公开(公告)号：US10924872B2

公开(公告)日：2021-02-16

申请号：US16079071

申请日：2017-02-16

Applicant: Dolby Laboratories Licensing Corporation

Inventor： David Gunawan ,  Glenn N. Dickins

IPC: H04R29/00 ,  H04M3/56 ,  H04R1/40 ,  H04R3/00 ,  H04M1/60 ,  G10L21/0232

Abstract: Described herein are audio capture systems and methods. One embodiment provides an audio capture system (1) including: microphones (9-11) positioned to capture respective audio signals from different directions or locations within an audio environment; a mixing module (7) configured to mix the audio signals in accordance with a mixing control signal to produce an output audio mix, wherein, upon the detection of vibration activity, the mixing control signal controls the mixing module (7) to selectively temporarily modify one or more of the audio signals to reduce the presence of noise associated with vibration activity in the output audio mix.

公开(公告)号：US10812759B2

公开(公告)日：2020-10-20

申请号：US16518887

申请日：2019-07-22

Applicant: Dolby Laboratories Licensing Corporation

Inventor： Erwin Goesnar ,  Hannes Muesch ,  David Gunawan ,  Michael Eckert ,  Glenn N. Dickins

IPC: H04N7/15 ,  H04M3/56 ,  H04N7/14 ,  H04L12/18 ,  G01S3/80 ,  G06T7/70 ,  G01S5/18 ,  H04R3/12 ,  H04S7/00 ,  G06K9/32 ,  G06K9/00

Abstract: Systems and methods are described for determining orientation of an external audio device in a video conference, which may be used to provide congruent multimodal representation for a video conference. A camera of a video conferencing system may be used to detect a potential location of an external audio device within a room in which the video conferencing system is providing a video conference. Within the detected potential location, a visual pattern associated with the external audio device may be identified. Using the identified visual pattern, the video conferencing system may estimate an orientation of the external audio device, the orientation being used by the video conferencing system to provide spatial audio video congruence to a far end audience.

公开(公告)号：US10560661B2

公开(公告)日：2020-02-11

申请号：US15918214

申请日：2018-03-12

Applicant: Dolby Laboratories Licensing Corporation

Inventor： Glenn N. Dickins ,  Ludovic Christophe Malfait ,  David Gunawan

IPC: H04N7/15 ,  G06K9/00

Abstract: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.

公开(公告)号：US10393571B2

公开(公告)日：2019-08-27

申请号：US15580242

申请日：2016-07-06

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Dong Shi ,  David Gunawan ,  Glenn N. Dickins ,  Kai Li

IPC: H04R29/00 ,  G01H7/00 ,  G10L25/21 ,  H04R1/40 ,  H04R3/00

Abstract: Example embodiments disclosed herein relate to a estimation of reverberant energy components from audio sources. A method of estimating a reverberant energy component from an active audio source (100) is disclosed. The method comprises determining a correspondence between the active audio source and a plurality of sample sources by comparing one or more spatial features of the active audio source with one or more spatial features of the plurality of sample sources, each of the sample sources being associated with an adaptive filtering model (101); obtaining an adaptive filtering model for the active audio source based on the determined correspondence (102); and estimating the reverberant energy component from the active audio source over time based on the adaptive filtering model (103). Corresponding system (800) and computer program product (900) are also disclosed.

公开(公告)号：US10319391B2

公开(公告)日：2019-06-11

申请号：US15569555

申请日：2016-04-27

Applicant: Dolby Laboratories Licensing Corporation

Inventor： David Gunawan ,  Dong Shi ,  Glenn N. Dickins

IPC: H04B15/00 ,  G10L21/0208 ,  G10L21/034 ,  H04R1/32 ,  G10L25/48

Abstract: Example embodiments disclosed herein relate to impulsive noise suppression. A method of impulsive noise suppression in an audio signal is disclosed. The method includes determining an impulsive noise related feature from a current frame of the audio signal. The method also includes detecting an impulsive noise in the current frame based on the impulsive noise related feature, and in response to detecting the impulsive noise in the current frame, applying a suppression gain to the current frame to suppress the impulsive noise. Corresponding system and computer program product of impulsive noise suppression in an audio signal are also disclosed.

公开(公告)号：US09521263B2

公开(公告)日：2016-12-13

申请号：US14419924

申请日：2013-09-09

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Yen-Liang Shue ,  David Gunawan ,  Glenn N. Dickins

IPC: H04M3/56 ,  G10L21/0208 ,  H03G3/32 ,  G10L21/0264 ,  G10L21/0324 ,  G10L25/84 ,  H03G3/30 ,  G10L25/78

CPC classification number: H04M3/568 ,  G10L21/0208 ,  G10L21/0264 ,  G10L21/0324 ,  G10L25/78 ,  G10L25/84 ,  G10L2021/02087 ,  H03G3/301 ,  H03G3/32 ,  H04M3/569

Abstract: The present document relates to audio communication systems. In particular, the present document relates to the control of the level of audio signals within audio communication systems. A method for leveling a near-end audio signal (211) using a leveling gain (214) is described. The near-end audio signal (211) comprises a sequence of segments, wherein the sequence of segments comprises a current segment and one or more preceding segments. The method comprises determining a nuisance measure (416) which is indicative of an amount of aberrant voice activity within the sequence of segments of the near-end audio signal (211); and determining the leveling gain (214) for the current segment of the near-end audio signal (211), at least based on the leveling gain (214) for the one or more preceding segments of the near-end audio signal (211), and by taking into account—according to a variable degree—an estimate of the level of the current segment of the near-end audio signal (211); wherein the variable degree is dependent on the nuisance measure (416).

Abstract translation: 本文件涉及音频通信系统。 特别地，本文件涉及音频通信系统中的音频信号的级别的控制。 描述了使用调平增益（214）来调平近端音频信号（211）的方法。 近端音频信号（211）包括段序列，其中片段序列包括当前片段和一个或多个先前片段。 该方法包括确定指示近端音频信号（211）的段的序列内的异常语音活动量的扰动度量（416）; 以及至少基于近端音频信号（211）的一个或多个先前段的调平增益（214）确定近端音频信号（211）的当前段的调平增益（214） ，并且根据可变程度考虑近端音频信号（211）的当前段的电平的估计; 其中所述可变度取决于所述妨扰措施（416）。

公开(公告)号：US20150264314A1

公开(公告)日：2015-09-17

申请号：US14435698

申请日：2013-10-11

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： Erwin Goesnar ,  David Gunawan ,  Glenn N. Dickins

IPC: H04N7/15 ,  H04B11/00 ,  H04L12/18

CPC classification number: H04N7/15 ,  H04B11/00 ,  H04L12/1818 ,  H04L12/1827

Abstract: A system and method for initiating conference calls with external devices are disclosed. Call participants are sent conference invitation and conference information regarding the designated conference call. This conference information is stored on the participant's external device. When the participants arrive at a conference call location having a conferencing device, the conferencing device is capable of communicating with the external device, initiating communications, exchanging conference information. If the participant is verified and/or authorized, the conference system may send the IP address of the conference device to the conference system to initiate the conference call. In one embodiment, the conference device uses an ultrasound acoustic communication band to initiate the call with the external device on a semi-automated basis. An acoustic signature comprising a pilot sequence for communications synchronization may be generated to facilitate the call. Audible and aesthetic acoustic protocols may also be employed.

Abstract translation: 公开了一种用外部设备发起电话会议的系统和方法。 呼叫参与者发送有关指定电话会议的会议邀请和会议信息。 该会议信息存储在参与者的外部设备上。 当参与者到达具有会议设备的会议呼叫位置时，会议设备能够与外部设备通信，发起通信，交换会议信息。 如果参与者被验证和/或授权，则会议系统可以将会议设备的IP地址发送到会议系统以发起会议呼叫。 在一个实施例中，会议设备使用超声波通信频带在半自动化的基础上与外部设备发起呼叫。 可以生成包括用于通信同步的导频序列的声学签名以便于呼叫。 还可以使用听觉和美学声学协议。

公开(公告)号：US20150058010A1

公开(公告)日：2015-02-26

申请号：US14384586

申请日：2013-03-21

Applicant: DOLBY LABORATORIES LICENSING CORPORATION

Inventor： David Gunawan ,  Glenn Dickins

IPC: G10L21/0316 ,  G10L25/78 ,  G10L25/48 ,  G10L25/18

CPC classification number: G10L21/0316 ,  G10L25/18 ,  G10L25/21 ,  G10L25/48 ,  G10L25/78

Abstract: Method for measuring level of speech determined by an audio signal in a manner which corrects for and reduces the effect of modification of the signal by the addition of noise thereto and/or amplitude compression thereof, and a system configured to perform any embodiment of the method. In some embodiments, the method includes steps of generating frequency banded, frequency-domain data indicative of an input speech signal, determining from the data a Gaussian parametric spectral model of the speech signal, and determining from the parametric spectral model an estimated mean speech level and a standard deviation value for each frequency band of the data; and generating speech level data indicative of a bias corrected mean speech level for each frequency band, including using at least one correction value to correct the estimated mean speech level for the frequency band, where each correction value has been predetermined using a reference speech model.

Abstract translation: 一种用音频信号测定的语音水平的方法，该方法通过增加噪声对其进行修正和/或降低其变化的影响和/或对其进行幅度压缩，以及被配置为执行该方法的任何实施例的系统 。 在一些实施例中，该方法包括以下步骤：产生表示输入语音信号的频带，频域数据，根据数据确定语音信号的高斯参数频谱模型，以及从参数频谱模型确定估计的平均语音电平 和数据的每个频带的标准偏差值; 以及生成指示针对每个频带的偏置校正的平均语音电平的语音电平数据，包括使用至少一个校正值来校正所述频带的估计平均语音电平，其中每个校正值已经使用参考语音模型预先确定。

公开(公告)号：US20140278380A1

公开(公告)日：2014-09-18

申请号：US14192649

申请日：2014-02-27

Applicant: Dolby Laboratories Licensing Corporation

Inventor： David Gunawan ,  Glenn N. Dickins ,  Paul Holmberg ,  Richard J. Cartwright

IPC: G10L21/0232

CPC classification number: G10L21/0232 ,  G10L21/0208 ,  H04M3/18 ,  H04M3/568 ,  H04M9/082

Abstract: In some embodiments, a method for modifying noise captured at endpoints of a teleconferencing system, including steps of capturing noise at each endpoint, and modifying the captured noise to generate modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set. In other embodiments, a teleconferencing method including steps of: at endpoints of a teleconferencing system, determining audio frames indicative of audio captured at each endpoint, each of a subset of the frames indicative of noise but not a significant level of speech; and at each endpoint, generating modified frames indicative of modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set, and generating encoded audio including by encoding the modified frames. Other aspects are systems configured to perform any embodiment of the method.

Abstract translation: 在一些实施例中，一种用于修改在电话会议系统的端点处捕获的噪声的方法，包括在每个端点处捕获噪声的步骤，以及修改所捕获的噪声，以产生具有与目标频谱匹配的频率幅度谱和空间属性的修正噪声 设置与目标空间属性集匹配。 在其他实施例中，电话会议方法包括以下步骤：在电话会议系统的端点处，确定指示在每个端点处捕获的音频的音频帧，指示噪声而不是显着级别的语音的帧的子集中的每一个; 并且在每个端点处，生成指示具有与目标频谱匹配的频率幅度谱和与目标空间属性集匹配的空间属性集合的频率幅度谱的修改后的帧，以及通过编码修改的帧来生成编码音频。 其他方面是被配置为执行该方法的任何实施例的系统。

公开(公告)号：US12273698B2

公开(公告)日：2025-04-08

申请号：US18255550

申请日：2021-12-02

Applicant: Dolby Laboratories Licensing Corporation

Inventor： Benjamin John Southwell ,  David Gunawan ,  Mark R. P. Thomas ,  Christopher Graham Hines

IPC: H04S7/00 ,  H04B1/707

Abstract: Some methods may involve receiving a first content stream that includes first audio signals, rendering the first audio signals to produce first audio playback signals, generating first direct sequence spread spectrum (DSSS) signals, generating first modified audio playback signals by inserting the first DSSS signals into the first audio playback signals, and causing a loudspeaker system to play back the first modified audio playback signals, to generate first audio device playback sound. The method(s) may involve receiving microphone signals corresponding to at least the first audio device playback sound and to second through Nth audio device playback sound corresponding to second through Nth modified audio playback signals (including second through Nth DSSS signals) played back by second through Nth audio devices, extracting second through Nth DSSS signals from the microphone signals and estimating at least one acoustic scene metric based, at least partly, on the second through Nth DSSS signals.