公开(公告)号：US09437209B2

公开(公告)日：2016-09-06

申请号：US14401198

申请日：2013-08-01

Applicant: Goertek, Inc.

Inventor： Liu Song ,  Bo Li ,  Shasha Lou

IPC: G10L21/00 ,  G10L21/0208 ,  H04M1/60 ,  G10L25/78 ,  G10L21/02 ,  G10L21/0216

CPC classification number: G10L21/0208 ,  G10L21/02 ,  G10L25/78 ,  G10L2021/02161 ,  H04M1/6008 ,  H04M2250/12

Abstract: The present invention discloses a speech enhancement method and device for mobile phones. By the method and device provided by the present invention, the mobile phone holding state of a user is detected when the user is talking on the phone, so that different denoising solutions will be employed according to the state of the user in holding the mobile phone. When the user holds the mobile phone normally, a solution integrating multi-microphone denoising and single-microphone denoising will be employed to effectively suppress both the steady noise and the non-steady noise; and when the user holds the mobile phone abnormally, a solution of single-microphone denoising will be employed only to suppress the steady noise. The distortion of speech by multi-microphone denoising is avoided, and the speech quality is ensured.

Abstract translation: 本发明公开了一种用于移动电话的语音增强方法和装置。 通过本发明提供的方法和装置，当用户正在通话时检测用户的手机保持状态，从而根据用户握持手机的状态采用不同的去噪方案 。 当用户正常握住手机时，将采用集成多麦克风去噪和单麦克风去噪的解决方案，有效抑制稳定噪声和非稳定噪声; 当用户手持手机不正常时，单麦克风去噪的解决方案只能用于抑制稳定的噪声。 避免了多麦克风去噪的语音失真，保证了语音质量。

公开(公告)号：US09269369B2

公开(公告)日：2016-02-23

申请号：US14407610

申请日：2013-04-01

Applicant: Goertek, Inc.

Inventor： Shasha Lou ,  Xiaojie Wu ,  Bo Li

IPC: G10L21/0208

CPC classification number: G10L21/0208 ,  G10L2021/02082

Abstract: The present invention relates to a method and device for dereverberation of single-channel speech. The method includes the following steps of framing an input single channel speech signal, and processing the frame signals as follows according to a time sequence: performing short-time Fourier transform on a current frame to obtain a power spectrum and a phase spectrum of the current frame; selecting several frames previous to the current frame and having a distance from the current frame within a set duration range, and performing linear superposition on the power spectra of these frames to estimate the power spectrum of a late reflection sound of the current frame; removing the estimated power spectrum of the late reflection sound of the current frame from the power spectrum of the current frame by a spectral subtraction method to obtain the power spectra of a direct sound and an early reflection sound of the current frame; and performing inverse short-time Fourier transform on the power spectra of the direct sound and the early reflection sound of the current frame and the phase spectrum of the current frame together to obtain a signal of the current frame after dereverberation. The dereverberation method and device can solve the problem that the estimation of a transfer function of a reverberation environment or the estimation of reverberation time is difficult in the dereverberation of single-channel speech.

Abstract translation: 本发明涉及一种用于单声道语音混响的方法和装置。 该方法包括以下步骤：根据时序对帧输入单声道语音信号进行成帧处理，如下：对当前帧执行短时傅里叶变换，获得当前的功率谱和相位谱 帧; 选择当前帧之前的几帧，并且在设定的持续时间范围内与当前帧有距离，并对这些帧的功率谱进行线性叠加，以估计当前帧的后期反射声的功率谱; 通过频谱相减法从当前帧的功率谱中去除当前帧的后期反射声的估计功率谱，以获得当前帧的直接声音和早期反射声的功率谱; 并对当前帧的直接声音和早期反射声音的功率谱和当前帧的相位频谱进行逆时空傅里叶变换，以获得混响后的当前帧的信号。 去混响方法和装置可以解决在单声道语音的混响中难以估计混响环境的传递函数的估计或混响时间的估计的问题。

公开(公告)号：US09065896B2

公开(公告)日：2015-06-23

申请号：US14351225

申请日：2013-05-30

Applicant: Goertek, Inc.

Inventor： Shasha Lou ,  Bo Li ,  Xiaojie Wu

IPC: H04J1/16 ,  H04M9/08 ,  G10L21/02 ,  H04L12/28

CPC classification number: H04M9/082 ,  G10L21/02

Abstract: Disclosed in the invention is a method and system for sampling rate mismatch correction of transmitting and receiving terminals, which can obtain a high-precision sampling rate mismatch in real time, carry out sampling rate correction on transmitting and receiving terminal signals, and send the transmitting terminal signal and the receiving terminal signal that have the same sampling rate after corrected to an echo cancellation system to carry out echo cancellation. The present invention can improve the quality of echo cancellation, simplify the computation and reduce the cost. The method for sampling rate mismatch correction of transmitting and receiving terminals provided in the embodiments of the invention comprises: calculating a transfer function of a receiving terminal signal relative to a transmitting terminal signal at each sampling timing according to the transmitting and receiving terminal signals; obtaining a transmission time delay of the transmitting and receiving terminals at each sampling timing using the transfer function; obtaining a sampling rate mismatch of the transmitting and receiving terminals at each sampling timing by means of parameter fitting using the transmission time delay and the linear relationship between the transmission time delay and the sampling rate mismatch; and adjusting the sampling rate of the transmitting terminal signal or the receiving terminal signal at each sampling timing according to the sampling rate mismatch.

Abstract translation: 本发明公开了一种发送和接收终端采样速率失配校正的方法和系统，其可以实时获得高精度采样率失配，对发送和接收终端信号进行采样率校正，并发送发送 终端信号和接收终端信号，其在校正到回波消除系统之后具有相同的采样率，以执行回声消除。 本发明可以提高回声消除的质量，简化计算并降低成本。 本发明实施例提供的发送和接收终端的采样率失配校正方法包括：根据发送和接收终端信号，在每个采样定时根据发送终端信号计算接收终端信号的传递函数; 使用传递函数在每个采样定时获得发送和接收终端的传输时间延迟; 通过使用传输时间延迟和传输时间延迟与采样率失配之间的线性关系的参数拟合，在每个采样定时获得发射和接收终端的采样率失配; 并且根据采样率不匹配在每个采样定时调整发送终端信号或接收终端信号的采样率。

公开(公告)号：US20150149160A1

公开(公告)日：2015-05-28

申请号：US14407610

申请日：2013-04-01

Applicant: Goertek, Inc.

Inventor： Shasha Lou ,  Xiaojie Wu ,  Bo Li

IPC: G10L21/0208

CPC classification number: G10L21/0208 ,  G10L2021/02082

Abstract: The present invention relates to a method and device for dereverberation of single-channel speech. The method includes the following steps of framing an input single channel speech signal, and processing the frame signals as follows according to a time sequence: performing short-time Fourier transform on a current frame to obtain a power spectrum and a phase spectrum of the current frame; selecting several frames previous to the current frame and having a distance from the current frame within a set duration range, and performing linear superposition on the power spectra of these frames to estimate the power spectrum of a late reflection sound of the current frame; removing the estimated power spectrum of the late reflection sound of the current frame from the power spectrum of the current frame by a spectral subtraction method to obtain the power spectra of a direct sound and an early reflection sound of the current frame; and performing inverse short-time Fourier transform on the power spectra of the direct sound and the early reflection sound of the current frame and the phase spectrum of the current frame together to obtain a signal of the current frame after dereverberation. The dereverberation method and device can solve the problem that the estimation of a transfer function of a reverberation environment or the estimation of reverberation time is difficult in the dereverberation of single-channel speech.

Abstract translation: 本发明涉及一种用于单声道语音混响的方法和装置。 该方法包括以下步骤：根据时序对帧输入单声道语音信号进行成帧处理，如下：对当前帧执行短时傅里叶变换，获得当前的功率谱和相位谱 帧; 选择当前帧之前的几帧，并且在设定的持续时间范围内与当前帧有距离，并对这些帧的功率谱进行线性叠加，以估计当前帧的后期反射声的功率谱; 通过频谱相减法从当前帧的功率谱中去除当前帧的后期反射声的估计功率谱，以获得当前帧的直接声音和早期反射声的功率谱; 并对当前帧的直接声音和早期反射声音的功率谱和当前帧的相位频谱进行逆时空傅里叶变换，以获得混响后的当前帧的信号。 去混响方法和装置可以解决在单声道语音的混响中难以估计混响环境的传递函数的估计或混响时间的估计的问题。

公开(公告)号：US10477319B2

公开(公告)日：2019-11-12

申请号：US16304623

申请日：2017-06-12

Applicant: Goertek Inc.

Inventor： Shasha Lou ,  Na Li ,  Bo Li

IPC: H04R1/10 ,  G06F3/01 ,  H04R29/00 ,  H04R5/04 ,  H04R5/033 ,  G06F1/3296 ,  G06F1/3231

Abstract: A wearing state detection method for a wearable device and a device. The detection method comprises: providing in the wearable device a sensor in an area capable of contacting the skin of a user, wherein the sensor outputs different measurement values when the user wears or takes off the wearable device; acquiring a base value indicating whether the wearable device is being worn; after wearing detection has been started up, collecting measurement values from the sensor at a preset sampling frequency; judging whether the wearable device is currently in a wearing state according to the measurement values and the base value; and controlling the wearable device to turn off a corresponding function that is running when the wearable device is in a non-wearing state. The method and the device can reduce the power consumption and simplify user operation.

公开(公告)号：US09805709B2

公开(公告)日：2017-10-31

申请号：US14901555

申请日：2014-07-04

Applicant: Goertek Inc

Inventor： Song Liu ,  Shasha Lou ,  Fupo Wang

IPC: G10K11/178 ,  H04R3/00 ,  H04R1/10 ,  H04R3/02

CPC classification number: G10K11/178 ,  G10K2210/1081 ,  G10K2210/3026 ,  G10K2210/3027 ,  G10K2210/3055 ,  G10K2210/506 ,  G10K2210/511 ,  H04R1/1016 ,  H04R3/00 ,  H04R3/02 ,  H04R2460/01

Abstract: The present invention discloses a howling suppression method and device applied to an ANR earphone. The method comprises: collecting signals by using a first microphone and a second microphone; wherein the first microphone is arranged in a position outside an auditory meatus when said ANR earphone is worn, and the second microphone is arranged in a position inside the auditory meatus when the ANR earphone is worn; according to a relation between signals collected by the first microphone and the second microphone, judging whether the current state of said ANR earphone is a state unable to produce a howling or a state able to produce a howling; and when the current state of said ANR earphone is a state able to produce a howling, starting processing for preventing howling production. The technical scheme can achieve that the ANR earphone does not produce a howling all the time.

公开(公告)号：US20160134986A1

公开(公告)日：2016-05-12

申请号：US14896175

申请日：2014-06-27

Applicant: Goertek, Inc.

Inventor： Song Liu ,  Shasha Lou ,  Bo Li

IPC: H04S7/00 ,  G06T7/00 ,  H04N13/00 ,  H04R5/04 ,  H04N5/232

CPC classification number: H04S7/303 ,  G06T7/50 ,  G06T2207/10028 ,  G06T2207/10148 ,  G06T2207/30201 ,  H04N5/23212 ,  H04N13/128 ,  H04N21/4223 ,  H04N21/439 ,  H04R1/403 ,  H04R5/04 ,  H04R2203/12

Abstract: A method and a system for achieving a self-adaptive surround sound. The method comprises: recognizing specific positions of a room and a user in the room by using an object recognition technology, capturing focusing images of recognized objects by controlling a camera using a focusing control technology, and recording corresponding focusing parameters (S110); calculating position information of the room relative to the camera and position information of the user relative to the camera according to the images and the parameters (S120); calculating sound beams that can achieve the surround sound at the position of the user in said room according to aforesaid calculated position information of the room and the user (S130); obtaining parameters of a filter group according to the calculated sound beams, and adjusting the filter group of a loudspeaker array according to the parameters (S140); and playing an audio signal via the loudspeaker array after the audio signal is filtered by the filter group that has been adjusted according to the parameters to form surround sound at the position of the user in the room (S150).

Abstract translation: 一种用于实现自适应环绕声的方法和系统。 该方法包括：通过使用对象识别技术来识别房间中的房间和用户的特定位置，通过使用聚焦控制技术控制摄像机捕获识别对象的聚焦图像，并记录对应的聚焦参数（S110）。 根据图像和参数，计算房间相对于相机的位置信息和用户相对于照相机的位置信息（S120）; 计算能够根据上述计算出的房间和用户的位置信息在所述房间中的用户的位置处实现环绕声的声束（S130）; 根据所计算的声束获得滤波器组的参数，并根据参数调整扬声器阵列的滤波器组（S140）; 并且在通过根据参数调整的滤波器组对音频信号进行滤波之后，通过扬声器阵列播放音频信号，以在用户在房间中的位置处形成环绕声（S150）。

公开(公告)号：US09325855B2

公开(公告)日：2016-04-26

申请号：US14411669

申请日：2013-12-12

Applicant: Goertek Inc.

Inventor： Song Liu ,  Shasha Lou ,  Bo Li

IPC: H04M9/08 ,  H04B3/20

CPC classification number: H04M9/082 ,  H04B3/20

Abstract: The present invention discloses an echo elimination device and method for a miniature hands-free voice communication system. The system comprises a receiver, a primary transmitter and an auxiliary transmitter, a distance from the primary transmitter to the receiver being greater than that from the auxiliary transmitter to the receiver. The device comprises an array echo elimination unit, a self-adaptive echo elimination unit and a residual echo elimination unit, which are structurally cascaded in turn. The array echo elimination unit, with inputs being a signal of the primary transmitter and a signal of the auxiliary transmitter, performs array filtering to obtain one path of output signals; the self-adaptive echo elimination unit, with the input signals being a signal of the receiver, the output signal of the array echo elimination unit and a signal of the auxiliary transmitter, performs self-adaptive filtering to obtain two paths of output signals; the residual echo elimination unit, with the input signals being the two paths of output signals of the self-adaptive echo elimination unit, performs voice probability estimation and echo matching to obtain an echo-eliminated voice signal. Thus, the duplex performance can be enhanced, and the phase consistency of the transmitters is not strictly required.

Abstract translation: 本发明公开了一种用于微型免提语音通信系统的回波消除装置和方法。 该系统包括接收机，主发射机和辅助发射机，从主发射机到接收机的距离大于从辅助发射机到接收机的距离。 该装置包括依次结构级联的阵列回波消除单元，自适应回波消除单元和残余回波消除单元。 阵列回波消除单元，其中输入是主发射机的信号和辅助发射机的信号，执行阵列滤波以获得一路输出信号; 自适应回波消除单元，其中输入信号是接收机的信号，阵列回波消除单元的输出信号和辅助发射机的信号，执行自适应滤波以获得两路输出信号; 剩余回波消除单元，其中输入信号是自适应回波消除单元的输出信号的两个路径，执行语音概率估计和回波匹配以获得回波消除的语音信号。 因此，可以提高双工性能，并且不严格地要求发射机的相位一致性。

公开(公告)号：US20150189431A1

公开(公告)日：2015-07-02

申请号：US14411651

申请日：2013-12-12

Applicant: Goertek Inc.

Inventor： Shasha Lou ,  Bo Li ,  Qiuchen Huang

IPC: H04R3/00

CPC classification number: H04R3/002 ,  G10L21/02 ,  G10L2021/02082 ,  G10L2021/02165 ,  H04R3/005 ,  H04R2225/43 ,  H04R2227/009

Abstract: The invention discloses a method and a device for reducing voice reverberation based on double microphones. The method comprises the steps of calculating a transfer function h(t) from a secondary microphone to a primary microphone according to an input signal x2(t) of the primary microphone and an input signal x1(t) of the secondary microphone; judging the strength of reverberation according to h(t) and calculating a regulatory factor β of a gain function by taking a tail section hr(t) of the h(t); obtaining a late reverberation estimation signal {circumflex over (r)}(t) of x2(t) with the convolution of x1(t) and hr(t); calculating the gain function according to the frequency spectrum of x2(t), β and frequency spectrum of {circumflex over (r)}(t); obtaining the reverberation removed frequency spectrum of x2(t) by multiplying the frequency spectrum of x2(t) by the gain function; and obtaining a late reverberation removed time-domain signal of x2(t) by frequency-time conversion. Thus, the late reverberation can be removed from the input signal of the primary microphone, early reverberation can be preserved, processed voice is not caused to be thin, and the voice quality is improved. Meanwhile, spectral subtraction intensity is adjusted according to the strength of the reverberation so as to ensure that the voice is not damaged on the condition that the reverberation is weak and the voice intelligibility is originally high. Accurate estimation of DOA of direct sound is not needed, and therefore the microphones are not required to have high consistency.

Abstract translation: 本发明公开了一种基于双麦克风来减少语音混响的方法和装置。 该方法包括根据主麦克风的输入信号x2（t）和次麦克风的输入信号x1（t）计算从次麦克风到主麦克风的传递函数h（t）的步骤; 根据h（t）判断混响强度，计算调节因子bgr; 通过获取h（t）的尾部hr（t）来获得增益函数; 以x1（t）和hr（t）的卷积获得x2（t）的后期混响估计信号（r（r）}（t））。 根据x2（t），＆bgr的频谱计算增益函数; （r）}（t）的频谱; 通过将x2（t）的频谱乘以增益函数获得混响去除x2（t）的频谱; 并通过频率 - 时间转换获得后期混响去除x2（t）的时域信号。 因此，可以从主麦克风的输入信号中去除后期混响，可以保持早期混响，处理的声音不会变薄，语音质量得到改善。 同时，根据混响的强度来调整频谱减法强度，以确保在混响弱并且语音清晰度最初高的条件下，声音不被损坏。 直接声音的DOA的精确估计是不需要的，因此麦克风不需要具有高的一致性。

公开(公告)号：US20150172468A1

公开(公告)日：2015-06-18

申请号：US14411669

申请日：2013-12-12

Applicant: Goertek Inc.

Inventor： Song Liu ,  Shasha Lou ,  Bo Li

IPC: H04M9/08 ,  H04B3/20

CPC classification number: H04M9/082 ,  H04B3/20

Abstract: The present invention discloses an echo elimination device and method for a miniature hands-free voice communication system. The system comprises a receiver, a primary transmitter and an auxiliary transmitter, a distance from the primary transmitter to the receiver being greater than that from the auxiliary transmitter to the receiver. The device comprises an array echo elimination unit, a self-adaptive echo elimination unit and a residual echo elimination unit, which are structurally cascaded in turn. The array echo elimination unit, with inputs being a signal of the primary transmitter and a signal of the auxiliary transmitter, performs array filtering to obtain one path of output signals; the self-adaptive echo elimination unit, with the input signals being a signal of the receiver, the output signal of the array echo elimination unit and a signal of the auxiliary transmitter, performs self-adaptive filtering to obtain two paths of output signals; the residual echo elimination unit, with the input signals being the two paths of output signals of the self-adaptive echo elimination unit, performs voice probability estimation and echo matching to obtain an echo-eliminated voice signal. Thus, the duplex performance can be enhanced, and the phase consistency of the transmitters is not strictly required.

Abstract translation: 本发明公开了一种用于微型免提语音通信系统的回波消除装置和方法。 该系统包括接收机，主发射机和辅助发射机，从主发射机到接收机的距离大于从辅助发射机到接收机的距离。 该装置包括依次结构级联的阵列回波消除单元，自适应回波消除单元和残余回波消除单元。 阵列回波消除单元，其中输入是主发射机的信号和辅助发射机的信号，执行阵列滤波以获得一路输出信号; 自适应回波消除单元，其中输入信号是接收机的信号，阵列回波消除单元的输出信号和辅助发射机的信号，执行自适应滤波以获得两路输出信号; 剩余回波消除单元，其中输入信号是自适应回波消除单元的输出信号的两个路径，执行语音概率估计和回波匹配以获得回波消除的语音信号。 因此，可以提高双工性能，并且不严格地要求发射机的相位一致性。