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公开(公告)号：US10026416B2
公开(公告)日：2018-07-17
申请号：US15317747
申请日：2015-03-31
申请人： D & M HOLDINGS INC.
发明人： Yasuo Takasu , Hiroshi Fukushima
IPC分类号： H03G3/00 , G10L21/034 , G10L21/02
CPC分类号： G10L21/034 , G10L21/0205 , H03G1/02 , H03G3/002 , H03G3/3005 , H03G3/32 , H03G5/16 , H04M1/00 , H04M9/08 , H04Q9/00 , H04R3/00
摘要： Provided is an audio system in which an audio signal to be heard can be automatically set to an appropriate volume or sound quality. The audio system is provided with: a mobile terminal device provided with a microphone unit that collects surrounding sound waves, and a transmission unit that, on the basis of the sound waves collected by the microphone unit, sets the volume or sound quality of an audio signal to be amplified by an audio device, and transmits the set audio signal volume or sound quality information to the audio device; and an audio device provided with an amplification unit that amplifies the audio signal, a control unit that adjusts the audio signal volume or sound quality by means of the amplifier, and a receiving unit that receives the audio signal volume or sound quality setting information from the mobile terminal device. The control unit controls the amplification unit so as to adjust the audio signal volume or sound quality on the basis of the audio signal volume or sound quality setting information received by the receiving unit.
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公开(公告)号：US09973870B2
公开(公告)日：2018-05-15
申请号：US15356308
申请日：2016-11-18
申请人： Bugatone Ltd.
发明人： Edmund Ben-Ami , Noam Petrank
IPC分类号： H04R29/00 , G06K9/00 , H04M9/08 , A61B5/00 , A61B7/04 , A61B5/0205 , A61B5/021 , A61B5/024 , A61B5/0295 , G06F3/16
CPC分类号： H04R29/001 , A61B5/0051 , A61B5/02055 , A61B5/02133 , A61B5/02438 , A61B5/0295 , A61B5/486 , A61B5/6803 , A61B5/6817 , A61B5/7228 , A61B5/7278 , A61B5/7405 , A61B7/045 , A61B2562/0204 , G06F3/165 , G06K9/00 , G06K2009/00939 , H04M9/08 , H04R1/1091 , H04R3/04
摘要： According to some embodiments of the present invention there is provided a method of using an earphone output speaker as a microphone for a phone call between two and/or more participants, or for measuring biometric data of a user. The method may comprise playing a received signal to an electro-acoustic output transducer of an earphone. The method may comprise instructing an audio processing circuit of a local client terminal to record an audio signal from the same electro-acoustic output transducer. The method may comprise calculating a voice signal and/or a biometric measurement based on a function combining the recorded audio signal, the received signal, and filtration coefficients, using a processing unit of the local client terminal. The method may comprise sending the voice signal and/or a biometric measurement through an output interface of the local client terminal.
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公开(公告)号：US09848261B2
公开(公告)日：2017-12-19
申请号：US15485434
申请日：2017-04-12
发明人： Junji Araki , Hiromasa Ohashi , Naoya Tanaka
摘要： A howling removing apparatus according to the present disclosure is a howling removing apparatus to be connected to a microphone and a speaker. The howling removing apparatus includes: a nonlinear converter that nonlinearly converts a sound signal input to the speaker and outputs a nonlinear signal; a delay unit that delays the sound signal by a fixed time and outputs a delay signal; a norm calculator that calculates a norm from the delay signal; a filter coefficient generator that, based on the nonlinear signal, the delay signal and the norm, generates an adaptive filter that simulates a transfer characteristic of a space where the sound signal is reproduced from the speaker and is returned to the microphone; a cancel signal generator that convolves the delay signal and the adaptive filter with each other and generates a cancel signal; and a subtracter that subtracts the cancel signal from the sound signal. When an average sound pressure level of the sound signal exceeds a threshold value, the filter coefficient generator initializes the adaptive filter, thereby sensing oscillation of the adaptive filter even when a positional relationship between the microphone and the speaker is dynamically changed, and returning an output to normal.
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公开(公告)号：US09818424B2
公开(公告)日：2017-11-14
申请号：US14889203
申请日：2014-05-05
申请人： WAVES AUDIO LTD.
发明人： Itai Neoran , Shmuel Nedjar , Roy Yam
IPC分类号： G10L21/0208 , G10L21/0264 , G10L21/0224 , G10L21/0232 , H04M9/08
CPC分类号： G10L21/0208 , G10L21/0224 , G10L21/0232 , G10L21/0264 , G10L2021/02082 , H04M9/08 , H04M9/082
摘要： A method, and one or more non-transitory computer-readable media storing instructions, and a device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed.
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公开(公告)号：US09736313B2
公开(公告)日：2017-08-15
申请号：US15294740
申请日：2016-10-16
发明人： Brian O'Donovan
CPC分类号： H04M3/568 , H04M9/08 , H04M2203/50 , H04M2203/509 , H04R3/02
摘要： A method and system for improved audio quality in teleconferencing are provided. The method includes analyzing the audio signal of multiple input lines in a teleconferencing system to detect if any two input lines contain substantially the same audio signal with a delay shorter than that of a conventional echo caused by an input line's own audio feedback via a teleconferencing server. The method further includes selecting the input line with the higher amplitude audio signal or the earlier received audio signal when two input lines with substantially the same audio signal are detected.
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公开(公告)号：US09668048B2
公开(公告)日：2017-05-30
申请号：US15011287
申请日：2016-01-29
CPC分类号： H04R3/005 , H04M9/08 , H04R2410/05 , H04R2499/11
摘要： Provided are systems and methods for contextual switching of microphones in audio devices. An example method includes detecting a change in conditions for capturing an acoustic signal by at least two microphones. A configuration is associated with the at least two microphones. The example method provides determining that the change in conditions has been stable for a pre-determined period of time. In response to the determination, the method changes the configuration associated with the at least two microphones. The conditions may include an absence or presence of far-end speech, reverberation, low/high signal-to-noise ratio, low/high signal-to-echo ratio, type of background noise, and so on. The changing of the configuration includes assigning a primary microphone and a secondary microphone based on a change in conditions. Based on the changed configuration, tuning parameters may be adjusted for noise suppression and acoustic echo cancellation.
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公开(公告)号：US20170141811A1
公开(公告)日：2017-05-18
申请号：US15323439
申请日：2015-07-03
申请人： WIZEDSP LTD.
发明人： OZ GABAI , HAIM PRIMO
IPC分类号： H04B1/401 , G10L21/0208 , H04M9/08 , H04B11/00 , H04M1/725
CPC分类号： H04B1/401 , G10L19/018 , G10L21/0208 , H04B11/00 , H04M1/72519 , H04M9/08 , H04W4/00 , H04W4/70 , H04W4/80
摘要： A communication device and method including a microphone, a first filter coupled to the microphone adapted to pass signals in an audio band only, a second filter coupled to the microphone adapted to pass signals only in an upper region of the audio band, an optional amplifier or two, an analog to digital converter (ADC) unit or two, an optional a switching unit connected between an input of the ADC unit and an output of each of the first and second filters and configured to selecting between the first and second filters, and a controller configured to control the switching unit and/or to select the second filter for acoustic communication, where the acoustic communication is adapted to the upper region of the audio band.
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8.发明授权Perceptual echo gate approach and design for improved echo control to support higher audio and conversational quality 有权感知回波门方法和设计，以改善回声控制，以支持更高的音频和会话质量公开(公告)号：US09503815B2
公开(公告)日：2016-11-22
申请号：US14289437
申请日：2014-05-28
申请人： Apple Inc.
发明人： Sean Anthony Ramprashad , Qing Yang
摘要： To address issues with present echo gate control, a method and apparatus for more intelligently operating an echo gate is described herein. In particular, the decision of whether to mute an uplink signal, or not, is formulated herein as primarily a perceptual decision based on an appropriate analysis of the perceptual interaction of the current residual echo and the current near-end signal(s). By doing so, the application of muting through an echo gate may be minimized and/or more appropriately engaged. This will lead to fewer dropouts and muting of speech onsets and offsets 1) during periods such as double-talk or 2) during periods of downlink playback in the presence of low near-end signal levels, two cases of particular importance.
摘要翻译： 为了解决当前回波门控制的问题，本文描述了用于更智能地操作回波门的方法和装置。 特别地，基于对当前残余回声和当前近端信号的感知交互的适当分析，本文中将基于上行链路信号的静音的决定在此形成为主要的感知决定。 通过这样做，通过回波门的静音的应用可以被最小化和/或更适当地接合。 两种情况尤其重要，这将导致在较低的近端信号电平存在的情况下，在下行链路重放的时段期间，诸如双向通话或2）期间的语音丢失和偏移量的丢失和静音1）。
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公开(公告)号：US09414162B2
公开(公告)日：2016-08-09
申请号：US14184877
申请日：2014-02-20
发明人： Xingping Long , Hong Liu , Muyong Cao
摘要： Systems and methods are provided for echo reduction. For example, first audio data associated with a current time unit is obtained, the first audio data being generated by an audio-acquisition device of a receiver based on at least information associated with a first audio signal acquired by the audio-acquisition device and a second audio signal output from an audio-playback device of the receiver, the second audio signal being output from the audio-playback device of the receiver based on at least information associated with second audio data transmitted previously by a sender before the current time unit; third audio data transmitted by the sender during a second time unit preceding the current time unit is obtained; and fourth audio data is obtained by performing echo reduction on the first audio data based on at least information associated with the third audio data.
摘要翻译： 提供了回波减少的系统和方法。 例如，获得与当前时间单元相关联的第一音频数据，所述第一音频数据由接收机的音频采集设备至少基于与由音频采集设备获取的第一音频信号相关联的信息生成，以及 从接收机的音频再现装置输出的第二音频信号，所述第二音频信号至少基于与当前时间单位之前的发送者先前发送的第二音频数据相关联的信息从所述接收机的音频回放装置输出; 获得在当前时间单元之前的第二时间单元期间由发送器发送的第三音频数据; 并且通过至少基于与第三音频数据相关联的信息对第一音频数据执行回波减小来获得第四音频数据。
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公开(公告)号：US20160227336A1
公开(公告)日：2016-08-04
申请号：US15011287
申请日：2016-01-29
CPC分类号： H04R3/005 , H04M9/08 , H04R2410/05 , H04R2499/11
摘要： Provided are systems and methods for contextual switching of microphones in audio devices. An example method includes detecting a change in conditions for capturing an acoustic signal by at least two microphones. A configuration is associated with the at least two microphones. The example method provides determining that the change in conditions has been stable for a pre-determined period of time. In response to the determination, the method changes the configuration associated with the at least two microphones. The conditions may include an absence or presence of far-end speech, reverberation, low/high signal-to-noise ratio, low/high signal-to-echo ratio, type of background noise, and so on. The changing of the configuration includes assigning a primary microphone and a secondary microphone based on a change in conditions. Based on the changed configuration, tuning parameters may be adjusted for noise suppression and acoustic echo cancellation.
摘要翻译： 提供了用于音频设备中麦克风的背景切换的系统和方法。 示例性方法包括通过至少两个麦克风来检测用于捕获声信号的条件变化。 配置与至少两个麦克风相关联。 该示例方法提供确定条件的变化在预定时间段内是稳定的。 响应于该确定，该方法改变与至少两个麦克风相关联的配置。 条件可能包括远端语音的缺失或存在，混响，低/高信噪比，低/高信号与回波比，背景噪声类型等。 配置的改变包括基于条件的改变来分配主麦克风和次麦克风。 基于改变的配置，可以调整调谐参数用于噪声抑制和声学回声消除。
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