，语音编码器脱离了常规CELP编码器的严格波形匹配标准，并努力识别输入信号的重要感知特征。 编码器通过移位或其他重新排列从单个归一化码矢量生成多个码矢量。 结果，提高了搜索速度，并且大大地减少了由这些代码矢量构建的码本的物理大小。

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：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：US06480822B2

公开(公告)日：2002-11-12

申请号：US09156648

申请日：1998-09-18

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L1900

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. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

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。 为了在低比特率编码模式下实现高质量，语音编码器脱离了常规CELP编码器的严格波形匹配标准，并努力识别输入信号的重要感知特征。 编码器通过移位或其他重新排列从单个归一化码矢量生成多个码矢量。 结果，提高了搜索速度，并且大大地减少了由这些代码矢量构建的码本的物理大小。

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：US09253568B2

公开(公告)日：2016-02-02

申请号：US12780179

申请日：2010-05-14

申请人： Elias Nemer ,  Wilfrid LeBlanc ,  Syavosh Zad-Issa ,  Jes Thyssen

发明人： Elias Nemer ,  Wilfrid LeBlanc ,  Syavosh Zad-Issa ,  Jes Thyssen

IPC分类号： G10L19/00 ,  H04R3/00

CPC分类号： H04R3/007

摘要： A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame.

摘要翻译： 描述了用于抑制音频信号中的诸如风噪声之类的非平稳噪声的技术。 根据该技术，分析音频信号的一系列帧以检测音频信号是否包括非平稳噪声。 如果检测到音频信号包括非平稳噪声，则执行多个步骤。 根据这些步骤，确定音频信号的帧是否包括非平稳噪声或语音和非平稳噪声。 如果确定帧包括非平稳噪声，则将第一滤波器应用于帧，并且如果确定帧包括语音和非平稳噪声，则将第二滤波器应用于帧。

公开(公告)号：US20110096942A1

公开(公告)日：2011-04-28

申请号：US12897548

申请日：2010-10-04

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： H04B15/00

CPC分类号： G10L21/0208 ,  G10L2021/02163

摘要： Systems and methods are described for applying noise suppression to one or more audio signals to generate a noise-suppressed audio signal therefrom. In a single-channel implementation, an input signal is received that comprises a desired audio signal and an additive noise signal. Noise suppression is then applied to the input signal to generate a noise-suppressed signal in a manner that is controlled by at least a parameter that specifies a degree of balance between distortion of the desired audio signal and unnaturalness of a residual noise signal included in the noise-suppressed signal. In an alternative single-channel implementation, a plurality of sub-band signals obtained by applying a frequency conversion process to a time domain representation of an input signal is received. Noise suppression is then applied to each of the sub-band signals by passing each of the sub-band signals through a time direction filter. Multi-channel noise suppression variants are also described.

摘要翻译： 描述了用于将噪声抑制应用于一个或多个音频信号以从其产生噪声抑制音频信号的系统和方法。 在单通道实现中，接收包括所需音频信号和加性噪声信号的输入信号。 噪声抑制然后被施加到输入信号以产生噪声抑制信号，该信号由至少一个参数指定的方式控制，该参数指定期望音频信号的失真与包括在所述音频信号中的残留噪声信号的不自然度之间的平衡程度 噪声抑制信号。 在替代的单通道实现中，接收通过将频率转换处理应用到输入信号的时域表示而获得的多个子带信号。 然后通过使每个子带信号通过时间方向滤波器将噪声抑制应用于每个子带信号。 还介绍了多通道噪声抑制方案。