公开(公告)号：US09515630B2

公开(公告)日：2016-12-06

申请号：US13979030

申请日：2012-01-10

申请人： Arne Wallander

发明人： Arne Wallander

IPC分类号： H03G5/00 ,  H03G5/16 ,  G10L21/0316 ,  H04R3/04 ,  G10H1/12 ,  G10H1/46 ,  H03G3/00 ,  G10L25/90

CPC分类号： H03G5/165 ,  G10H1/125 ,  G10H1/46 ,  G10L21/0316 ,  G10L25/90 ,  H03G5/005 ,  H04R3/04 ,  H04R2430/01 ,  H04R2430/03

摘要： An improved method and arrangement for altering musical dynamics of a sound S included in a sound signal is disclosed. The altering of the musical dynamics is performed by filtering and amplification of the sound signal. The filtering is performed by the use of a parametric equalizer, the parametric equalizer having a first gain G1 and a resonance frequency fr being related to a pitch frequency fp of said sound S. The amplification is performed by an amplifier amplifying the sound signal with a second gain G2, the second gain G2 being dependent on the first gain G1.

摘要翻译： 公开了一种用于改变包括在声音信号中的声音S的音乐动力学的改进方法和装置。 通过对声音信号的滤波和放大来执行音乐动力学的改变。 通过使用参数均衡器执行滤波，参数均衡器具有第一增益G1和与所述声音S的音调频率fp相关的谐振频率fr。放大由放大器将声音信号放大 第二增益G2，第二增益G2取决于第一增益G1。

公开(公告)号：US09431981B2

公开(公告)日：2016-08-30

申请号：US14493941

申请日：2014-09-23

申请人： Google Inc.

发明人： Alejandro José Kauffmann ,  Lee Brandon Keely

IPC分类号： G10L21/034 ,  H03G3/20 ,  G10L25/78 ,  H04M1/60 ,  G10L21/0316 ,  H03G3/32 ,  H03G3/30 ,  G10L15/08 ,  G10L25/21 ,  G10L17/00 ,  G10L21/0216

CPC分类号： H03G3/20 ,  G10L17/00 ,  G10L21/0316 ,  G10L21/034 ,  G10L25/21 ,  G10L25/78 ,  G10L2015/088 ,  G10L2021/02166 ,  H03G3/3026 ,  H03G3/32 ,  H04M1/6016 ,  H04M1/6058 ,  H04M2250/74

摘要： In one example, a technique may include outputting, by a computing device associated with a user and for playback at a first volume level by an audio output device, first audio data, receiving, by the computing device, audio input data, and responsive to determining, by the computing device, that the audio input data includes speech associated with an entity different from the user, determining, by the computing device and based at least in part on the audio input data, whether to output second audio data. The method may also include, responsive to determining to output the second audio data: determining, by the computing device, a second volume level lower than the first volume level, and outputting, by the computing device and for playback at the second volume level by the audio output device, the second audio data.

摘要翻译： 在一个示例中，技术可以包括通过与用户相关联的计算设备和由音频输出设备在第一音量级别播放第一音频数据，由计算设备接收音频输入数据并且响应于 由计算设备确定音频输入数据包括与不同于用户的实体相关联的语音，由计算设备至少部分地基于音频输入数据确定是否输出第二音频数据。 该方法还可以包括：响应于确定输出第二音频数据：由计算设备确定低于第一音量电平的第二音量电平，并由计算设备输出，并且在第二音量级别通过 音频输出设备，第二音频数据。

公开(公告)号：US20160140983A1

公开(公告)日：2016-05-19

申请号：US14857681

申请日：2015-09-17

申请人： AVNERA CORPORATION

发明人： Xudong Zhao

IPC分类号： G10L21/043 ,  G10L19/008 ,  G10L21/0232 ,  G10L19/02 ,  G10L19/24 ,  G10L21/0224

CPC分类号： H03H17/0628 ,  G10L19/00 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/0356 ,  G10L2019/001 ,  H03H17/028 ,  H03M5/00 ,  H03M7/00 ,  H03M13/00 ,  H03M13/27 ,  H03M13/33

摘要： Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power.

摘要翻译： 本发明的实施例可以用于实现速率转换器，其包括：正向（音频）路径中的6个信道，每个信道具有每信道24位信号路径，110dB的端到端SNR，全部在 20 Hz至20 KHz带宽。 实施例还可以用于实现速率转换器，其具有：在反向路径中的2个信道，例如用于语音信号，每信道16位信号路径，93dB的端到端SNR，均在20Hz至20 KHz带宽。 速率转换器可以包括诸如8,11.025,12,16,22.05,24,34,44,48和96KHz的采样率。 此外，根据实施例的速率转换器可以包括低功率模式的门控时钟以节省功率。

公开(公告)号：US09299351B2

公开(公告)日：2016-03-29

申请号：US13963342

申请日：2013-08-09

申请人： Samsung Electronics Co. Ltd.

发明人： Won-Cheol Kim ,  Joon-Sang Ryu ,  Tae-Kyun Jung

IPC分类号： G10L21/00 ,  G10L21/02 ,  G10L19/005 ,  G10L21/0316 ,  G10L21/0232

CPC分类号： G10L19/005 ,  G10L21/0232 ,  G10L21/0316

摘要： A method and apparatus of suppressing a vocoder noise are provided. The method includes receiving from a channel decoder a vocoder frame and first information, the first information indicating whether the vocoder frame has an error, generating speech data by performing voice decoding on the vocoder frame, determining whether a tonal noise has been detected in the speech data, if the first information indicates that the vocoder frame has an error, and attenuating the volume of the speech data and outputting the volume-attenuated speech data through a speaker, upon detection of the tonal noise in the speech data.

摘要翻译： 提供抑制声码器噪声的方法和装置。 所述方法包括从信道解码器接收声码器帧和第一信息，所述第一信息指示声码器帧是否具有错误，通过在声码器帧上执行语音解码来产生语音数据，确定在语音中是否检测到音调噪声 数据，如果第一信息指示声码器帧具有错误，并且在检测到语音数据中的音调噪声时，通过扬声器衰减语音数据的音量并输出音量衰减的语音数据。

公开(公告)号：US20160064010A1

公开(公告)日：2016-03-03

申请号：US14829052

申请日：2015-08-18

申请人： Marvell World Trade Ltd.

发明人： Jin Xie ,  Kapil Jain

IPC分类号： G10L21/0264 ,  G10L21/0316 ,  G10L25/21

CPC分类号： G10L21/0264 ,  G10L19/0204 ,  G10L21/0232 ,  G10L21/0316 ,  G10L25/21 ,  G10L2021/02085 ,  G10L2021/02163

摘要： A system including first and second gain modules, an operator module, and a priori and posteriori modules. The first gain module applies a non-linear function to generate a gain signal based on an amplitude of a first speech signal and an estimated a priori variance of noise included in the first speech signal. The operator module generates an operator based on the gain signal and the estimated a priori variance of noise. The a priori module determines an a priori signal-to-noise ratio based on the operator. The posteriori module determines a posteriori signal-to-noise ratio based on the amplitude of the first speech signal and (ii) the estimated a priori variance of noise. The second gain module: determines a gain value based on the a priori signal-to-noise ratio and the a posteriori signal-to-noise ratio; and generates, based on the amplitude of the first speech signal and the gain value, a second speech signal that corresponds to an estimate of an amplitude of the first speech signal, where the second speech signal is substantially void of music noise.

摘要翻译： 包括第一增益模块和第二增益模块的系统，操作模块和先验和后验模块。 第一增益模块应用非线性函数，以基于第一语音信号的幅度和包括在第一语音信号中的估计的噪声的先验方差来产生增益信号。 操作员模块基于增益信号和估计的噪声的先验方差来生成操作者。 先验模块基于操作者确定先验信噪比。 后验模块基于第一语音信号的幅度确定后验信噪比，以及（ii）所估计的先验噪声方差。 第二增益模块：基于先验信噪比和后验信噪比来确定增益值; 并且基于第一语音信号和增益值的幅度生成对应于第一语音信号的幅度的估计的第二语音信号，其中第二语音信号基本上没有音乐噪声。

公开(公告)号：US09142222B2

公开(公告)日：2015-09-22

申请号：US13613792

申请日：2012-09-13

申请人： Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

发明人： Mi-Suk Lee ,  Do-Young Kim ,  Byung-Sun Lee

IPC分类号： G10L21/00 ,  G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G11B20/10 ,  H03G3/34 ,  G10L21/038

CPC分类号： G10L21/0208 ,  G10L19/24 ,  G10L21/0316 ,  G10L21/038 ,  G11B20/10527 ,  G11B2020/10555 ,  H03G3/341

摘要： An apparatus and method of improving the quality of a speech codec are provided. In the method, a first energy of a signal decoded by a low-band codec is calculated, and a second energy of a signal decoded by a low-band enhancement mode is calculated. Then, when the first energy is less than a first threshold value or less than a product of the second energy and a second threshold value, a size of the decoded signal is scaled. Accordingly, generation of a quantization error with respect to a silence segment is reduced.

摘要翻译： 提供了一种提高语音编解码器质量的装置和方法。 在该方法中，计算由低频带编解码器解码的信号的第一能量，并且计算通过低频带增强模式解码的信号的第二能量。 然后，当第一能量小于第一阈值或小于第二能量与第二阈值的乘积时，解码信号的大小被缩放。 因此，减少相对于静音段的量化误差的产生。

公开(公告)号：US09117455B2

公开(公告)日：2015-08-25

申请号：US13559450

申请日：2012-07-26

申请人： James Tracey ,  Daekyong Noh ,  Xing He

发明人： James Tracey ,  Daekyong Noh ,  Xing He

IPC分类号： G10L25/90 ,  G10L25/93 ,  G10L25/00 ,  G10L21/00 ,  G10L19/12 ,  G10L19/02 ,  G10L21/02 ,  G10L15/00 ,  G10L15/20 ,  H03G3/20 ,  H04R25/00 ,  H04B15/00 ,  G10L21/003 ,  G10L21/0316 ,  G10L21/0364 ,  G10L19/07 ,  G10L25/15

CPC分类号： G10L21/003 ,  G10L19/07 ,  G10L21/0316 ,  G10L21/0364 ,  G10L25/15

摘要： Systems and methods for adaptively processing speech to improve voice intelligibility are described. These systems and methods can adaptively identify and track formant locations, thereby enabling formants to be emphasized as they change. As a result, these systems and methods can improve near-end intelligibility, even in noisy environments. The systems and methods can be implemented in Voice-over IP (VoIP) applications, telephone and/or video conference applications (including on cellular phones, smart phones, and the like), laptop and tablet communications, and the like. The systems and methods can also enhance non-voiced speech, which can include speech generated without the vocal track, such as transient speech.

摘要翻译： 描述了用于自适应地处理语音以提高语音可懂度的系统和方法。 这些系统和方法可以自适应地识别和跟踪共振峰位置，从而使共振体在变化时被强调。 因此，即使在嘈杂的环境中，这些系统和方法也可以改善近端的清晰度。 系统和方法可以在IP语音（VoIP）应用，电话和/或视频会议应用（包括蜂窝电话，智能电话等），膝上型计算机和平板电脑等实现。 系统和方法还可以增强非语音语音，其可以包括没有声道的语音，例如瞬态语音。

公开(公告)号：US20150120310A1

公开(公告)日：2015-04-30

申请号：US14582871

申请日：2014-12-24

申请人： Roger ROBERTS

发明人： Roger ROBERTS

IPC分类号： G10L21/057 ,  G10L21/043 ,  G10L25/48

CPC分类号： G10L21/057 ,  G10L21/02 ,  G10L21/0316 ,  G10L21/043 ,  G10L25/48 ,  H04R1/1008 ,  H04R1/1091 ,  H04R25/04 ,  H04R25/552 ,  H04R25/554 ,  H04R25/70 ,  H04R2225/41 ,  H04R2225/61 ,  H04R2420/07

摘要： An audio input device is provided which can include a number of features. In some embodiments, the audio input device includes a housing, a microphone carried by the housing, and a processor carried by the housing and configured to modify an input sound signal so as to amplify frequencies corresponding to a target human voice and diminish frequencies not corresponding to the target human voice. In another embodiment, an audio input device is configured to treat an auditory gap condition of a user by extending gaps in continuous speech and outputting the modified speech to the user. In another embodiment, the audio input device is configured to treat a dichotic hearing condition of a user. Methods of use are also described.

摘要翻译： 提供了可以包括多个特征的音频输入设备。 在一些实施例中，音频输入设备包括外壳，由外壳承载的麦克风和由外壳承载并被配置为修改输入声音信号的处理器，以便放大对应于目标人声的频率并减少不对应的频率 到目标人的声音。 在另一个实施例中，音频输入设备被配置为通过扩展连续语音中的间隙并将修改的语音输出给用户来处理用户的听觉间隙状况。 在另一个实施例中，音频输入设备被配置为治疗用户的双耳听觉状况。 还描述了使用方法。

公开(公告)号：US20150043762A1

公开(公告)日：2015-02-12

申请号：US14242598

申请日：2014-04-01

申请人： SAMSUNG ELECTRONICS CO., LTD.

发明人： Sang Wook KIM ,  Jun Il SOHN ,  Dong Wook KIM

IPC分类号： H04R25/00 ,  G10L15/22 ,  G10L15/25 ,  H03G1/02

CPC分类号： H04R25/558 ,  G10L17/00 ,  G10L21/0316 ,  H03G1/02 ,  H03G3/32 ,  H04R25/405 ,  H04R25/552 ,  H04R2225/61 ,  H04R2430/01 ,  H04R2460/03 ,  H04R2460/13

摘要： A hearing device, an acoustic apparatus and a sound processing method thereof are provided. The method includes detecting audio signals, determining whether the audio signals include a voice signal of a user, and controlling amplification of the audio signals according to a result of the determination.

摘要翻译： 提供了听力装置，声学装置和声音处理方法。 该方法包括检测音频信号，确定音频信号是否包括用户的语音信号，以及根据确定的结果控制音频信号的放大。

公开(公告)号：US08949120B1

公开(公告)日：2015-02-03

申请号：US12422917

申请日：2009-04-13

申请人： Mark Every ,  Ludger Solbach ,  Carlo Murgia ,  Ye Jiang

发明人： Mark Every ,  Ludger Solbach ,  Carlo Murgia ,  Ye Jiang

IPC分类号： G10L21/00 ,  G10L15/00 ,  G10K11/16 ,  H04B15/00 ,  G10L21/0208 ,  G10L21/0316 ,  G10L21/02

CPC分类号： G10L21/0208 ,  G10L21/02 ,  G10L21/0316 ,  G10L21/034 ,  G10L2021/02166

摘要： Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

摘要翻译： 提出了用于控制噪声抵消适应性的系统和方法。 一个或多个音频信号由一个或多个相应的麦克风接收。 一个或多个信号可以被分解成频率子带。 对所述一个或多个音频信号执行与所识别的适应约束一致的噪声消除。 然后可以从频率子带重构一个或多个音频信号，并通过输出设备输出。