公开(公告)号：US08032363B2

公开(公告)日：2011-10-04

申请号：US10215048

申请日：2002-08-09

Applicant: Juin-Hwey Chen ,  Jes Thyssen ,  Chris C Lee

Inventor： Juin-Hwey Chen ,  Jes Thyssen ,  Chris C Lee

IPC: G10L19/14 ,  G10L21/02

CPC classification number: G10L19/26

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

Abstract translation: 一种处理包括连续DS帧的解码语音（DS）信号的方法，每个DS帧包括DS采样。 该方法包括：对DS信号进行自适应滤波以产生滤波信号; 利用DS帧更新一次自适应增益来对滤波后的信号进行增益放大，从而产生增益标度信号; 并且执行平滑操作以平滑增益调节信号中的可能的波形不连续性。

公开(公告)号：US07353168B2

公开(公告)日：2008-04-01

申请号：US10183418

申请日：2002-06-28

Applicant: Jes Thyssen ,  Chris C Lee ,  Juin-Hwey Chen

Inventor： Jes Thyssen ,  Chris C Lee ,  Juin-Hwey Chen

IPC: G10L19/10

CPC classification number: G10L19/26

Abstract: A method to eliminate discontinuities in an adaptively filtered signal includes filtering a beginning portion of a current signal frame using a past set of filter coefficients, thereby producing a first filtered frame portion. The method also includes filtering the beginning portion of the current signal frame using a current set of filter coefficients, thereby producing a second filtered frame portion. The method also includes modifying the second filtered frame portion with the first filtered frame portion so as to smooth a possible filtered signal discontinuity between the second filtered frame portion and a past filtered frame produced using the past filter coefficients.

Abstract translation: 消除自适应滤波信号中的不连续性的方法包括使用过去的滤波器系数集合来过滤当前信号帧的开始部分，由此产生第一滤波帧部分。 该方法还包括使用当前的一组滤波器系数对当前信号帧的开始部分进行滤波，从而产生第二滤波帧部分。 该方法还包括用第一经滤波的帧部分修改第二经滤波的帧部分，以平滑第二滤波帧部分与使用过去的滤波器系数产生的过去滤波的帧之间可能的滤波信号不连续。

公开(公告)号：US09336785B2

公开(公告)日：2016-05-10

申请号：US12464498

申请日：2009-05-12

Applicant: Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

Inventor： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

IPC: G10L19/14 ,  G10L19/012 ,  G10L21/0208 ,  G10L21/0232

CPC classification number: G10L21/0208 ,  G10L19/012 ,  G10L21/0232

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract translation: 描述了一种语音清晰度增强（SIE）系统和方法，当音频设备位于具有大声背景噪声的环境中时，提高了由音频设备重放的语音信号的可懂度。 在一个实施例中，音频设备包括近端电话终端，并且语音信号包括通过通信网络从远端电话终端接收的用于在近端电话终端回放的语音信号。

公开(公告)号：US09330675B2

公开(公告)日：2016-05-03

申请号：US13250355

申请日：2011-09-30

Applicant: Xianxian Zhang ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

Inventor： Xianxian Zhang ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

IPC: A61F11/06 ,  G10L21/0208 ,  G10L21/0272 ,  H04R1/24 ,  G10L21/0216

CPC classification number: G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract translation: 不同于声音的压力波，无处不在，由风引起的空气湍流通常是一个相当地方的事件。 因此，在使用两个或多个空间分离的麦克风拾取声音信号（例如语音）的系统中，由麦克风之一拾取的风噪声通常不会被拾取（或至少不相同的程度），由 另一个麦克风。 描述利用这种技巧和其他技术来有效地检测和抑制使用空间分离的多个麦克风的风噪声的方法和装置的实施例。

公开(公告)号：US09130643B2

公开(公告)日：2015-09-08

申请号：US13362244

申请日：2012-01-31

Applicant: Juin-Hwey Chen ,  Thomas F. Baker ,  Evan S. McCarthy ,  Jes Thyssen ,  Walter J. Wihardja ,  David C. Garrett

Inventor： Juin-Hwey Chen ,  Thomas F. Baker ,  Evan S. McCarthy ,  Jes Thyssen ,  Walter J. Wihardja ,  David C. Garrett

IPC: H04B1/10 ,  H03D1/04 ,  H04B1/16 ,  H04B1/7183 ,  H03G3/34 ,  H04B1/04

CPC classification number: H04B1/1661 ,  H03G3/345 ,  H04B1/04 ,  H04B1/7183

Abstract: Systems and methods are described for enhancing the audio quality of an FM receiver. In embodiments described herein, quadrature L−R demodulation is applied to a composite baseband signal output by an FM demodulator to obtain an L−R noise signal. A channel quality measure is calculated based on the L−R noise signal and is used to control whether a pop suppression technique is applied to an L+R signal obtained from the composite baseband signal to detect and remove noise pulses therefrom. The channel quality measure and the L−R noise signal are also leveraged to perform single-channel noise suppression in the frequency domain on an L−R signal obtained from the composite baseband signal and on the L+R signal. The channel quality measure is also used to control the application of a fast fading compensation process that replaces noisy segments of the L−R and L+R signal with replacement waveforms generated via waveform extrapolation.

Abstract translation: 描述了用于增强FM接收机的音频质量的系统和方法。 在本文描述的实施例中，正交L-R解调被应用于由FM解调器输出的复合基带信号以获得L-R噪声信号。 基于L-R噪声信号计算信道质量度量，并且用于控制是否将弹出抑制技术应用于从复合基带信号获得的L + R信号，以从其检测和去除噪声脉冲。 信道质量测量和L-R噪声信号也被用于在从复合基带信号和L + R信号获得的L-R信号上对频域执行单信道噪声抑制。 信道质量测量还用于控制快速衰落补偿过程的应用，其中替换了通过波形外推产生的替换波形的L-R和L + R信号的噪声段。

公开(公告)号：US08670990B2

公开(公告)日：2014-03-11

申请号：US12847120

申请日：2010-07-30

Applicant: Juin-Hwey Chen ,  Hong-Goo Kang ,  Robert W. Zopf ,  Jes Thyssen

Inventor： Juin-Hwey Chen ,  Hong-Goo Kang ,  Robert W. Zopf ,  Jes Thyssen

IPC: G10L21/04 ,  G10L11/06

CPC classification number: G10L19/22 ,  G10L19/08

Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

Abstract translation: 描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。 根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。 编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。 通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。 此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

公开(公告)号：US20120121100A1

公开(公告)日：2012-05-17

申请号：US13250355

申请日：2011-09-30

Applicant: Xianxian ZHANG ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

Inventor： Xianxian ZHANG ,  Juin-Hwey Chen ,  Huaiyu Zeng ,  Jes Thyssen

IPC: G10K11/16

CPC classification number: G10L21/0208 ,  G10L21/0272 ,  G10L2021/02165 ,  H04R1/245 ,  H04R2410/07

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this tact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract translation: 不同于声音的压力波，无处不在，由风引起的空气湍流通常是一个相当地方的事件。 因此，在使用两个或多个空间分离的麦克风拾取声音信号（例如语音）的系统中，由麦克风之一拾取的风噪声通常不会被拾取（或至少不相同的程度），由 另一个麦克风。 描述利用这种技巧和其他技术来有效地检测和抑制使用空间分离的多个麦克风的风噪声的方法和装置的实施例。

公开(公告)号：US20110029317A1

公开(公告)日：2011-02-03

申请号：US12847120

申请日：2010-07-30

Applicant: Juin-Hwey Chen ,  Hong-goo Kang ,  Robert W. Zopf ,  Jes Thyssen

Inventor： Juin-Hwey Chen ,  Hong-goo Kang ,  Robert W. Zopf ,  Jes Thyssen

IPC: G10L19/00

CPC classification number: G10L19/22 ,  G10L19/08

Abstract: Systems and methods are described that utilize dynamic time scale modification (TSM) to achieve reduced bit rate audio coding. In accordance with embodiments, different levels of TSM compression are selectively applied to segments of an input speech signal prior to encoding thereof by an encoder. Encoded TSM-compressed segments are received at a decoder which decodes such segments and then applies an appropriate level of TSM decompression to each based on information received from the encoder. By selectively applying different levels of TSM compression to segments of an input speech signal prior to encoding, a coding bit rate associated with the encoder/decoder is reduced. Furthermore, by selecting a level of TSM compression for each segment of the input speech signal that takes into account certain local characteristics of that signal, such bit rate reduction is provided without introducing unacceptable levels of distortion into an output speech signal produced by the decoder.

Abstract translation: 描述了利用动态时间尺度修正（TSM）来实现降低的比特率音频编码的系统和方法。 根据实施例，在由编码器对TSM压缩进行编码之前，将不同级别的TSM压缩选择性地应用于输入语音信号的段。 编码的TSM压缩段在解码器处被接收，解码器对这些段进行解码，然后基于从编码器接收的信息向每个TSM解压缩应用适当级别的TSM解压缩。 通过在编码之前选择性地对输入语音信号的段应用不同级别的TSM压缩，减少与编码器/解码器相关联的编码比特率。 此外，通过选择考虑到该信号的某些局部特性的输入语音信号的每个段的TSM压缩级别，提供这样的比特率降低，而不会将不可接受的失真电平引入到由解码器产生的输出语音信号中。

公开(公告)号：US20090287496A1

公开(公告)日：2009-11-19

申请号：US12510445

申请日：2009-07-28

Applicant: Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

Inventor： Jes Thyssen ,  Wilfrid LeBlanc ,  Juin-Hwey Chen

IPC: G10L19/00 ,  H03G3/00

CPC classification number: H03G9/025 ,  H03G7/007 ,  H03G9/005

Abstract: A loudness enhancement system and method is described that increases the loudness of an audio signal being played back by an audio device that places limits on the dynamic range of the audio signal. In an embodiment, the loudness enhancement system and method compresses the audio signal to an adaptively-determined compression limit that is greater than or equal to a maximum desired output level and then applies an adaptively-determined degree of soft clipping to the compressed audio signal. The compression limit and degree of soft clipping may be determined based on an overload measure that is calculated for successive portions of the audio signal. The loudness enhancement system and method advantageously operates in a manner that generates less distortion than the method of simply over-driving the audio signal such that hard-clipping occurs.

Abstract translation: 描述了一种响度增强系统和方法，其增加了对限制音频信号的动态范围的音频设备正在播放的音频信号的响度。 在一个实施例中，响度增强系统和方法将音频信号压缩到大于或等于最大期望输出电平的自适应确定的压缩极限，然后对压缩音频信号应用自适应确定的软限幅度。 可以基于为音频信号的连续部分计算的过载测量来确定软限幅的压缩极限和程度。 响度增强系统和方法有利地以比简单地过度驱动音频信号的方法产生更少的失真的方式进行操作，从而发生硬削波。

公开(公告)号：US20090281803A1

公开(公告)日：2009-11-12

申请号：US12464590

申请日：2009-05-12

Applicant: Juin-Hwey Chen ,  Jes Thyssen

Inventor： Juin-Hwey Chen ,  Jes Thyssen

IPC: G10L19/14

CPC classification number: G10L21/0208 ,  G10L19/012 ,  G10L21/0232

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Abstract translation: 描述了一种语音清晰度增强（SIE）系统和方法，当音频设备位于具有大声背景噪声的环境中时，提高了由音频设备重放的语音信号的清晰度。 在一个实施例中，音频设备包括近端电话终端，并且语音信号包括通过通信网络从远端电话终端接收的用于在近端电话终端回放的语音信号。