公开(公告)号：US09319510B2

公开(公告)日：2016-04-19

申请号：US13768946

申请日：2013-02-15

Applicant: Qualcomm Incorporated

Inventor： Lae-Hoon Kim ,  Sang-Uk Ryu ,  Jongwon Shin

IPC: H04M1/76 ,  H04M3/22 ,  H04M9/08 ,  G10L21/038 ,  G10L17/00

CPC classification number: H04M3/22 ,  G10L17/00 ,  G10L21/038 ,  H04M9/08

Abstract: A personalized (i.e., speaker-derivable) bandwidth extension is provided in which the model used for bandwidth extension is personalized (e.g., tailored) to each specific user. A training phase is performed to generate a bandwidth extension model that is personalized to a user. The model may be subsequently used in a bandwidth extension phase during a phone call involving the user. The bandwidth extension phase, using the personalized bandwidth extension model, will be activated when a higher band (e.g., wideband) is not available and the call is taking place on a lower band (e.g., narrowband).

Abstract translation: 提供个性化（即，可引发扬声器）带宽扩展，其中用于带宽扩展的模型被个性化（例如，定制）给每个特定用户。 执行训练阶段以生成针对用户个性化的带宽扩展模型。 在涉及用户的电话呼叫期间，该模型可以随后用于带宽扩展阶段。 当较高频带（例如，宽带）不可用并且呼叫正在较低频带（例如，窄带）上发生时，使用个性化带宽扩展模型的带宽扩展阶段将被激活。

公开(公告)号：US20160005408A1

公开(公告)日：2016-01-07

申请号：US14850776

申请日：2015-09-10

Applicant: QUALCOMM Incorporated

Inventor： Erik Visser ,  Lae-Hoon Kim ,  Pei Xiang

IPC: G10L19/008

CPC classification number: G10L19/008 ,  G10L19/00 ,  H04S3/006

Abstract: A method for encoding three dimensional audio by a wireless communication device is disclosed. The wireless communication device detects an indication of a plurality of localizable audio sources. The wireless communication device also records a plurality of audio signals associated with the plurality of localizable audio sources. The wireless communication device also encodes the plurality of audio signals.

Abstract translation: 公开了一种由无线通信设备对三维音频进行编码的方法。 无线通信设备检测多个本地化音频源的指示。 无线通信设备还记录与多个可本地化音频源相关联的多个音频信号。 无线通信设备还对多个音频信号进行编码。

公开(公告)号：US20150245133A1

公开(公告)日：2015-08-27

申请号：US14191321

申请日：2014-02-26

Applicant: QUALCOMM Incorporated

Inventor： Lae-Hoon Kim ,  Phuong Lam Ton ,  Erik Visser ,  Jeremy P. Toman ,  Francis Bernard MacDougall

IPC: H04R1/32 ,  G06K9/00 ,  G10L15/25

CPC classification number: H04R3/005 ,  G06K9/00221 ,  G06K9/00315 ,  G10L15/25 ,  G10L21/02 ,  G10L21/0208 ,  G10L2021/02166 ,  H04N7/147 ,  H04R1/326 ,  H04R1/406

Abstract: Systems, devices, and methods are described for recognizing and focusing on at least one source of an audio communication as part of a communication including a video image and an audio communication derived from two or more microphones when a relative position between the microphones is known. In certain embodiments, linked audio and video focus areas providing location information for one or more sound sources may each be associated with different user inputs, and an input to adjust a focus in either the audio or video domain may automatically adjust the focus in the another domain.

Abstract translation: 描述了系统，设备和方法，用于当麦克风之间的相对位置是已知的时，用于识别和聚焦至少一个音频通信源，作为包括从两个或更多个麦克风导出的视频图像和音频通信的通信的一部分。 在某些实施例中，提供一个或多个声源的位置信息的链接的音频和视频焦点区域可以各自与不同的用户输入相关联，并且用于调整音频或视频域中的焦点的输入可以自动调整另一个 域。

公开(公告)号：US20140146984A1

公开(公告)日：2014-05-29

申请号：US13830894

申请日：2013-03-14

Applicant: QUALCOMM INCORPORATED

Inventor： Lae-Hoon Kim ,  Pei Xiang

IPC: H04S3/00

CPC classification number: H04R5/02 ,  H04R5/00 ,  H04R2205/024 ,  H04R2420/07 ,  H04S3/00 ,  H04S7/308 ,  H04S2400/13

Abstract: In general, techniques are described for performing constrained dynamic amplitude panning in collaborative sound systems. A headend device comprising one or more processors may perform the techniques. The processors may be configured to identify, for a mobile device participating in a collaborative surround sound system, a specified location of a virtual speaker of the collaborative surround sound system and determine a constraint that impacts playback of audio signals rendered from an audio source by the mobile device. The processors may be further configure to perform dynamic spatial rendering of the audio source with the determined constraint to render audio signals that reduces the impact of the determined constraint during playback of the audio signals by the mobile device.

Abstract translation: 通常，描述了在协作声音系统中进行约束动态幅度平移的技术。 包括一个或多个处理器的头端设备可以执行这些技术。 处理器可以被配置为针对参与协作环绕声系统的移动设备识别协作环绕声系统的虚拟扬声器的指定位置，并且确定影响由音频源呈现的音频信号的再现的约束 移动设备。 处理器可以被进一步配置为利用所确定的限制来执行音频源的动态空间渲染，以渲染音频信号，以减少由移动设备重放音频信号期间确定的约束的影响。

公开(公告)号：US20140003635A1

公开(公告)日：2014-01-02

申请号：US13801021

申请日：2013-03-13

Applicant: QUALCOMM INCORPORATED

Inventor： Asif Iqbal Mohammad ,  Lae-Hoon Kim ,  Erik Visser

IPC: G10K11/16

CPC classification number: G10K11/16 ,  G10L2021/02166 ,  H04R3/005 ,  H04R3/02 ,  H04R2430/23 ,  H04S7/301 ,  H04S7/305

Abstract: A method includes, while operating an audio processing device in a use mode, retrieving first direction of arrival (DOA) data corresponding to a first audio output device from a memory of the audio processing device and generating a first null beam directed toward the first audio output device based on the first DOA data. The method also includes retrieving second DOA data corresponding to a second audio output device from the memory of the audio processing device and generating a second null beam directed toward the second audio output device based on the second DOA data. The first DOA data and the second DOA data are stored in the memory during operation of the audio processing device in a calibration mode.

Abstract translation: 一种方法包括在以使用模式操作音频处理设备的同时，从音频处理设备的存储器检索对应于第一音频输出设备的第一到达方向（DOA）数据，并产生朝向第一音频的第一空波束 输出设备基于第一个DOA数据。 该方法还包括从音频处理设备的存储器检索对应于第二音频输出设备的第二DOA数据，并且基于第二DOA数据生成指向第二音频输出设备的第二空波束。 第一DOA数据和第二DOA数据在校准模式下在音频处理装置的操作期间被存储在存储器中。

公开(公告)号：US20140003611A1

公开(公告)日：2014-01-02

申请号：US13933005

申请日：2013-07-01

Applicant: QUALCOMM Incorporated

Inventor： Asif I. Mohammad ,  Lae-Hoon Kim ,  Erik Visser

IPC: H04R3/00

CPC classification number: H04R3/005 ,  G10L2021/02082 ,  H04B3/20 ,  H04R2430/23

Abstract: A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Abstract translation: 描述了一种通过电子设备进行回波减少的方法。 该方法包括使至少一个扬声器归零。 该方法还包括基于一组声学路径混合一组运行时音频信号以确定参考信号。 该方法还包括接收至少一个基于该组运行时音频信号的复合音频信号。 该方法还包括基于参考信号减少至少一个复合音频信号中的回波。

公开(公告)号：US20130300648A1

公开(公告)日：2013-11-14

申请号：US13674789

申请日：2012-11-12

Applicant: QUALCOMM INCORPORATED

Inventor： Lae-Hoon Kim ,  Jongwon Shin ,  Erik Visser

IPC: G06F3/01

CPC classification number: G06F3/013 ,  G01S3/8083 ,  G04G21/00 ,  G06F1/1613 ,  G06F3/011 ,  G10L17/00 ,  G10L25/48 ,  G10L2021/02166 ,  H04M3/568 ,  H04N7/15 ,  H04R1/406 ,  H04R3/005 ,  H04R29/005 ,  H04S7/304

Abstract: Disclosed is an application interface that takes into account the user's gaze direction relative to who is speaking in an interactive multi-participant environment where audio-based contextual information and/or visual-based semantic information is being presented. Among these various implementations, two different types of microphone array devices (MADs) may be used. The first type of MAD is a steerable microphone array (a.k.a. a steerable array) which is worn by a user in a known orientation with regard to the user's eyes, and wherein multiple users may each wear a steerable array. The second type of MAD is a fixed-location microphone array (a.k.a. a fixed array) which is placed in the same acoustic space as the users (one or more of which are using steerable arrays).

Abstract translation: 公开了一种应用界面，其考虑到在呈现基于音频的上下文信息和/或基于视觉的语义信息的交互式多参与者环境中的用户的注视方向。 在这些各种实施方案中，可以使用两种不同类型的麦克风阵列装置（MAD）。 第一类型的MAD是可操纵的麦克风阵列（一个可操纵的阵列），其由用户以相对于用户的眼睛已知的方向佩戴，并且其中多个用户可以佩戴可操纵的阵列。 第二种类型的MAD是固定位置麦克风阵列（即固定阵列），其被放置在与用户（其中​​一个或多个使用可控阵列）相同的声学空间中。

公开(公告)号：US20130282369A1

公开(公告)日：2013-10-24

申请号：US13827894

申请日：2013-03-14

Applicant: QUALCOMM INCORPORATED

Inventor： Erik Visser ,  Lae-Hoon Kim ,  Jongwon Shin ,  Yinyi Guo ,  Sang-Ut Ryu ,  Andre Gustavo P. Schevciw

IPC: G10L21/0208

CPC classification number: G10L21/0208 ,  G10L15/20 ,  G10L21/0316 ,  G10L25/93 ,  G10L2021/02165

Abstract: A method for signal level matching by an electronic device is described. The method includes capturing a plurality of audio signals from a plurality of microphones. The method also includes determining a difference signal based on an inter-microphone subtraction. The difference signal includes multiple harmonics. The method also includes determining whether a harmonicity of the difference signal exceeds a harmonicity threshold. The method also includes preserving the harmonics to determine an envelope. The method further applies the envelope to a noise-suppressed signal.

Abstract translation: 描述了一种由电子设备进行信号电平匹配的方法。 该方法包括从多个麦克风中捕获多个音频信号。 该方法还包括基于麦克风间减法确定差分信号。 差分信号包括多个谐波。 该方法还包括确定差分信号的谐波是否超过谐波阈值。 该方法还包括保存谐波以确定信封。 该方法进一步将包络应用于噪声抑制信号。

公开(公告)号：US12069425B2

公开(公告)日：2024-08-20

申请号：US18349920

申请日：2023-07-10

Applicant: Qualcomm Incorporated

Inventor： Lae-Hoon Kim ,  Dongmei Wang ,  Fatemeh Saki ,  Taher Shahbazi Mirzahasanloo ,  Erik Visser ,  Rogerio Guedes Alves

IPC: A61F11/06 ,  G10K11/16 ,  H03B29/00 ,  H04R1/10

CPC classification number: H04R1/1083 ,  H04R1/1075 ,  H04R2420/07 ,  H04R2460/01 ,  H04R2460/13

Abstract: A wearable device may include a processor configured to detect a self-voice signal, based on one or more transducers. The processor may be configured to separate the self-voice signal from a background signal in an external audio signal based on using a multi-microphone speech generative network. The processor may also be configured to apply a first filter to an external audio signal, detected by at least one external microphone on the wearable device, during a listen through operation based on an activation of the audio zoom feature to generate a first listen-through signal that includes the external audio signal. The processor may be configured to produce an output audio signal that is based on at least the first listen-through signal that includes the external signal, and is based on the detected self-voice signal.

公开(公告)号：US12051429B2

公开(公告)日：2024-07-30

申请号：US18138684

申请日：2023-04-24

Applicant: QUALCOMM Incorporated

Inventor： Lae-Hoon Kim ,  Shankar Thagadur Shivappa ,  S M Akramus Salehin ,  Shuhua Zhang ,  Erik Visser

IPC: G10L19/008 ,  G10L19/002 ,  G10L19/038 ,  H04R5/00

CPC classification number: G10L19/038 ,  G10L19/002 ,  H04R5/00 ,  G10L19/008 ,  H04R2430/21 ,  H04S2420/11

Abstract: A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are configured to apply one adaptive network, based on a constraint that includes preservation of a spatial direction of one or more audio sources in the soundfield at the different time segments, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint. The one or more processors are also configured to apply an additional adaptive network.