公开(公告)号：US08874441B2

公开(公告)日：2014-10-28

申请号：US13174964

申请日：2011-07-01

申请人： Xianxian Zhang ,  Jes Thyssen ,  Kwan Young Shin

发明人： Xianxian Zhang ,  Jes Thyssen ,  Kwan Young Shin

IPC分类号： G10L21/02

CPC分类号： G10L21/0208 ,  G10L2021/02161

摘要： Techniques are described herein that suppress noise using multiple sensors (e.g., microphones) of a communication device. Noise modeling (e.g., estimation of noise basis vectors and noise weighting vectors) is performed with respect to a noise signal during operation of a communication device to provide a noise model. The noise model includes noise basis vectors and noise coefficients that represent noise provided by audio sources other than a user of the communication device. Speech modeling (e.g., estimation of speech basis vectors and speech weighting) is performed to provide a speech model. The speech model includes speech basis vectors and speech coefficients that represent speech of the user. A noisy speech signal is processed using the noise basis vectors, the noise coefficients, the speech basis vectors, and the speech coefficients to provide a clean speech signal.

摘要翻译： 本文描述了使用通信设备的多个传感器（例如，麦克风）抑制噪声的技术。 在通信设备的操作期间，相对于噪声信号执行噪声建模（例如噪声基矢量和噪声加权矢量的估计）以提供噪声模型。 噪声模型包括噪声基矢量和表示由通信设备的用户以外的音频源提供的噪声的噪声系数。 执行语音建模（例如，语音基本向量的估计和语音加权）以提供语音模型。 语音模型包括表示用户语音的语音基向量和语音系数。 使用噪声基矢量，噪声系数，语音基矢量和语音系数来处理噪声语音信号以提供干净的语音信号。

公开(公告)号：US08041562B2

公开(公告)日：2011-10-18

申请号：US12474927

申请日：2009-05-29

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L21/02

CPC分类号： G10L19/0204 ,  G10L19/005 ,  G10L19/04

摘要： A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

摘要翻译： 本文描述了一种技术，用于通过在代表预测编码系统中的编码音频信号的一系列帧中对接收到的帧进行解码而产生的音频输出信号中减少可听见的伪影。 根据该技术，确定接收到的帧是否是在一系列帧中的丢失帧之后的预定数量的接收帧中的一个。 响应于确定接收到的帧是预定数量的接收帧之一，与所接收的帧的解码相关联的至少一个参数或信号从与正常解码相关联的状态改变。 接收的帧然后根据至少一个参数或信号被解码以产生解码的音频信号。 然后基于解码的音频信号产生音频输出信号。

公开(公告)号：US20100217590A1

公开(公告)日：2010-08-26

申请号：US12391879

申请日：2009-02-24

申请人： Elias Nemer ,  Jes Thyssen

发明人： Elias Nemer ,  Jes Thyssen

IPC分类号： G10L15/20

CPC分类号： G01S3/86 ,  G01S3/8006 ,  G10L15/26 ,  G10L2021/02166

摘要： A system and method for performing speaker localization is described. The system and method utilizes speaker recognition to provide an estimate of the direction of arrival (DOA) of speech sound waves emanating from a desired speaker with respect to a microphone array included in the system. Candidate DOA estimates may be preselected or generated by one or more other DOA estimation techniques. The system and method is suited to support steerable beamforming as well as other applications that utilize or benefit from DOA estimation. The system and method provides robust performance even in systems and devices having small microphone arrays and thus may advantageously be implemented to steer a beamformer in a cellular telephone or other mobile telephony terminal featuring a speakerphone mode.

摘要翻译： 描述用于执行扬声器定位的系统和方法。 系统和方法利用说话者识别来提供相对于包括在系统中的麦克风阵列从期望的扬声器发出的语音声波的到达方向（DOA）的估计。 候选DOA估计可以由一个或多个其它DOA估计技术预先选择或产生。 该系统和方法适用于支持可导向波束形成以及利用或受益于DOA估计的其他应用。 该系统和方法即使在具有小麦克风阵列的系统和设备中也提供了强大的性能，因此可以有利地实现以引导具有扬声器电话模式的蜂窝电话或其他移动电话终端中的波束形成器。

公开(公告)号：US20090232228A1

公开(公告)日：2009-09-17

申请号：US12474927

申请日：2009-05-29

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： H04B14/04

CPC分类号： G10L19/0204 ,  G10L19/005 ,  G10L19/04

摘要： A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

摘要翻译： 本文描述了一种技术，用于通过在代表预测编码系统中的编码音频信号的一系列帧中对接收到的帧进行解码而产生的音频输出信号中减少可听见的伪影。 根据该技术，确定接收到的帧是否是在一系列帧中的丢失帧之后的预定数量的接收帧中的一个。 响应于确定接收到的帧是预定数量的接收帧之一，与所接收的帧的解码相关联的至少一个参数或信号从与正常解码相关联的状态改变。 接收的帧然后根据至少一个参数或信号被解码以产生解码的音频信号。 然后基于解码的音频信号产生音频输出信号。

公开(公告)号：US20070038443A1

公开(公告)日：2007-02-15

申请号：US11494633

申请日：2006-07-28

申请人： Jes Thyssen ,  Kelly Hale

发明人： Jes Thyssen ,  Kelly Hale

IPC分类号： G10L15/20

CPC分类号： H04M1/82 ,  H04M1/72513 ,  H04M1/72558 ,  H04M1/72563 ,  H04M1/80 ,  H04M3/4285

摘要： A communication device receives an encoded signal from a primary input source. The encoded signal includes periods of speech and periods of non-speech. The communication device includes a decoder to decode the received signal to produce a decoded signal. A detector of the communication device detects the periods of speech and the periods of non-speech within the decoded signal. A controller of the communication device provides the decoded signal to an output of the communication device during the periods of speech. The controller interrupts the decoded signal during the periods of non-speech and provides an alternate input from a secondary input source to the communication device.

摘要翻译： 通信设备从主输入源接收编码信号。 编码信号包括语音周期和非语音周期。 通信设备包括解码器，用于对接收到的信号进行解码以产生解码信号。 通信设备的检测器检测解码信号中的语音周期和非语音周期。 通信设备的控制器在语音周期期间将解码的信号提供给通信设备的输出。 控制器在非语音期间中断解码信号，并从辅助输入源向通信设备提供备用输入。

公开(公告)号：US20050192800A1

公开(公告)日：2005-09-01

申请号：US11065132

申请日：2005-02-24

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L19/04

CPC分类号： G10L19/04

摘要： A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

摘要翻译： 一种噪声反馈编码（NFC）系统和方法，其利用简单且相对便宜的一般结构配置，但是在控制编码噪声的形状方面实现了改进的灵活性。 NFC系统和方法利用全零噪声反馈滤波器，其被配置为近似极零噪声反馈滤波器的响应。

公开(公告)号：US06757649B1

公开(公告)日：2004-06-29

申请号：US10409404

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： 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：US06260010B1

公开(公告)日：2001-07-10

申请号：US09156650

申请日：1998-09-18

申请人： Yang Gao ,  Jes Thyssen ,  Adil Benyassine

发明人： Yang Gao ,  Jes Thyssen ,  Adil Benyassine

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 support lower bit rate encoding modes, a variety of techniques are applied many of which involve the classification of the input signal. The encoder utilizes gain normalization wherein LPC (linear predictive coding) gain provides a smoothing factor for combining both open and closed loop gains. The lower the LPC gain, the greater the open loop gain contribution to a gain normalization factor. The greater the LPC gain, the greater the closed loop gain contribution. For background noise, the smaller of the closed and open loop gains are used as the normalization factor. The normalization factor is limited by the LPC gain to prevent influencing the coding quality.

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。 编码器利用增益归一化，其中LPC（线性预测编码）增益提供用于组合开路和闭环增益的平滑因子。 LPC增益越低，开环增益对增益归一化因子的贡献越大。 LPC增益越大，闭环增益贡献越大。 对于背景噪声，使用较小的闭环和开环增益作为归一化因子。 归一化因子受LPC增益的限制，以防止影响编码质量。

公开(公告)号：US06205423B1

公开(公告)日：2001-03-20

申请号：US09420876

申请日：1999-10-19

申请人： Huan-Yu Su ,  Eric Kwok Fung Yuen ,  Adil Benyassine ,  Jes Thyssen

发明人： Huan-Yu Su ,  Eric Kwok Fung Yuen ,  Adil Benyassine ,  Jes Thyssen

IPC分类号： G10L1904

CPC分类号： G10L19/012 ,  G10L19/12 ,  G10L2019/0002

摘要： A method of coding speech under background noise conditions or during noise-like speech periods wherein during active voice speech segments an analysis-by-synthesis method is used. However, when a background noise segment or noise-like speech segment is detected, an adaptive code book (pitch prediction) contribution is used as a source of a pseudo-random sequence in order to provide a better representation of the background noise or the noise-like speech. An improved gain quantization scheme is also employed when a background noise segment is detected, wherein energy of the total excitation with quantized gains is matched to the energy of total excitation with unquantized gains.

摘要翻译： 一种在背景噪声条件下或在噪声状语音周期期间对语音进行编码的方法，其中在活动语音语音段中使用按合成分析方法。 然而，当检测到背景噪声段或类噪声语音段时，将自适应码本（音调预测）​​贡献用作伪随机序列的源，以便提供背景噪声或噪声的更好表示 像演讲 当检测到背景噪声段时，还采用改进的增益量化方案，其中具有量化增益的总激励的能量与具有非量化增益的总激励能量相匹配。

公开(公告)号：US06188980B1

公开(公告)日：2001-02-13

申请号：US09154653

申请日：1998-09-18

申请人： Jes Thyssen

发明人： Jes Thyssen

IPC分类号： G10L1100

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. The encoder produces a series of LSF (line spectral frequencies) vectors. For filter stability, each LSF vector comprises an ascending sequence of LSF values. Occasionally, pairs of LSF values are produced (or become through an introduction of channel error) out of ascending order. In response, the encoder performs frame erasure, LSF concealment or pair flipping. With a relatively large number of out of order pairs, frame erasure is applied. With a single out of order pair, within the LSF vector, the pair are flipped. Likewise, with two pairs out of order, the previous LSF vector is used to generate the current LSF vector using concealment. The performance of frame erasure, LSF concealment or pair flipping is also performed in the decoder as well as in the encoder in some embodiments.

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。 编码器产生一系列LSF（线谱频率）矢量。 为了滤波器的稳定性，每个LSF向量包括LSF值的上升序列。 偶尔会出现升序的LSF值对（或通过引入通道错误）。 作为响应，编码器执行帧擦除，LSF隐藏或对翻转。 使用相对大量的乱序对，应用帧擦除。 对于LSF向量中的单个失序对，该对被翻转。 同样地，使用两对乱序，先前的LSF向量用于使用隐藏生成当前的LSF向量。 在一些实施例中，在解码器以及编码器中也执行帧擦除，LSF隐藏或对翻转的执行。