公开(公告)号：US20160275963A1

公开(公告)日：2016-09-22

申请号：US15166842

申请日：2016-05-27

申请人： Ronald C. Eddington, JR.

发明人： Ronald C. Eddington, JR.

IPC分类号： G10L21/0224 ,  G10L15/20 ,  G10L19/022 ,  G10L19/02 ,  G10L19/26

CPC分类号： G10L21/0224 ,  G10L15/20 ,  G10L19/0212 ,  G10L19/022 ,  G10L19/26 ,  G10L21/0208 ,  G10L2021/02082 ,  G10L2021/02166

摘要： An integrated sensor-array processor and method includes sensor array time-domain input ports to receive sensor signals from time-domain sensors. A sensor transform engine (STE) creates sensor transform data from the sensor signals and applies sensor calibration adjustments. Transducer time-domain input ports receive time-domain transducer signals, and a transducer output transform engine (TTE) generates transducer output transform data from the transducer signals. A spatial filter engine (SFE) applies suppression coefficients to the sensor transform data, to suppress target signals received from noise locations and/or amplification locations. A blocking filter engine (BFE) applies subtraction coefficients to the sensor transform data, to subtract the target signals from the sensor transform data. A noise reduction filter engine (NRE) subtracts noise signals from the BFE output. An inverse transform engine (ITE) generates time-domain data from the NRE output.

摘要翻译： 集成的传感器阵列处理器和方法包括传感器阵列时域输入端口，用于接收来自时域传感器的传感器信号。 传感器变换引擎（STE）从传感器信号创建传感器变换数据，并应用传感器校准调整。 传感器时域输入端口接收时域传感器信号，传感器输出变换引擎（TTE）从传感器信号产生传感器输出变换数据。 空间滤波器引擎（SFE）将抑制系数应用于传感器变换数据，以抑制从噪声位置和/或放大位置接收的目标信号。 阻塞滤波器引擎（BFE）对传感器变换数据应用减法系数，从传感器变换数据中减去目标信号。 降噪滤波器引擎（NRE）从BFE输出中减去噪声信号。 逆变换引擎（ITE）从NRE输出生成时域数据。

公开(公告)号：US20160232916A1

公开(公告)日：2016-08-11

申请号：US15011465

申请日：2016-01-29

申请人： Oki Electric Industry Co., Ltd.

发明人： Masaru FUJIEDA

IPC分类号： G10L21/0224 ,  G10L25/87 ,  G10L21/02

CPC分类号： G10L25/78 ,  G10L25/87 ,  G10L2025/786

摘要： An object sound period detection apparatus includes a first calculating unit, a second calculating unit, a first detecting unit, and a second detecting unit. The first calculating unit calculates a first threshold every unit time. The second calculating unit calculates a second threshold every unit time. The first detecting unit compares first feature amount based on the input signal with the first threshold and detects the object sound period in the input signal. The second detecting unit compares second feature amount based on the input signal with the second threshold, detects the object sound period in the input signal, and outputs a detecting result. The first calculating unit calculates the first threshold based on a detecting result before unit time by the second detecting unit. The second calculating unit calculates the second threshold based on a detecting result in same unit time by the first detecting unit.

摘要翻译： 物体声音周期检测装置包括第一计算单元，第二计算单元，第一检测单元和第二检测单元。 第一计算单元每单位时间计算第一阈值。 第二计算单元每单位时间计算第二阈值。 第一检测单元将基于输入信号的第一特征量与第一阈值进行比较，并检测输入信号中的对象声音周期。 第二检测单元将基于输入信号的第二特征量与第二阈值进行比较，检测输入信号中的对象声音周期，并输出检测结果。 第一计算单元基于第二检测单元在单位时间之前的检测结果来计算第一阈值。 第二计算单元基于第一检测单元在相同单位时间内的检测结果来计算第二阈值。

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：US09275624B2

公开(公告)日：2016-03-01

申请号：US13779605

申请日：2013-02-27

申请人： CANON KABUSHIKI KAISHA

发明人： Masafumi Kimura

IPC分类号： G10K11/16 ,  G10L21/0224 ,  G10L21/0356 ,  G03B31/00

CPC分类号： G10K11/16 ,  G03B31/00 ,  G10L21/0224 ,  G10L21/0356

摘要： An audio processing apparatus includes an acquisition unit configured to acquire an audio signal, and an audio processing unit configured to reduce noise contained in the audio signal, wherein the audio processing unit complements an audio signal in a section containing noise of the audio signal with a signal generated based on an audio signal in a predetermined section before the section containing noise and an audio signal in a predetermined section after the section containing noise, and wherein, in a case where noise is contained in one of the audio signal in the predetermined section before the section containing noise and the audio signal in the predetermined section after the section containing noise, the audio processing unit complements the audio signal in the section containing noise with a signal generated based on the audio signal in a noise-free section.

摘要翻译： 音频处理装置包括被配置为获取音频信号的获取单元和被配置为减少包含在音频信号中的噪声的音频处理单元，其中音频处理单元在包含音频信号的噪声的部分中补充音频信号， 基于包含噪声的部分之前的预定部分中的音频信号产生的信号和在包含噪声的部分之后的预定部分中的音频信号，并且其中，在噪声被包含在预定部分中的一个音频信号中的情况下 在包含噪声的部分和在包含噪声的部分之后的预定部分中的音频信号之前，音频处理单元在包含噪声的部分中利用在无噪声部分中基于音频信号产生的信号来补充音频信号。

公开(公告)号：US20160019894A1

公开(公告)日：2016-01-21

申请号：US14794847

申请日：2015-07-09

申请人： Panasonic Intellectual Property Corporation of America

发明人： MISAKI TSUJIKAWA ,  ISAO SHIMADA

IPC分类号： G10L15/26 ,  G10L17/22 ,  G10L21/0224 ,  G10L21/0208 ,  G10L15/30

CPC分类号： G10L15/26 ,  G10L15/30 ,  G10L17/00 ,  G10L2021/02087 ,  H04M1/72522 ,  H04M2250/74

摘要： A voice information control method for a terminal used in a system including a server device which creates text data on the basis of the voice information received from the terminal device, the method including: acquiring plurality items of first voice information; specifying a time interval that includes second voice information which is one of the plurality items of the first voice information, and which includes spoken voice of a first speaker who uses the first terminal device; and transmitting the second voice information included in the specified time interval being transmitted to the server device.

摘要翻译： 一种用于包括基于从终端设备接收的语音信息创建文本数据的服务器设备的系统中的终端的语音信息控制方法，所述方法包括：获取多个第一语音信息; 指定包括作为所述第一语音信息的多个项目之一的第二语音信息的时间间隔，并且包括使用所述第一终端设备的第一说话者的口语; 以及发送包括在指定的时间间隔内的第二语音信息被发送到服务器设备。

公开(公告)号：US09135924B2

公开(公告)日：2015-09-15

申请号：US12453173

申请日：2009-04-30

申请人： Shoji Hayakawa ,  Naoshi Matsuo

发明人： Shoji Hayakawa ,  Naoshi Matsuo

IPC分类号： H04B15/00 ,  G10L21/0208 ,  G10L21/0224 ,  G10L21/0216

CPC分类号： G10L21/0208 ,  G10L21/0216 ,  G10L21/0224 ,  G10L2021/02165 ,  G10L2021/02166

摘要： There is provided a noise suppressing device, for suppressing a noise component contained in a sound, including: at least two sound receiving parts receiving sounds from a plurality of directions containing a sound from a direction of a given sound source and converting the sounds to digital sound signals in a time domain, respectively; an estimating part acquiring both direction information on a direction of the given sound source and distance information on a distance from the given sound source based upon the digital sound signals converted by the sound receiving parts, and estimating a component value of a noise component contained in the signal by use of the direction information and the distance information; and a controlling part acquiring a control value of a suppression amount for controlling a range of a direction of the digital sound signals.

摘要翻译： 提供了一种用于抑制包含在声音中的噪声成分的噪声抑制装置，包括：至少两个声音接收部件，其从包含来自给定声源的方向的声音的多个方向接收声音，并将声音转换为数字 分别在时域中的声音信号; 估计部分基于由声音接收部分转换的数字声音信号，获取关于给定声源的方向的两个方向信息和距给定声源的距离的距离信息，以及估计包含在 信号通过使用方向信息和距离信息; 以及控制部，其获取用于控制数字声音信号的方向的范围的抑制量的控制值。

公开(公告)号：US09107010B2

公开(公告)日：2015-08-11

申请号：US13762504

申请日：2013-02-08

申请人： Cirrus Logic, Inc.

发明人： Ali Abdollahzadeh Milani

IPC分类号： H04R29/00 ,  H03G3/00 ,  G10L21/0224 ,  G10K11/178

CPC分类号： H04R29/00 ,  G10K11/178 ,  G10K2210/108 ,  G10K2210/3023 ,  G10L21/0216 ,  G10L21/0224 ,  G10L2021/02165 ,  H04R29/004

摘要： An RMS detector uses the concept of the k-NN (classifying using nearest neighbors)—algorithm in order to obtain RMS values. A rms detector using first-order regressor with a variable smoothing factor is modified to penalize samples from center of data in order to obtain RMS values. Samples which vary greatly from the background noise levels, such as speech, scratch, wind and other noise spikes, are dampened in the RMS calculation. When background noise changes, the system will track the changes in background noise and include the changes in the calculation of the corrected RMS value. A minimum tracker runs more often (e.g. two or three times) than the rate as in prior art detectors and methods, tracks the minimum rms value, which is to compute a normalized distance value, which in turn is used to normalize the smoothing factor. From this data, a corrected or revised RMS value is determined as the function of the previous RMS value multiplied by one minus the smoothing factor plus the smooth factor times the minimum rms value to output the corrected RMS for the present invention. The rms value is used to generate a reset signal for the minimum tracker and is used to avoid deadlock in the tracker, for example, when the background signal increases/decreases over time.

摘要翻译： RMS检测器使用k-NN（使用最近邻）分类的算法 - 算法来获得RMS值。 使用具有可变平滑因子的一阶回归器的有效值检测器被修改以惩罚来自数据中心的样本，以获得RMS值。 在RMS计算中，与背景噪声水平差异很大的样本（如语音，划痕，风和其他噪声尖峰）都会受到影响。 当背景噪声改变时，系统将跟踪背景噪声的变化，并包括校正RMS值的计算中的变化。 最小跟踪器比现有技术的检测器和方法中的速率更频繁（例如两次或三次），跟踪最小均方根值，其是计算归一化距离值，其又被用于归一化平滑因子。 根据该数据，校正或修正的RMS值被确定为先前RMS值乘以1的函数减去平滑因子加上平滑因子乘以最小均方根值，以输出本发明的校正RMS。 均方根值用于产生最小跟踪器的复位信号，并用于避免跟踪器中的死锁，例如当背景信号随时间增加/减小时。

公开(公告)号：US20140172420A1

公开(公告)日：2014-06-19

申请号：US14187523

申请日：2014-02-24

申请人： Huawei Technologies Co., Ltd.

发明人： Anisse Taleb ,  Jianfeng Xu ,  Liyun Pang ,  Lei Miao

IPC分类号： G10L21/0224

CPC分类号： G10L21/0224 ,  G10L19/005 ,  G10L21/04 ,  H04J3/0632 ,  H04L29/06462 ,  H04L29/06489 ,  H04L29/06517 ,  H04L29/06523 ,  H04N21/4392 ,  H04N21/4398 ,  H04N21/44209 ,  H04N21/6437

摘要： A voice or audio signal processor for processing received network packets received over a communication network to provide an output signal, the voice or audio signal processor comprising a jitter buffer being configured to buffer the received network packets, a voice or audio decoder being configured to decode the received network packets as buffered by the jitter buffer to obtain a decoded voice or audio signal, a controllable time scaler being configured to amend a length of the decoded voice or audio signal to obtain a time scaled voice or audio signal as the output voice or audio signal, and an adaptation control means being configured to control an operation of the time scaler in dependency on a processing complexity measure.

摘要翻译： 一种用于处理通过通信网络接收的接收的网络分组以提供输出信号的语音或音频信号处理器，所述语音或音频信号处理器包括被配置为缓冲所接收的网络分组的抖动缓冲器，语音或音频解码器被配置为解码 接收到的由抖动缓冲器缓冲的网络分组以获得解码的语音或音频信号，可控时间缩放器被配置为修改解码的语音或音频信号的长度，以获得时间缩放的语音或音频信号作为输出语音，或 音频信号和适配控制装置被配置为根据处理复杂度测量来控制时间缩放器的操作。

公开(公告)号：US12067998B2

公开(公告)日：2024-08-20

申请号：US17775273

申请日：2020-10-29

申请人： Sony Interactive Entertainment Inc.

发明人： Takashi Enokihara ,  Kojiro Matsuyama ,  Takafumi Ogura ,  Tetsuro Horikawa

IPC分类号： G10L21/0224 ,  A63F13/215 ,  A63F13/285 ,  A63F13/54 ,  G06F3/01 ,  G06F3/033 ,  G06F3/0338 ,  G10L21/0208 ,  G10L21/0216 ,  G10L21/0232 ,  H04R1/08 ,  H04R3/00

CPC分类号： G10L21/0224 ,  A63F13/54 ,  G06F3/016 ,  G10L21/0232 ,  H04R1/08 ,  H04R3/00 ,  G10L2021/02082 ,  G10L2021/02163

摘要： Disclosed is an operation device that executes a noise removal process of removing noise from a collected audio signal collected by a microphone. In an acting state where noise generation possibly occurs, the operation device transmits noise generation information indicating that the operation device is in the acting state where noise generation possibly occurs, and changes the details of the noise removal process, according to the noise generation information.

公开(公告)号：US12046253B2

公开(公告)日：2024-07-23

申请号：US17445064

申请日：2021-08-13

申请人： Harman International Industries, Incorporated

发明人： Sreenivas Eeshwaroju ,  Robert K. Cadena

IPC分类号： G10L21/0232 ,  G06F11/32 ,  G10L21/0224

CPC分类号： G10L21/0232 ,  G06F11/322 ,  G10L21/0224

摘要： Methods and systems are provided for detecting artifacts in an electronic signal. In an embodiment, a method is provided comprising: connecting a first input of an electronic device to a first signal line of a signal processing device, such as an amplification device; connecting a second input of the electronic device to a second signal line of the signal processing device, the second signal line being downstream from the first signal line; establishing, based on an observed behavior of a first signal on the first signal line, an expected behavior of a second signal on the second signal line; and determining whether a difference exists between the expected behavior of the second signal and an observed behavior of the second signal. If a difference is detected, the expected behavior of a second signal and the observed behavior of the second signal may be recorded for later analysis.