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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 7557. Отображено 100.
22-03-2012 дата публикации

Signal processing method, apparatus and program

Номер: US20120072210A1
Автор: Kaoru Suzuki, Tadashi Amada
Принадлежит: Toshiba Corp

In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

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Номер записи: 1
26-04-2012 дата публикации

Echo canceler

Номер: US20120099723A1
Автор: Noritaka Kokido, Shin Kato, Takuya Taniguchi
Принадлежит: Individual

An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

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Номер записи: 2
31-05-2012 дата публикации

Apparatus And Method For Cancelling Echo In Joint Time Domain And Frequency Domain

Номер: US20120136654A1
Автор: Shasha Lou, Song Liu
Принадлежит: Goertek Inc

Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain. The method includes: receiving an input receiver signal and an input transmitter signal; implementing echo cancellation on the received transmitter signal, based on the received receiver signal, by using a first echo canceller which is either a time domain echo canceller or a frequency domain echo canceller, to obtain a first echo-cancelled signal; implementing echo cancellation again on the first echo-cancelled signal, based on the received receiver signal, by using a second echo canceller which is the other one of the time domain echo canceller and the frequency domain echo canceller, to obtain a second echo-cancelled signal; wherein, the filter parameters of the second filter of the second echo canceller is updated based on the second echo-cancelled signal, and the first and second echo canceller respectively include the corresponding first and second filters. By using said method in the present invention, fast response to echo reflecting environment can be achieved with little residual echo, thus the effect of echo cancellation is entirely improved.

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Номер записи: 3
07-06-2012 дата публикации

System and method for echo reduction in audio and video telecommunications over a network

Номер: US20120140918A1
Автор: Marcus Lee Sherry
Принадлежит: PAGEBITES Inc

A method and a system use an intermediate server to process the communication between two parties, so as to eliminate echoes between them. The server performs echo cancellation in a network-based voice communication system handling a large number of conversations. In one implementation, the server allocates two echo cancellation modules to each conversation, with each echo cancellation module including (a) a communication interface for communicating with a client program associated with the echo cancellation module; (b) a first buffer for storing audio data received from the client program for transmission to another echo cancellation module; (c) a second buffer for storing audio data received from the other echo cancellation module for transmitting to the associated client program; and (d) a set of filters using the audio data in both the first buffer and the second buffer to cancel echoes in the audio data in the second buffer.

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Номер записи: 4
19-07-2012 дата публикации

Device and method for controlling damping of residual echo

Номер: US20120183133A1
Автор: Christian Schuldt, Fredrick Lindström, Ingvar Claesson
Принадлежит: Limes Audio AB

The present invention relates to a device, such as a communication device, comprising an adaptive foreground filter configured to calculate a first echo estimation signal based on a first input signal, and an adaptive background filter being more rapidly adapting than the foreground filter and configured to calculate a second echo estimation signal based on said first input signal. Embodiments of the device further comprise damping control means for controlling damping of an echo-cancelled output signal. The device in various embodiments includes that the damping control means is configured to calculate a maximum echo estimation signal using both the first and the second echo estimation signals, and control the damping of the echo-cancelled output signal based on said maximum echo estimation signal and/or a signal derived from said maximum echo estimation signal.

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Номер записи: 5
02-08-2012 дата публикации

Speech quality enhancement in telecommunication system

Номер: US20120195423A1
Автор: Seungil Kim
Принадлежит: EMPIRE TECHNOLOGY DEVELOPMENT LLC

Technologies are generally described for an echo cancelling device of a telecommunication system. In some examples, an echo canceling device may include a noise reduction unit configured to reduce a background noise around a near-end talker from a near-end signal provided by a microphone, a double talk detector configured to detect a double talk event based on the noise-reduced near-end signal and a far-end signal, and a filtering unit configured to receive the far-end signal and the near-end signal provided by the microphone.

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Номер записи: 6
20-09-2012 дата публикации

Nonlinear reference signal processing for echo suppression

Номер: US20120237047A1
Автор: Glenn N. Dickins, Timothy J. Neal
Принадлежит: Dolby Laboratories Licensing Corp

An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

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Номер записи: 7
22-11-2012 дата публикации

Method and apparatus for reducing noise pumping due to noise suppression and echo control interaction

Номер: US20120294453A1
Автор: Andrew John Macdonald, Björn Völcker, Jan Skoglund
Принадлежит: Andrew John Macdonald, Jan Skoglund, Volcker Bjoern

An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

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Номер записи: 8
14-02-2013 дата публикации

Methods and apparatuses for echo cancelation with beamforming microphone arrays

Номер: US20130039504A1
Автор: Ashutosh Pandey, Darrin T. Thurston, David K. Lambert, Tracy A. Bathurst
Принадлежит: ClearOne Communications Inc

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of microphones oriented to cover a corresponding plurality of direction vectors and to develop a corresponding plurality of microphone signals. A processor is operably coupled to the plurality of microphones. The processor is configured to perform a beamforming operation to combine the plurality of microphone signals to a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals. The processor is also configured perform an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals and select one of the plurality of combined echo-canceled signals for transmission.

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Номер записи: 9
14-03-2013 дата публикации

Echo Cancelling-Codec

Номер: US20130066638A1
Автор: Phillip Alan Hetherington, Shree Paranjpe, Steven George MASON
Принадлежит: QNX Software Systems Ltd

Echo-cancellation is utilized in terminal devices such as speakerphones to compensate for acoustic echoes and interaction of the audio signal with the surrounding environment. An echo-cancelling codec incorporates encoding, decoding and acoustic echo-cancellation in a single device, enabling processing to be utilized that reduces processing and memory resources. The configuration enables processing information to also be shared between encoding, decoding and acoustic echo-cancellation functions to optimize operational characteristics. The acoustic echo cancelling codec interfaces between the amplitude signal domain, speaker and microphone, and an encoded data domain, a data interface, reducing component requirements required to provide echo-cancellation and coding functions.

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Номер записи: 10
04-04-2013 дата публикации

Methods and apparatuses for multi-channel acoustic echo cancelation

Номер: US20130083911A1
Автор: Ashutosh Pandey, Brett Harris, Bryan Shaw, David K. Lambert, Derek Graham, Michael Braithwaite, Michael Tilelli, Paul R. Bryson, Peter H. Manley, Russel S. Ericksen, Sandeep Kalra, Tracy A. Bathurst
Принадлежит: ClearOne Communications Inc

Embodiments include methods and apparatuses for echo cancelation involving multiple audio channels and the production, sensing, or a combination thereof of multiple audio channels in conferencing systems. A conferencing apparatus with a plurality of speakers configured to generate outgoing acoustic waves responsive to a multi-channel audio signal. One or more microphones are configured to sense incoming acoustic waves from the plurality of speakers and from locally produced acoustic waves from a participant of a conference to generate one or more incoming audio signals. A processor is operably coupled to the plurality of speakers and the one or more microphones. The processor is configured to perform acoustic echo cancelation on the one or more incoming audio signals relative to at least two different channels of the multi-channel audio signal.

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Номер записи: 11
23-05-2013 дата публикации

ECHO CANCELLER

Номер: US20130129078A1
Автор: Fadili Moulay, Zargouni Labib
Принадлежит: ALCATEL LUCENT

The proposed echo canceller comprises:—an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;—a subtractor for subtracting this estimate from the received signal and supplying a residual signal;—means for detecting () a ring back tone in said residual signal;—means for blocking () the received signal and replacing it by a locally generated ring back tone if a ring back tone is detected in the received signal;—a timer to determine a time period;—and means (-) for, during said time period, replacing the residual signal by some synthetic comfort noise when there is no ring back tone. 1) An echo canceller comprising:an adaptive filter for receiving a signal affected by an echo, and for supplying a filtered signal that is an estimate of the echo;a subtractor for subtracting this estimate from the received signal and supplying a residual signal; means for comparing the difference between the energy of the received signal and the energy of a transmitted signal with a first threshold, and then concluding that a ring back tone is detected when this difference is greater than this first threshold,', 'and means for comparing the energy of the transmitted signal to a second threshold, and comparing the cumulated time during which this energy is greater than the second threshold with a fourth threshold, then assuming that the ringing phase is elapsed when this cumulated time is greater than this fourth threshold;, 'means for detecting a ring back tone in said residual signal, comprisingmeans for blocking the received signal and replacing it by a locally generated ring back tone when a ring back tone has been detected in the received signal, and until the ringing phase is assumed to be elapsed.2) An echo canceller according to claim 1 , further comprising:a timer to determine a time period;and means for, during said time period, replacing the residual signal by some synthetic comfort noise when there is ...

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Номер записи: 12
13-06-2013 дата публикации

VOICE SWITCHING FOR VOICE COMMUNICATION ON COMPUTERS

Номер: US20130148801A1
Автор: He Chao, LI QIN
Принадлежит: MICROSOFT CORPORATION

A voice communication end device performs quality checks to determine whether acoustic echo cancellation would be ineffective, such as due to noise or clock drift or discontinuities between incoming and outgoing voice channels. In the case where echo cancellation would prove ineffective, the device falls back on a tri-state voice switching operation that includes a bi-direction state in which both channels are on in full duplex operation, which provides a smoother transition switching between active channels. The tri-state voice switching supports both voluntary transitions where the active user voluntarily stops to yield the active channel, and forced transitions where the active user is forcedly interrupted by the other user speaking more loudly. 1. A computer-readable media storing instructions thereon for executing a method of preventing acoustic echo in a two-way voice communication end device , the method comprising:upon starting a communication session with another communication end device, operating in a full duplex voice communication with acoustic echo cancellation mode;determining whether the voice communication with the other communication end device has sufficient quality for effective acoustic echo cancellation; andin the event that the voice communication is determined to lack sufficient quality, operating in a tri-state voice switching mode; operating in an outgoing state when voice activity is detected on an outgoing channel;', 'operating in incoming state when voice activity is detected on an incoming channel;', 'operating in a bi-directional state when voice activity ceases for over a threshold period of time while in the incoming or outgoing states;', 'from the incoming state, forcing a transition to the outgoing state when there is voice activity on both the incoming and outgoing channels, but a first energy level associated with the volume of the voice activity on outgoing channel exceeds a second energy level of the voice activity on the ...

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Номер записи: 13
20-06-2013 дата публикации

Optimizing audio processing functions by dynamically compensating for variable distances between speaker(s) and microphone(s) in a mobile device

Номер: US20130156209A1
Автор: Eric S. Mattis, Erick Visser, Joseph Robert Fitzgerald, Kwokleung Chan, Michael Joseph Contour
Принадлежит: Qualcomm Inc

Mobile communication devices, having multiple speakers and/or microphones to perform a number of audio functions, for use with mobile devices, are provided. The microphones may be housed within the communication device housing. To compensate for the unwanted signal feedback between the speakers and microphones, acoustic echo cancellation may be implemented to determine the proper distance and relative location between the speakers and microphones. Acoustic echo cancellation removes the echo from voice communications to improve the quality of the sound. The removal of the unwanted signals captured by the microphones may be accomplished by characterizing the audio signal paths from the speakers to the microphones (speaker-to-microphone path distance profile), including the distance and relative location between the speakers and microphones. The optimal distance and relative location between the speakers and microphones is provided to the user to optimize performance.

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Номер записи: 14
18-07-2013 дата публикации

Echo Canceler Circuit and Method

Номер: US20130184036A1
Автор: Owens Keith L., Piket James B., Springfield Christopher W.
Принадлежит: CONTINENTAL AUTOMOTIVE SYSTEMS, INC.

An echo canceler circuit () and method attenuates at least post-echo canceler uplink data () to produce attenuated uplink data () in response to uplink echo return loss based attenuation data (). The echo canceler circuit () includes an echo return loss based attenuation data generator () and at least an uplink data attenuator (). The echo return loss based attenuation data generator () produces the uplink echo return loss based attenuation data () in response to echo return loss data (). The echo return loss data () is based on at least one of: attenuated downlink data (), pre-echo canceler uplink data (), and/or amplifier gain data (). The uplink data attenuator () attenuates the post-echo canceler uplink data () to produce attenuated uplink data () based on the uplink echo return loss based attenuation data (). 1. An echo canceler circuit comprising:an uplink data attenuator operative to receive at least post-echo canceler uplink data and uplink echo return loss based attenuation data and in response to attenuate the post-echo canceler uplink data to produce attenuated uplink data; andan echo return loss (ERL) based attenuation data generator operatively coupled to the uplink data attenuator and operative to produce the uplink echo return loss based attenuation data in response to echo return loss data, wherein the echo return loss data is calculated based on a ratio of the attenuated downlink data to the pre-echo canceler uplink data.2. (canceled)3. The echo canceler circuit of wherein the uplink data attenuator is operative to attenuate the post-echo canceler uplink data over a period of time to produce the attenuated uplink data.4. An echo canceler circuit comprising:an uplink data attenuator operative to receive post-echo canceler uplink data and uplink echo return loss based attenuation data and in response to attenuate the post-echo canceler uplink data to produce attenuated uplink data;a downlink data attenuator operative to receive downlink data and ...

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Номер записи: 15
25-07-2013 дата публикации

SYSTEM AND METHOD FOR PROCESSING AUDIO SIGNALS RECEIVED BY A COMMUNICATION DEVICE

Номер: US20130191873A1
Автор: Pearson Larry B.
Принадлежит: AT&T Intellectual Property I, LP

A system that incorporates teachings of the present disclosure may include, for example, a communication device having a controller to receive a media program signal from a set-top box (STB), monitor ambient sound, and suppress a portion of the ambient sound according to the media program signal. The media program signal can correspond to a media program presentable at least audibly by the STB. Other embodiments are disclosed.

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Номер записи: 16
22-08-2013 дата публикации

ECHO CANCELLATION USING CLOSED-FORM SOLUTIONS

Номер: US20130216057A1
Автор: Chen Juin-Hwey, SOLLENBERGER Nelson, Thyssen Jes, Zeng Huaiyu
Принадлежит: BROADCOM CORPORATION

A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal. 1. An echo cancellation system , comprising:a filter that is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal;filter parameter determination logic that is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics; anda combiner that is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.2. The echo cancellation system of claim 1 , wherein the filter processes a frequency-domain representation of the far-end audio signal.3. The echo cancellation system of claim 1 , wherein the filter comprises a hybrid frequency-domain filter that generates the estimated echo signal by passing each of a plurality of frequency components of a frequency-domain representation of the far-end audio signal through a respective one of a plurality of time direction filters.4. The echo cancellation system of ...

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Номер записи: 17
12-09-2013 дата публикации

Method and System for Stereo Echo Cancellation for VOIP Communication Systems

Номер: US20130236004A1
Автор: James C.H. Thi, Theodore F. Rabenko
Принадлежит: Broadcom Corp

An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaptation is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

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Номер записи: 18
19-09-2013 дата публикации

Methods, apparatus and articles of manufacture to cancel echo for communication paths having long bulk delays

Номер: US20130243184A1
Автор: David Ramsden
Принадлежит: AT&T INTELLECTUAL PROPERTY I LP

Example methods, apparatus and articles of manufacture to cancel echo for a communication path having long bulk delays are disclosed. A disclosed example method includes determining a first location of a first coefficient having a largest magnitude of a first plurality of magnitudes associated with a first plurality of coefficients of a first phase; determining a second location of a second coefficient having a largest magnitude of a second plurality of magnitudes associated with a second plurality of coefficients of a second phase different than the first phase; comparing a difference between the first and second locations to a threshold; and, when the difference is less than the threshold, selecting a first offset based on a greater of the magnitude of the first coefficient and the magnitude of the second coefficient; and cancelling an echo contained in a signal using the first offset.

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Номер записи: 19
31-10-2013 дата публикации

Reduced-delay subband signal processing system and method

Номер: US20130287226A1
Автор: Yair Kerner
Принадлежит: Conexant Systems LLC

A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively.

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Номер записи: 20
05-12-2013 дата публикации

Method and apparatus for a frequency domain echo suppression filter

Номер: US20130324195A1
Автор: David Barron
Принадлежит: Continental Automotive Systems Inc

Residual frequency components of a reference signal are suppressed from an error signal. A magnitude of the frequency domain representation of the reference signal is divided by a magnitude of the frequency domain representation of LMS-filtered representation of the error signal to obtain a frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal. The frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal is multiplied by the frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal, to obtain a frequency domain signal having reduced residual frequency components of the reference signal.

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Номер записи: 21
23-01-2014 дата публикации

COMPUTATION SAVING ECHO CANCELLER FOR A WIDE BAND AUDIO SIGNAL

Номер: US20140023188A1
Автор: Fadili Moulay, Pinier François
Принадлежит: ALCATEL-LUCENT

A canceller splits a signal, transmitted from a near end terminal (ET) to a far ET, into a sub-sampled signal corresponding to a higher frequency sub-band of the signal transmitted to the far ET, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal transmitted to the far ET; and splits a signal received from a far ET into a sub-sampled signal corresponding to a higher frequency sub-band of the signal received from the far ET, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal received from the far ET. The canceller includes a first adaptive filter for filtering the sub-sampled signal corresponding to the lower frequency sub-band, a second adaptive filter for filtering the sub-sampled signal corresponding to the higher frequency sub-band, and controls the adaptation of the first and second adaptive filters so that these two adaptations are never simultaneous. 1. A wide band echo canceller comprising:means for splitting a signal transmitted from a near end terminal to a far end terminal, into a sub-sampled signal corresponding to a higher frequency sub-band of the signal transmitted to the far end terminal, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal transmitted to the far end terminal,means for splitting a signal received from a far end terminal into a sub-sampled signal corresponding to a higher frequency sub-band of the signal received from the far end terminal, and a sub-sampled signal corresponding to a lower frequency sub-band of the signal received from the far end terminal,a first adaptive filter for filtering the sub-sampled signal corresponding to the lower frequency sub-band of the signal transmitted to the far end terminal, and restituting a first filtered signal,a second adaptive filter for filtering the sub-sampled signal corresponding to the higher frequency sub-band of the signal transmitted to the far end terminal, and restituting a second filtered ...

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Номер записи: 22
02-01-2020 дата публикации

CALL QUALITY IMPROVEMENT SYSTEM, APPARATUS AND METHOD

Номер: US20200005806A1
Автор: CHOI Hyeon Sik, LEE Keun Sang, SEO Jae Pil
Принадлежит:

Provided is a call quality improvement method configured to operate a call quality improvement system and a call quality improvement apparatus by executing an artificial intelligence (AI) algorithm and/or a machine learning algorithm in a 5G environment connected for the Internet of Things. According to one embodiment of the present disclosure, the call quality improvement method may include receiving a voice signal from a far-end speaker, receiving a sound signal including a voice signal from a near-end speaker, receiving an image of a face of the near-end speaker, including lips, and extracting the voice signal of the near-end speaker from the received sound signal. 1. A call quality improvement system using lip-reading , the call quality improvement system comprising:a microphone configured to collect a sound signal including a voice signal of a near-end speaker;a speaker configured to output a voice signal from a far-end speaker;a camera configured to photograph a face of the near-end speaker, including lips; anda sound processor configured to extract the voice signal of the near-end speaker from the sound signal collected from the microphone,wherein the sound processor comprises an echo reduction module including an adaptive filter configured to filter out an echo component from the sound signal collected through the microphone based on a signal inputted to the speaker, and a filter controller configured to control the adaptive filter, andthe filter controller changes parameters of the adaptive filter based on lip movement information of the near-end speaker.2. The call quality improvement system according to claim 1 , wherein the sound processor further comprises:a noise reduction module configured to reduce a noise signal in the sound signal from the echo reduction module; anda voice reconstructor configured to reconstruct the voice signal of the near-end speaker damaged during a noise reduction process through the noise reduction module, based on the lip ...

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Номер записи: 23
07-01-2016 дата публикации

DETECTION OF DOUBLE TALK IN TELECOMMUNICATIONS NETWORKS

Номер: US20160006480A1
Автор: Tan Mizhou
Принадлежит:

In one embodiment, the presence of double talk (DT) is detected in a telecommunications network having a near-end user and a far-end user. The energies of both (1) a signal received from the far-end user by the near-end user and (2) a signal to be communicated from the near-end user to the far-end user are computed. An echo return loss (ERL) estimate is calculated based on the energy calculations, and a preliminary decision is made as to whether DT is present based on the ERL estimate and the energy calculations. If DT is detected, then a counter is set to a hangover value. If DT is not detected, then the counter is reduced. This process is repeated, and, for each iteration, a final decision as to whether DT is present is made based on the counter value. 1. One or more machine-readable storage media comprising a plurality of instructions stored thereon that , when executed , cause a controller to:generate a measure of average energy of a near-end signal in the telecommunications network;generate a measure of average energy of a far-end signal in the telecommunications network;generate a double-talk (DT) decision statistic based on the near-end average energy measure and the far-end average energy measure, wherein to generate the DT decision statistic comprises to (i) generate an estimate of echo return loss (ERL) based on the near-end average energy measure and the far-end average energy measure and (ii) generate the DT decision statistic based on the ERL estimate, the near-end average energy measure, and the far-end average energy measure; andgenerate a DT decision as to whether or not double talk is present in the telecommunications network based on the DT decision statistic.2. The one or more machine-readable storage media of claim 1 , wherein to generate the measure of average energy of the near-end signal in the telecommunications network comprises to generate a weighted moving average of instantaneous near-end energy measures claim 1 , andwherein to generate ...

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Номер записи: 24
07-01-2016 дата публикации

Audio Capture and Render Device Having a Visual Display and User Interface for Audio Conferencing

Номер: US20160006879A1
Автор: Anthony Tucker, David GUNAWAN, Gary Spittle, Glenn N. Dickins
Принадлежит: Dolby Laboratories Licensing Corp

A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.

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Номер записи: 25
07-01-2016 дата публикации

VARIABLE STEP SIZE ECHO CANCELLATION WITH ACCOUNTING FOR INSTANTANEOUS INTERFERENCE

Номер: US20160006880A1
Автор: Lu Youhong, Zad Issa Syavosh
Принадлежит:

Examples of the disclosure provide variable step size (VSS) adaptive echo cancellation in the presence of near-end noise such as dense double talk without using an explicit double talk detector and/or without using a dual-filter. During a conversation, the present value for an error signal is monitored. Based on the monitored present value for the error signal, a first function is determined. A second function is determined based on long-term statistics describing a reference signal, a near-end noise signal, and the error signal. An adaptation coefficient is calculated for the VSS adaptive filter based on the determined first function and the determined second function. The calculated adaptation coefficient is used in the VSS adaptive filter for echo cancellation against interference due to the near-end noise signal during the conversation. 1. A system capable of converging in the presence of constant double talk using a variable step size (VSS) adaptive filter , said system comprising:a memory area for storing long-term statistics describing a reference signal, a near-end noise signal, and an error signal; and monitor a present value corresponding to the error signal during a conversation;', 'determine a first function based on the monitored present value for the error signal;', 'determine a second function based on the stored long-term statistics; and', 'calculate, based on applying a first weight to the determined first function and a second weight to the determined second function, an adaptation coefficient for the VSS adaptive filter, the first weight and the second weight representing a level of confidence in the first function and the second function, respectively; and, 'a processor programmed toreduce the near-end noise signal during the conversation by applying the calculated adaptation coefficient to the VSS adaptive filter.2. (canceled)3. The system of claim 1 , wherein the processor is further programmed to update claim 1 , in real-time claim 1 , a ...

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Номер записи: 26
04-01-2018 дата публикации

SOUND EMISSION AND COLLECTION DEVICE, AND SOUND EMISSION AND COLLECTION METHOD

Номер: US20180007186A1
Автор: UKAI Satoshi, Yamakawa Takashi
Принадлежит:

A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter. 110-. (canceled)11. A sound emission and collection device comprising:a speaker;at least one microphone;a reverberation time estimation section configured to estimate a reverberation time for each frequency band in a space where the speaker and the at least one microphone are present; andan arithmetic operation section configured to specify a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time and to suppress power of the specified frequency band.12. The sound emission and collection device according to claim 11 , further comprising:at least one echo canceller configured to cancel a regression sound signal of sound emitted by the speaker from a sound collection signal output by the at least one microphone, wherein:the reverberation time estimation section estimates a reverberation time for each frequency band in a space where the speaker and the at least one microphone are present based on an adaptive filter coefficient obtained from the at least one echo canceller, andthe ...

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Номер записи: 27
03-01-2019 дата публикации

TARGET OPERATION DEVICE COMMUNICATING SYSTEM, MOBILE TERMINAL, AND TARGET OPERATION DEVICE COMMUNICATION COOPERATING METHOD

Номер: US20190007149A1
Автор: FUJITA Kouji
Принадлежит:

A permissible range of a mobile terminal that operates in cooperation with a target operation device in accordance with a user's whereabouts in a house is dynamically switched. An intercom has a function permission number control unit that generates calling information to require a cooperated operation to be carried out with the mobile terminal. The mobile terminal includes a function permission number storing unit and a CPU. The function permission number storing unit stores function permission numbers respectively set to mobile terminals. When a function permission number transmitted from the intercom is received, the CPU determines whether the function permission number coincides with a function permission number of a function permission information storing unit or not, and sets an application for carrying out the cooperated operation with the intercom to active in a case where they coincide with each other. 1. A target operation device communicating system comprising:a target operation device; andmobile terminals configured to operate in cooperation with the target operation device by communicating with the target operation device,wherein the target operation device includes:a function permission information control unit configured to generate cooperation information and first function permission information when to carry out a cooperated operation with any of the mobile terminals, the cooperation information being used to require the cooperated operation with the corresponding mobile terminal; anda first communicating unit configured to transmit the cooperation information and the first function permission information generated by the function permission information control unit to the corresponding mobile terminal,wherein each of the mobile terminals includes:a function permission information storing unit configured to store second function permission information that is set to the mobile terminal;an application storing unit configured to store an application ...

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Номер записи: 28
07-01-2021 дата публикации

HOWLING SUPPRESSION APPARATUS, AND METHOD AND PROGRAM FOR THE SAME

Номер: US20210006899A1
Автор: Harada Noboru, KOBAYASHI Kazunori, Nakagawa Akira, SAITO Shoichiro
Принадлежит: Nippon Telegraph and Telephone Corporation

A howling suppression apparatus includes: an integration processing part that obtains the maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by multiple microphones; and a howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value. 1. A howling suppression apparatus comprising:an integration processing part that obtains a maximum value among L values corresponding to n-th frames of L i-th signals, for i=1, 2, . . . , L, L being any integer equal to or greater than 2, the L i-th signals being frequency-domain signals obtained from sound signals collected by a plurality of microphones; anda howling suppression processing part that performs howling suppression processing on at least any of the L i-th signals using the maximum value.2. The howling suppression apparatus according to claim 1 , wherein the howling suppression processing part performs the howling suppression processing by utilizing a fact that howling components are out of phase among the sound signals collected by the plurality of microphones.3. The howling suppression apparatus according to or claim 1 , wherein claim 1 , at least either (i) if the maximum value is greater than a value indicating predetermined power or (ii) if a value indicating a variation in the maximum value is greater than a value indicating a predetermined variation claim 1 , the howling suppression processing part performs the howling suppression processing by multiplying at least any of the L i-th signals by a smaller one of a first gain obtained based on the maximum value and a second gain obtained based on the value indicating the variation in the maximum value.4. The howling suppression apparatus according to or claim 1 , comprising a smoothing processing part that ...

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Номер записи: 29
02-01-2020 дата публикации

ACOUSTIC ECHO SUPPRESSION DEVICE AND ACOUSTIC ECHO SUPPRESSION METHOD

Номер: US20200007690A1
Автор: MIYAMOTO Masanari, OHASHI Hiromasa, Tanaka Naoya
Принадлежит: Panasonic Intellectual Property Management Co., Ltd.

A microphone picks-up voice of a driver. A first echo suppression unit outputs a voice signal after first echo suppression based on a voice signal of the driver and a voice signal after echo suppression in the past (first reference signal) stored in a buffer memory. A second echo suppression unit outputs a voice signal after second echo suppression based on a voice signal of the driver and a voice signal after the echo suppression in the past (second reference signal) stored in a buffer memory. An output signal selector selects one of the voice signals after the first echo suppression or the voice signal after the second echo suppression according to a detection result of the presence or absence of a system variation by a system variation detector, and causes a speaker to output the selected voice signal. 1. An acoustic echo suppression device that suppresses acoustic echo in a room where a sound pick-up unit is installed , the device comprising:a first filter processor which is connected to the sound pick-up unit and outputs a first sound signal obtained by updating an echo component included in a picked-up sound signal acquired by the sound pick-up unit at a first rate;a second filter processor which is connected to the sound pick-up unit and outputs a second sound signal obtained by updating the echo component included in the picked-up sound signal at a second rate faster than that of the first filter processor, against a sudden variation in a sound field environment in the room;a detector which detects presence or absence of the variation in the sound field environment in the room; andan output selector which selects one of the first sound signal and the second sound signal according to a detection result of the presence or absence of the variation in the sound field environment in the room and causes a voice output unit to output the selected sound signal.2. The acoustic echo suppression device of claim 1 ,wherein an update rate of a coefficient of a filter ...

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Номер записи: 30
08-01-2015 дата публикации

COMMUNICATION DEVICE WITH ECHO SUPPRESSION

Номер: US20150011266A1
Автор: DYRBY Adam, FELDT Svend, PETRI Stig
Принадлежит:

The application relates to a communication device, e.g. a speakerphone, comprising a microphone signal path, MSP, and a loudspeaker signal path, SSP, the microphone signal path comprising a microphone unit, an MSP-filter, and a transmitter unit operationally connected to each other and configured to transmit a processed signal originating from an input sound picked up by the microphone, the loudspeaker signal path comprising a receiver unit, an SSP-filter, and a loudspeaker unit operationally connected to each other and configured to provide an acoustic sound signal originating from a signal received by the receiver unit. The communication device comprises a control unit for dynamically controlling the filtering characteristics of the MSP and SSP-filters based on one or more control input signals. This has the advantage of providing a simple and flexible scheme for decreasing echo in a communication device, while ensuring an acceptable sound quality in the transmitted signal. 1. A communication device comprising a microphone signal path , termed MSP , and a loudspeaker signal path , termed SSP , the microphone signal path comprising a microphone unit , an MSP-signal processing unit comprising an MSP-filter , and a transmitter unit operationally connected to each other and configured to transmit a processed signal originating from an input sound picked up by the microphone , the loudspeaker signal path comprising a receiver unit , an SSP-signal processing unit comprising an SSP-filter , and a loudspeaker unit operationally connected to each other and configured to provide an acoustic sound signal originating from a signal received by the receiver unit , wherein the filtering characteristics of the MSP-filter and/or the SSP-filter are configurable , and wherein the communication device comprises a control unit for dynamically controlling the configurable filtering characteristics of the MSP and SSP-filters based on one or more control input signals dependent of a ...

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Номер записи: 31
14-01-2021 дата публикации

Double talk detection method, double talk detection apparatus and echo cancellation system

Номер: US20210013927A1
Автор: Guoliang Li, Hongjing GUO, Hu ZHU, Wenkai HAN, Xinshan WANG
Принадлежит: Shenzhen Goodix Technology Co Ltd

A double talk detection method, a double talk detection apparatus and an echo cancellation system are provided. The double talk detection method comprises: determining, according to an energy ratio between a far-end digital voice signal and a near-end digital voice signal, and a frequency coherence value between the near-end digital voice signal and the far-end digital voice signal, whether a near-end speaker's digital voice signal is present in the near-end digital voice signal. The double talk detection method avoids missing detection and false detection, improves the accuracy of double talk detection, cancels the echo in the near-end voice signal thoroughly when applied in the field of echo cancellation, and improves the communication experience of both talk parties.

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Номер записи: 32
21-01-2016 дата публикации

ACOUSTIC ECHO MITIGATION APPARATUS AND METHOD, AUDIO PROCESSING APPARATUS AND VOICE COMMUNICATION TERMINAL

Номер: US20160019909A1
Автор: Dickins Glenn N., SHI DONG, Sun Xuejing
Принадлежит: DOLBY LABORATORIES LICENSING CORPORATION

The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

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Номер записи: 33
21-01-2016 дата публикации

Full duplex wireless communications on devices with limited echo cancellation capabilities

Номер: US20160020894A1
Автор: Eldad Perahia, Robert J. Stacey, Thomas A. Tetzlaff
Принадлежит: Intel IP Corp

Disclosed in some examples are methods, systems, and machine readable mediums which allow for wireless devices with limited echo cancellation capabilities to participate in full-duplex communications. In some examples, by carefully controlling transmission powers and the modulation and coding schemes (MCS) used in the transmissions, both devices can engage in full-duplex communication.

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Номер записи: 34
18-01-2018 дата публикации

LED LIGHT BULB, LAMP FIXTURE WITH SELF-NETWORKING INTERCOM, SYSTEM AND METHOD THEREFORE

Номер: US20180020530A1
Автор: Arno Michael, Carlin John A., Grupp Kevin A., Scordato Stephen
Принадлежит: Athena Patent Development LLC.

A networked light for illumination and intercom for communications in a single housing, with voice command and control, hands-free. The system in a housing configured to conventional looking lamp, bulb, fixture, lighting devices, suitable for a direct replacement of conventional illuminating devices typical found in homes or buildings. A network of such voice command and control systems may be further monitored and controlled from a base station that facilitates programming, communications, and higher functionality therebetween. The system provides speech recognition for powering on and off, dimming, brightening, and adjusting the lighting to preset, night and emergency settings. The voice recognition command controls the intercom to be active and attentive to requests, connecting two or more locations within a home or building structure, for speech exchanges in communications, via radio frequency transmitting and receiving of signal messages between the individual light and intercom system devices within a network of devices. 1. A two-part housing apparatus for hands-free operations of lighting and for intercom communications that are self-networking within a structure , the apparatus comprising:at least one lighting fixture defined by a conventional bulb housing, the at least one lighting fixture illuminating a light, the light controlled with at least one of the following commands: ON, OFF, DIM, BRIGHT, PRESET, NIGHT and EMERGENCY;a transmitter for transmitting radio frequency communications;a receiver for receiving radio frequency communications;a rechargeable battery operational with the at least one lighting fixture;an audible portion indicating a status of the rechargeable battery;a microphone for listening;a speaker for broadcasting;a network identification establishing a type, style, and location of the at least one lighting fixture;a coding system operational with the base station, the coding system establishing identification of a device manufacturing ...

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Номер записи: 35
16-01-2020 дата публикации

CLOUD-BASED ACOUSTIC ECHO CANCELLER

Номер: US20200021329A1
Автор: Schulz Dieter
Принадлежит: Mitel Cloud Services, Inc.

A cloud based echo canceller is set forth for recreating an estimate of a lost packet or data at a server without requiring redundant data over the network or freezing operation of the echo canceller. In an exemplary embodiment, the echo cancelling function is not located in a single device, but is shared between the end-point and a cloud service, where the function of the end-point is to provide a time synchronized copy of the signal from the end-point loudspeaker and the signal received by the end-point microphone. Consequently, the high CPU intensive operations can be offloaded to a server such as a cloud server. In addition, several users can share the echo canceller, thereby reducing the cost of the overall function. According to an additional aspect, a further synchronization block is provided, in the form of a packet estimator, to compensate for packet or data loss in the send direction. 1. A method of compensating for lost packets in a distributed echo canceler , where successive input packets are stored in memory of the distributed echo canceler , comprising:a) detecting a lost signal packet Ro′(n);b) freezing operation of the distributed echo canceler;c) invoking a packet loss compensation (PLC) algorithm for one of either recreating an estimated output packet from previous output packets or halting transmission of output packets;d) estimating the lost signal packet Ro′(n) by performing a correlation of a previously received signal packet with an input signal packet Rin, 'A) upon receipt of a next packet repeating steps b) and c) until effect of the lost signal packet Ro′(n) is flushed out of an echo canceler memory and resuming operation of the distributed echo canceler; or', 'i) if said correlation is poor then'} B) using a relative shift offset of the lost signal packet Ro′(n) to the input signal packet Rin to read an estimated buffer value (Ro″) out of the successive input packets and replacing the signal packet Ro′(n) by the estimated buffer value (Ro ...

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Номер записи: 36
25-01-2018 дата публикации

Leasing terminal of mobile power supply

Номер: US20180025573A1
Автор: Bingsong YUAN, Chingming Luo, Weige Yu
Принадлежит: Shenzhen Laidian Technology Co ltd

A leasing terminal of a mobile power supply comprises a CPU, a network communication module and a main control MCU which are respectively connected to the CPU, and at least one charging module, at least one charging cabinet motion drive motor and at least one mobile power supply information reading module which are respectively connected to the main control MCU; and each charging cabinet corresponds to one charging module, one charging cabinet motion drive motor and one mobile power supply information reading module. By means of the present invention, when a user is outside, a mobile power supply can be leased from a leasing terminal of the mobile power supply in a self-service manner, thereby providing the flexible charging service to the user.

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Номер записи: 37
10-02-2022 дата публикации

Robust Short-Time Fourier Transform Acoustic Echo Cancellation During Audio Playback

Номер: US20220044695A1
Автор: Giacobello Daniele
Принадлежит:

Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each niteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nframe of the model signal from an nframe of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC. 1. A playback device comprising:an audio input interface;an audio stage comprising an audio processor and an audio amplifier;one or more speakers;one or more microphones;at least one processor; and receive, via the audio input interface, one or more audio signals;', 'play back at least one audio signal of the one or more audio signals via the one or more speakers and the audio stage;', 'while playing back the at least one audio signal, capture, via the one or more microphones, audio within an acoustic environment, wherein at least a portion of the captured audio represents sound produced by the one or more speakers in playing back the at least one audio signal via the one or more speakers;', 'receive at least one playback signal from the audio stage representing the at least one audio signal being played back by the one or more speakers and the audio stage;', 'transform into a short time Fourier transform (STFT) domain the captured audio within the acoustic environment to generate a measured signal representing actual acoustic echo;', 'transform into the STFT domain the ...

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Номер записи: 38
25-01-2018 дата публикации

INTERCOM SYSTEM FOR COLLECTIVE HOUSING

Номер: US20180027124A1
Автор: MARUYAMA Norio
Принадлежит:

An intercom system for an apartment house according to the invention includes room units (A, B) which are installed in respective rooms of the apartment house, and a control unit () which is connected to the room units (A, B) of the respective rooms via an intercom line (L) so as to be capable of communicating therewith, and is capable of communicating with an external server (S) via an external communication network. The control unit () acquires, based on a request signal transmitted from the room unit () via the intercom line (L), predetermined information corresponding to the request signal from the external server (S) via the external communication network, and transmits the acquired predetermined information via the intercom line (L) to the room unit () which transmitted the request signal. 1. An intercom system for an apartment house , comprising:room units which are installed in respective rooms of the apartment house; anda control unit which is connected to the room units of the respective rooms via an intercom line so as to be capable of communicating therewith, and is capable of communicating with an external server via an external communication network, whereinthe control unit acquires, based on a request signal transmitted from the room unit via the intercom line, predetermined information corresponding to the request signal from the external server via the external communication network, and transmits the acquired predetermined information via the intercom line to the room unit which transmitted the request signal.2. The intercom system for an apartment house according to claim 1 , whereinthe intercom line includes at least two communication channels, andthe control unit checks a use state of the intercom line when receiving the request signal, and acquires the predetermined information from the external server when the intercom line has a vacant channel.3. The intercom system for an apartment house according to claim 2 , whereinthe control unit checks ...

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Номер записи: 39
25-01-2018 дата публикации

ADAPTIVE FILTER UNIT FOR BEING USED AS AN ECHO CANCELLER

Номер: US20180027125A1
Автор: FELDT Svend, PETRI Stig
Принадлежит: Sennheiser Communications A/S

The invention relates to an adaptive filter unit, in particular for being used as an echo canceller, comprising a first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t), a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t), a processor and a filter output. The processor is configured to calculate and provide audio estimation data X(f, A(t, . . . , t)) in the frequency domain; to calculate a transformed second audio signal Y(f, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal. 1. An adaptive filter unit , in particular for being used as an echo canceller , comprisinga first filter input, configured to receive a first electric audio signal, indicative of a first audio signal A(t);a second filter input, configured to receive a second electric audio signal, indicative of a second audio signal B(t); receive the first and second electric audio signal;', {'sub': n', '1', 'M(fn)', 'n', '1', 'N', 'n, 'calculate and provide audio estimation data X(f, A(t, . . . , t)) in the frequency domain by calculating a FFT transform of the first audio signal A(t), for frequencies f=f, . . . , f, wherein N is a number of FFT bins, and with a number of sampling points M(f) of the first audio signal A(t);'}, {'sub': 'n', 'calculate a transformed second audio signal Y(f, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain;'}, 'calculate a filtered audio signal by ...

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Номер записи: 40
29-01-2015 дата публикации

Voice Protocol Connector Apparatus

Номер: US20150029897A1
Автор: Radhakrishnan Ramesh, Swaminathan Balaji
Принадлежит:

A connection apparatus has a USB port coupled to a USB interface for connection to a computing device enabled to manage IP audio calls, a land-line telephone port coupled to a Four-Wire Analog telephone interface, for connection to a Four-Wire port of a land-line telephone, a wireless interface for coupling to a communication device enabled to manage voice calls over a network, an analog audio port for a headset, coupled to an analog audio interface, a plurality of protocol translation modules, and switching circuitry enabled to connecting individual ones of the ports through individual ones of the protocol translation modules. 1. A connection apparatus , comprising:a USB port coupled to a USB interface for connection to a computing device enabled to manage IP audio calls;a land-line telephone port coupled to a Four-Wire Analog telephone interface, for connection to a Four-Wire port of a land-line telephone;a wireless interface for coupling to a communication device enabled to manage voice calls over a network;an analog audio port for a headset, coupled to an analog audio interface;a plurality of protocol translation modules; andswitching circuitry enabled to connecting individual ones of the ports through individual ones of the protocol translation modules.2. The connection apparatus of wherein the plurality of protocol translation modules comprise a module including software enabled to translate audio signals between four-wire analog audio protocol and analog audio protocol.3. The connection apparatus of wherein the plurality of protocol translation modules comprise a module including software enabled to translate audio signals between Bluetooth™ audio and analog audio protocol.4. The connection apparatus of wherein the plurality of protocol translation modules comprise a module including software enabled to translate audio signals between USB audio class protocol and analog audio protocol.5. The connection apparatus of wherein the plurality of protocol ...

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Номер записи: 41
23-01-2020 дата публикации

Pre-distortion system for cancellation of nonlinear distortion in mobile devices

Номер: US20200028970A1
Автор: Dimitrios Dimitriadis, Donald J. Bowen, Horst J. Schroeter, Lusheng Ji
Принадлежит: AT&T INTELLECTUAL PROPERTY I LP

A pre-distortion system for improved mobile device communications via cancellation of nonlinear distortion is disclosed. The pre-distortion system may transmit an acoustic signal from a network to a device, wherein the acoustic signal includes a linear signal and a nonlinear cancellation signal that cancels at least a portion of nonlinear distortions created once a loudspeaker in the device emits the linear signal. Thus, when a loudspeaker of a mobile device is operating and nonlinear distortions are generated by the loudspeaker or adjacent components of the mobile device in close proximity to the loudspeaker, the pre-distortion system may create one or more nonlinear cancellation signals in the network. The nonlinear cancellation signal may be combined with the linear signal sent to the loudspeaker to cancel the nonlinear distortion signal created by the loudspeaker emitting acoustic sounds from the linear signal. Thus, the nonlinear cancellation signal becomes a pre-distortion signal.

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Номер записи: 42
28-01-2021 дата публикации

PORTABLE KEYPANEL FOR INTERCOM SYSTEM

Номер: US20210029237A1
Автор: Andersen Jim, Borjesson Fredrik, Brchan Jason, Friesen La Ruhe Gene, Jenkins Keith, Moore Kevin B., Strader Ralph K.
Принадлежит:

A mobile intercom device is described that is configured to facilitate the assignment and reassignment of functions during use. A plurality of button pairs, each including a talk button and a listen button, and a display screen are provided on the mobile intercom device. An electronic processor of the mobile intercom device is configured to display a list of alphanumeric identifiers on the display screen, each alphanumeric identifier in the list of alphanumeric identifiers corresponding to a different one of a plurality of channels on the intercom system. A selection of a first alphanumeric identifier from the list is received and a channel on the intercom system corresponding to the first alphanumeric identifier is assigned to a first button pair of the plurality of button pairs. 1. A mobile intercom device comprising:a plurality of button pairs, each button pair including a talk button and a listen button;a display screen; and display a list of alphanumeric identifiers on the display screen, each alphanumeric identifier in the list of alphanumeric identifiers corresponding to a different one of a plurality of channels on an intercom system,', 'receive a selection of a first alphanumeric identifier from the list of alphanumeric identifiers displayed on the display screen,', 'assign a channel on the intercom system corresponding to the first alphanumeric identifier to a first button pair of the plurality of button pairs in response to receiving the selection of the first alphanumeric identifier from the list, and', 'transmit audio stream data from the mobile intercom device to the channel on the intercom system corresponding to the first alphanumeric identifier when the talk button of the first button pair is activated., 'an electronic processor configured to'}2. The mobile intercom device of claim 1 , wherein the electronic processor is further configured toreceive a second selection of a second alphanumeric identifier from the list of alphanumeric identifiers ...

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Номер записи: 43
31-01-2019 дата публикации

COMPUTER-PROGRAMMED TELEPHONE-ENABLED DEVICES FOR PROCESSING AND MANAGING NUMEROUS SIMULTANEOUS VOICE CONVERSATIONS CONDUCTED BY AN INDIVIDUAL OVER A COMPUTER NETWORK AND COMPUTER METHODS OF IMPLEMENTING THEREOF

Номер: US20190037079A1
Автор: Detterman Eric, Tassone Anthony
Принадлежит:

In some embodiments, the present invention provides for a computer-implemented method, including: causing, by a specifically programmed computer call management communication system, to transform, over a computer network, computing devices of users, into corresponding specialized call management devices, by having each computing device to execute a specialized call management client software application being in electronic communication with the specifically programmed computer call management communication system over the computer network by utilizing SIP; where the specialized call management client software application generates specialized graphical user interfaces configured to allow each user to concurrently initiate and maintain, over the computer network, a plurality of voice communications of distinct types with other users, by, for example, allowing each user to independently and dynamically divert, in real-time, any voice communication of any type to any audio device associated with a corresponding specialized call management device of such user. 1causing, by a specifically programmed computer call management communication system, to transform, over a computer network, a plurality of computing devices of a plurality of users, into a corresponding plurality of specialized call management devices, by having each computing device to execute a specialized call management client software application being in electronic communication with the specifically programmed computer call management communication system over the computer network;wherein the specialized call management client software application, upon the execution, generates a plurality of specialized graphical user interfaces configured to allow each user of the plurality of users to concurrently initiate and maintain, over the computer network, a plurality of voice communications of distinct types with other users based, at least in part, on:maintaining each voice communication independent from ...

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Номер записи: 44
31-01-2019 дата публикации

Recording Method, Recording Play Method, Apparatuses, and Terminals

Номер: US20190037308A1
Автор: HAN Bo, Liu Jie, Liu Li, Wang Zhihua, Yang Wei, ZHOU Xiong
Принадлежит:

A recording method, a recording play method, apparatuses and terminals, where the recording method comprises obtaining, by a terminal, recording data in all sound source directions input by at least three microphones, and generating, by the terminal, a recording file according to the obtained recording data, wherein the recording file saves the recording data in all the sound source directions. 125.-. (canceled)26. A recording method , comprising:obtaining, by a terminal, recording data in all sound source directions from at least three microphones;generating, by the terminal, a recording file according to the obtained recording data; andsaving, by the terminal in the recording file, the recording data in all the sound source directions.27. The recording method of claim 26 , further comprising:obtaining, by the terminal, a recording direction from a user, the recording direction comprising one of at least two sound source directions, and the at least two sound source directions being obtained by the at least three microphones at a same time point; andsaving, by the terminal, the recording direction in the recording file.28. The recording method of claim 27 , wherein the recording direction is obtained according to a sound source adjustment gesture from the user claim 27 , the sound source adjustment gesture adjusting the recording direction.29. The recording meal od of claim 28 , wherein the sound source adjustment gesture comprises at least one of a gesture of touching and tapping claim 28 , a gesture of sliding claim 28 , or an air gesture.30. The recording method of claim 27 , wherein the recording direction comprises the at least two different recording directions set separately by the user at different time points.31. The recording method of claim 28 , wherein the recording direction comprises the at least two different recording directions set separately by the user at different time points.32. A terminal claim 28 , comprising:at least three microphones ...

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Номер записи: 45
04-02-2021 дата публикации

METHOD AND DEVICE OF SUSTAINABLY UPDATING COEFFICIENT VECTOR OF FINITE IMPULSE RESPONSE FILTER

Номер: US20210035593A1
Автор: Liang Min
Принадлежит:

A method and a device of sustainably updating a coefficient vector of a finite impulse response FIRfilter. The method includes obtaining () a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal; updating () the coefficient vector of the FIR filter according to the time-varying regularization factor. 172.-. (canceled)73. A sustainable adaptive updating method of a coefficient vector of a Finite Impulse Response (FIR) filter , comprising:obtaining a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal;updating the coefficient vector of the FIR filter according to the time-varying regularization factor.74. The sustainable adaptive updating method according to claim 73 , wherein the preset signal comprises one of combined pairs of following:a far-end reference speech signal inputted in an Acoustic Echo Canceller (AEC) and a near-end speech signal received by a microphone;a noise reference signal and a system input signal in an adaptive noise cancellation system;an interference reference signal and a system input signal in an adaptive interference cancellation system; andan excitation input signal and an unknown system output signal to be identified in adaptive system identification.75. The sustainable adaptive updating method according to claim 73 , wherein the preset signal comprises a far-end reference speech signal inputted in an Acoustic Echo Canceller (AEC) and a near-end speech signal received by a microphone;obtaining the time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing the preset signal, comprises:obtaining a power of a signal received ...

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Номер записи: 46
12-02-2015 дата публикации

ECHO CANCELLER FOR VOIP NETWORKS

Номер: US20150043361A1
Автор: Ho Dominic, Rabipour Rafi
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

An echo canceller in an IP network includes an adaptive filter that models the echo path between a receiving output port of the echo canceller and a sending input port. The adaptive filter filters a receiving input signal to generate an estimate of an echo signal. The estimate of the echo signal is subtracted from a sending input signal to cancel the echo in the sending input signal and to generate a sending output signal. A packet loss detection circuit detects when packet loss occurs in the echo path. Responsive to detection of packet loss in the echo path, the echo canceller applies packet loss concealment to either the sending output signal or the receiving input signal.

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Номер записи: 47
12-02-2015 дата публикации

ECHO CANCELLER FOR VOIP NETWORKS

Номер: US20150043571A1
Автор: Ho Dominic, Rabipour Rafi
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

An echo canceller for an IP network includes an adaptive filter that models the echo path and generates an estimate of the echo signal from a receiving input signal. The echo canceller subtracts the estimate of the echo signal from a sending input signal to generate a sending output signal with reduced echo. Variation in the echo delay is detected. A delay circuit compensates for the changes in the echo delay to provide proper time-alignment between the estimate of the echo signal and the sending input signal so that the echo signal will be more effectively cancelled. 1. A method of echo cancellation to handle variation of an echo delay , said method comprising:generating, from a receiving input signal received on a first input port of an echo canceller, a first estimate of an echo signal using an adaptive filter that models an echo path between a first output port and a second input port of the echo canceller;computing a first estimate of the echo delay by correlating the first estimate of the echo signal with a sending input signal received on said second input port;time-aligning the first estimate of the echo signal with the sending input signal based on the first estimate of the echo delay; andsubtracting the time-aligned first estimate of the echo signal from the sending input signal to generate a first sending output signal with reduced echo for output over a second output port of the echo canceller.2. The method of wherein computing the first estimate of the echo delay by correlating the estimate of the echo signal with the sending input signal comprises:correlating the first estimate of the echo signal with a sending input signal received on said second input port to generate a correlation signal; andcomputing the first estimate of the echo delay by locating a peak in the correlation signal.3. The method of further comprising:detecting variation in the echo delay;wherein computing the first estimate of the echo delay comprises computing the estimate of the ...

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Номер записи: 48
09-02-2017 дата публикации

SOUND EMISSION AND COLLECTION DEVICE, AND SOUND EMISSION AND COLLECTION METHOD

Номер: US20170041445A1
Автор: UKAI Satoshi, Yamakawa Takashi
Принадлежит:

A sound emission and collection device includes a speaker, a filter processing a sound emission signal, microphones, echo cancellers cancelling regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones, a first integration section integrating adaptive filter coefficients taken out from the plurality of echo cancellers, a reverberation time estimation section estimating the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient, and an arithmetic operation section specifying a frequency band having a long reverberation time from the sound emission signal based on the estimated reverberation time, calculating a filter coefficient for suppressing power of the specified frequency band, and setting the filter coefficient to the filter. 1. A sound emission and collection device comprising:a speaker;a filter configured to process a sound emission signal serving as a sound signal to be supplied to the speaker;a plurality of microphones;a plurality of echo cancellers provided so as to respectively correspond to the plurality of microphones and configured to cancel regression sound signals of the sound emitted by the speaker from the sound collection signals of the corresponding microphones;a first integration section configured to integrate adaptive filter coefficients taken out from the plurality of echo cancellers;a reverberation time estimation section configured to estimate the reverberation time for each frequency band in the space in which the speaker and the plurality of microphones are present on the basis of the integrated adaptive filter coefficient; andan arithmetic operation section configured to specify a frequency band having a long reverberation time from the sound emission signal on the basis of the estimated reverberation time, to calculate a filter coefficient for ...

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Номер записи: 49
06-02-2020 дата публикации

MULTI-CHANNEL ACOUSTIC ECHO CANCELLATION

Номер: US20200043460A1
Автор: Kadri Romi, Sereshki Saeed Bagheri
Принадлежит:

A playback device is configured to receive, via a network interface, a source stream of audio including first and second channel streams of audio, and to produce, via respective first and second speaker drivers, a first channel audio output and a second channel audio output. The playback device is also configured to receive, via one or more microphones, a captured stream of audio including first and second portions corresponding to the respective first and second channel audio outputs. The playback device is also configured to combine at least the first channel stream of audio and the second channel stream of audio into a compound audio signal and perform acoustic echo cancellation on the compound audio signal and thereby produce an acoustic echo cancellation output, then to apply the acoustic echo cancellation output to the captured stream of audio and thereby increase a signal-to noise ratio of the captured stream of audio. 1. A playback device comprising:a first speaker driver;at least a second speaker driver;at least one processor;a network interface;a non-transitory computer-readable medium; and receiving, via the network interface, a source stream of audio comprising source audio content to be played back by the playback device, wherein the source audio content comprises a first channel stream of audio and a second channel stream of audio;', 'producing a first channel audio output by playing back, via the first speaker driver, the first channel stream of audio;', 'producing a second channel audio output by playing back, via the second speaker driver, the second channel stream of audio;', 'receiving, via one or more microphones, a captured stream of audio comprising (i) a first portion corresponding to the first channel audio output and (ii) a second portion corresponding to the second channel audio output, wherein the captured stream of audio has a first signal-to-noise ratio;', 'combining at least the first channel stream of audio and the second channel ...

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Номер записи: 50
06-02-2020 дата публикации

Robust Short-Time Fourier Transform Acoustic Echo Cancellation During Audio Playback

Номер: US20200043507A1
Автор: Giacobello Daniele
Принадлежит:

Example techniques involve noise-robust acoustic echo cancellation. An example implementation may involve causing one or more speakers of the playback device to play back audio content and while the audio content is playing back, capturing, via the one or more microphones, audio within an acoustic environment that includes the audio playback. The example implementation may involve determining measured and reference signals in the STFT domain. During each niteration of an acoustic echo canceller (AEC): the implementation may involve determining a frame of an output signal by generating a frame of a model signal by passing a frame of the reference signal through an instance of an adaptive filter and then redacting the nframe of the model signal from an nframe of the measured signal. The implementation may further involve determining an instance of the adaptive filter for a next iteration of the AEC. 1. A system comprising:an audio stage comprising an audio processor and an audio amplifier;one or more speakers;one or more microphones;one or more processors;data storage storing instructions executable by the one or more processors that cause the system to perform functions comprising:while audio content is playing back via the one or more speakers, capturing, via the one or more microphones, audio within an acoustic environment, wherein the captured audio comprises audio signals representing sound produced by the one or more speakers in playing back the audio content;receiving a playback signal from the audio stage representing the audio content being played back by the one or more speakers;transforming into a short time Fourier transform (STFT) domain the captured audio within the acoustic environment to generate a measured signal representing actual acoustic echo;transforming into the STFT domain the received playback signal from the audio stage to generate a reference signal;{'sup': 'th', 'claim-text': [{'sup': th', 'th, 'claim-text': [{'sup': th', 'th', 'th, ' ...

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Номер записи: 51
07-02-2019 дата публикации

EFFICIENT REUTILIZATION OF ACOUSTIC ECHO CANCELER CHANNELS

Номер: US20190045064A1
Автор: Argyropoulos Paraskevas, Bou Daher Elie, Hera Cristian M., Kathavarayan Vigneish, VAUTIN Jeffery R.
Принадлежит:

Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients. 1. A method of reducing echo content of an audio signal , comprising:receiving an audio program content signal;receiving a sound stage rendering parameter;selecting a set of echo filter coefficients, from among a plurality of stored sets of echo filter coefficients, based upon the sound stage rendering parameter;filtering the audio program content signal, using the selected set of filter coefficients, to generate an estimated echo signal;receiving a microphone signal, configured to include a signal component representative of an echo of the audio program content signal; andsubtracting the estimated echo signal from the microphone signal to generate an output audio signal.2. The method of further comprising loading the selected set of echo filter coefficients to an audio filter and activating the audio filter to perform the filtering.3. The method of further comprising rendering the audio program content signal into an acoustic signal claim 1 , based upon the selected sound stage rendering parameter.4. The method of further comprising loading the selected set of echo filter coefficients to an adaptive filter claim 1 , adapting the adaptive filter coefficients claim 1 , and copying the adaptive filter coefficients to an active audio filter that performs the filtering.5. The method of further comprising loading the selected set of echo filter coefficients to an adaptive filter claim 1 , the adaptive filter performing the ...

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Номер записи: 52
07-02-2019 дата публикации

MULTI-CHANNEL RESIDUAL ECHO SUPPRESSION

Номер: US20190045065A1
Автор: Bou Daher Elie, Hera Cristian M., Jain Ankita D.
Принадлежит:

Audio systems and methods for suppressing residual echo are provided. First and second audio program content signals are received, and a residual signal from an echo canceler is received. A first spectral mismatch is determined based at least upon a cross power spectral density of the first program content signal and the residual signal. A second spectral mismatch is determined based at least upon a cross power spectral density of the second program content signal and the residual signal. The residual signal is filtered to reduce residual echo, based at least upon the first spectral mismatch, the second spectral mismatch, a spectral density of the first program content signal, a spectral density of the second program content signal, and a spectral density of the residual signal. 1. A method of suppressing residual echo , comprising:receiving a residual signal from an echo cancelation subsystem;receiving a first program content signal;determining a first spectral mismatch based at least upon a cross power spectral density of the first program content signal and the residual signal;receiving a second program content signal;determining a second spectral mismatch based at least upon a cross power spectral density of the second program content signal and the residual signal; andcontrolling a filter to filter the residual signal based upon the first spectral mismatch, the second spectral mismatch, a spectral density of the first program content signal, a spectral density of the second program content signal, and a spectral density of the residual signal.2. The method of wherein controlling the filter includes calculating filter coefficients and providing the filter coefficients to the filter.3. The method of wherein controlling the filter to filter the residual signal is based upon a previously determined first spectral mismatch and a previously determined second spectral mismatch during a period of time when a double-talk condition is detected.4. The method of further ...

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Номер записи: 53
07-02-2019 дата публикации

MITIGATING IMPACT OF DOUBLE TALK FOR RESIDUAL ECHO SUPPRESSORS

Номер: US20190045066A1
Автор: Bou Daher Elie, Hera Cristian M.
Принадлежит:

Audio systems and methods of suppressing residual echo are provided that determine spectral mismatches by comparing a spectral density of a residual signal from an acoustic echo canceler to a spectral density of a program content signal. At least one spectral mismatch is stored in memory. The systems and methods select a spectral mismatch to use for calculating a filter coefficient, from among one or more of the stored or actively determined spectral mismatches, and filter the residual signal based upon the calculated filter coefficient. 1. A method of suppressing residual echo , comprising:determining a first spectral mismatch of an acoustic echo canceler based upon a program content signal and a residual signal;storing the first spectral mismatch in a memory;determining a second spectral mismatch of the acoustic echo canceler based upon the program content signal and the residual signal at a different time than the first spectral mismatch;selecting one of the first spectral mismatch or the second spectral mismatch;calculating a filter coefficient based upon the selected spectral mismatch; andfiltering the residual signal based upon the calculated filter coefficient.2. The method of wherein selecting one of the first spectral mismatch or the second spectral mismatch is based at least in part upon detecting a double-talk condition.3. The method of wherein selecting one of the first spectral mismatch or the second spectral mismatch is based at least in part upon a sound stage configuration selected for rendering the program content signal into an acoustic signal.4. The method of further comprising storing additional spectral mismatches in the memory to provide a plurality of stored spectral mismatches.5. The method of further comprising detecting a double talk condition and wherein selecting one of the first spectral mismatch or the second spectral mismatch includes selecting one of the plurality of stored spectral mismatches based on an amount of time for the double ...

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Номер записи: 54
19-02-2015 дата публикации

Acoustic Echo Cancellation for Audio System with Bring Your Own Devices (BYOD)

Номер: US20150050967A1
Автор: BAO Feng, Kunapuli Subrahmanyam Venkata, Sundsbarm Tor A., Yang Fei
Принадлежит: Cisco Technology, Inc

A controller for the conference session receives at least one audio signal to generate a speaker signal. The controller correlates the speaker signal with network timing information and generates speaker timing information. The controller transmits the correlated speaker signal and timing information to a mobile device participating in the conference session. The mobile device generates an echo cancelled microphone signal from a microphone of the mobile device, and transmits the echo cancelled signal back to the controller. The controller also receives array microphone signals associated with an array of microphones at known positions in the room. The controller removes acoustic echo from the array microphone signals, and estimates a relative location of the mobile device. The controller dynamically selects as audio output corresponding to the mobile device location either (a) the array microphone signal, or (b) the echo cancelled microphone signal from the mobile device. 1. A method comprising:receiving at least one audio signal at a controller associated with a conference session;receiving, at the controller, a plurality of array microphone signals associated with an array of microphones at corresponding known positions in a room of the conference session;generating a speaker signal from the at least one audio signal and the plurality of array microphone signals;correlating, at the controller, the speaker signal with network timing information to generate speaker timing information;transmitting the speaker signal with the speaker timing information via a network to a mobile device that is participating in the conference session to enable the mobile device to generate an echo cancelled microphone signal from a microphone of the mobile device;receiving, at the controller, the echo cancelled remote microphone signal;removing an acoustic echo from the plurality of array microphone signals to generate a plurality of echo cancelled array microphone signals;estimating a ...

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Номер записи: 55
18-02-2021 дата публикации

MUTED COMPONENT DETECTION

Номер: US20210051233A1
Автор: JR. Robert James, Kapinos Robert James, Li Scott Wentao, Norton, VanBlon Russell Speight
Принадлежит:

One embodiment provides a method, comprising: transmitting, from a communication component, a signal down a communication channel; determining, using a processor, whether an echo associated with the signal is detected by the communication component; and providing, responsive to determining that the echo is not detected, a notification to a user that a mute control is enabled at another communication component along the communication channel. Other aspects are described and claimed. 1. A method , comprising:transmitting, from a communication component, a signal down a communication channel, wherein the signal is shaped to comprise a predetermined waveform;determining, using a processor, whether the signal is detected again by the communication component; andproviding, responsive to determining that the signal is not detected, a notification to a user that a mute control is enabled at another communication component along the communication channel, wherein the notification comprises an identification of the another communication component at which the mute control is enabled.2. The method of claim 1 , wherein the communication component is a component selected from the group consisting of a headset microphone claim 1 , a VoIP client claim 1 , and a VoIP server.3. (canceled)4. The method of claim 1 , wherein the predetermined waveform is able to circumvent at least one noise cancellation mechanism present in the communication channel.5. The method of claim 1 , wherein the notification comprises identification of the another communication component at which the mute control is enabled.6. The method of claim 5 , further comprising dynamically deactivating the mute control on the identified another communication component.7. The method of claim 6 , further comprising aligning the mute control with a global operating system.8. The method of claim 1 , wherein the providing the notification comprises providing an audible notification through a headset.9. The method of claim ...

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Номер записи: 56
15-02-2018 дата публикации

SINGLE-CHANNEL, BINAURAL AND MULTI-CHANNEL DEREVERBERATION

Номер: US20180047378A1
Автор: Kokkinis Elias, Tsilfidis Alexandros
Принадлежит:

A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented. 118.-. (canceled)19. A method , in a multimedia signal processing system , that allows a user to adjust suppression of reverberation or noise from a digital signal that represents a sound signal , comprising:analyzing the digital signal by determining a plurality of time-frequency frames of the digital signal;deriving a first estimation of reverberation or noise from one of the frames in a first time instant;deriving a first suppression gain from the first estimation;selecting, by the user, a first exponent from a predetermined set of exponents;deriving a modified first suppression gain based on the first suppression gain raised to a power related to the first exponent;wherein the modified first suppression gain is applied to the one frame in the first time instant;deriving a second estimation of reverberation or noise from one of the frames in a second time instant;deriving a second suppression gain from the second estimation;selecting, by the user, a second exponent from a predetermined set of exponents;deriving a modified second suppression gain based on the second suppression gain raised to a power related to the second exponent;wherein the modified second suppression gain is applied to the one frame in the second time instant;outputting a signal that involves processing the frame in the first time instant utilizing the first modified suppression gain;and outputting a signal that involves processing the frame in the second time instant utilizing the second modified suppression gain;wherein the first exponent and the second exponent are different from one another, andwherein the second time instant is subsequent to the first time instant.20. The method of claim 19 , ...

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Номер записи: 57
15-02-2018 дата публикации

SYSTEM AND METHOD FOR ADDRESSING ACOUSTIC SIGNAL REVERBERATION

Номер: US20180047408A1
Автор: Tebelmann Lars, Wolff Tobias
Принадлежит:

A method, computer program product, and computer system for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal. 1. A computer-implemented method comprising:receiving, at one or more microphones, a first audio signal;receiving, at the one or more microphones, a reverberation audio signal;processing at least one of the first audio signal and the reverberation audio signal; andlimiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal.2. The computer-implemented method of claim 1 , further comprising:receiving the one or more outputs from the model based reverberation equalizer at a postfilter.3. The computer-implemented method of claim 1 , further comprising:receiving a beamformer output at a postfilter.4. The computer-implemented method of claim 1 , further comprising:adjusting the model based reverberation equalizer to obtain a particular direct-to-noise ratio.5. The computer-implemented method of claim 4 , further comprising:measuring the direct-to-noise ratio using, at least in part, at least one temporal criteria.6. The computer-implemented method of claim 4 , further comprising:using the model based reverberation equalizer for the particular direct-to- noise ratio as a constraint ...

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Номер записи: 58
06-02-2020 дата публикации

NETWORK COMMUNICATION SYSTEM

Номер: US20200045156A1
Автор: Riedel Thomas, SCHAAF Stephan, SIEBEN Hubert
Принадлежит:

A network communication station has a housing with a housing front and a housing rear side, a plurality of operating elements on the housing front, and a touchscreen display movable on the housing front between a first position and a second position physically offset therefrom for, in a first function state of the display, displaying information of a first kind and/or allowing data of a first kind to be entered on the display and for, in a second function state of the display, displaying information of a second kind and/or allowing data of a second kind to be entered on the display. A controller sets the display in the first function state when in the first position and in the second function state when in the second position. 1. In a network , a communication station comprising:a housing with a housing front and a housing rear side,a plurality of operating elements on the housing front; anda touchscreen display on the housing front, the display displaying, in a first function state of the display, information of a first kind and/or allowing data of a first kind to be entered on the display, whereinthe display can be moved relative to the housing between a rest position in which the display is in the first function state and a switching position, andas a result of moving the display relative to the housing into the switching position, the display enters into a second function state, in which information of a second kind can be displayed and/or data of a second kind can be entered on the display.2. The communication station according to claim 1 , wherein the housing is shaped as a parallelepiped or essentially shaped as a parallelepiped.3. The communication station according to claim 1 , wherein at least one operating element is designed to be programmable.4. The communication station according to claim 1 , wherein the communication station comprises a plurality of touchscreen displays that can each be shifted between the rest position and the switching position.5. ...

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Номер записи: 59
18-02-2021 дата публикации

Message delivery device, method, and recording medium

Номер: US20210051450A1
Автор: Tadayuki Oono
Принадлежит: NEC Corp

In order to make it possible to suppress increase in the cost of message retransmission, this message delivery device delivers a message, stores a history relating to the success or failure of the delivery to each of destinations of the message in a history storage unit, when the delivery of the message has failed, determines on the basis of the history whether to perform retransmission of the message, and when determining to perform the retransmission, performs the retransmission of the message.

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Номер записи: 60
08-05-2014 дата публикации

Audiovisual apparatus for reducing echo

Номер: US20140125756A1
Автор: Chi-Hsing Hsu, Ming-Shun Chen
Принадлежит: AzureWave Technologies Inc

The present disclosure relates to an audiovisual apparatus, which includes an audiovisual capturing unit, an audiovisual broadcast unit, and a transmission cable module. The two ends of the transmission cable module are respectively connected to the audiovisual capturing unit and the audiovisual broadcast unit. The transmission cable module has a video cable group for transmitting video signals and an audio cable group for transmitting audio signals suitable for reducing echo. The audio cable group includes a cable for providing a digital clock for the audio signal, a cable for providing a clock for the left right channel switching audio signal, a cable for inputting serial audio signal, and a cable for outputting serial audio signal. With this arrangement, the audio signals captured by the audiovisual capturing unit and transmitted by the transmission cable module to the audiovisual broadcast unit can be effectively removed of echo.

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Номер записи: 61
14-02-2019 дата публикации

Message Recording System for Broadcast Intercoms

Номер: US20190052743A1
Автор: ANDERSON Shawn, McKinnon Sean, Mueggenberg Timothy, Puttick James
Принадлежит:

An intercom device and a method of recording an audio stream at the intercom device. The intercom device includes a user interface, a memory, and an electronic processor. The electronic processor is communicatively coupled to the user interface and the memory and is configured to receive a user-selectable entry from an input mechanism of the user interface. The electronic processor sets an away mode of the intercom device in response to the user-selectable entry. An audio stream is received by the intercom device from another intercom device. Are cord signal indicative of a request to record the audio stream is also received from the another intercom device. The intercom device records the audio stream in the memory in response to receiving the record signal when operating in the away mode. 1. An intercom device for recording an audio stream , the intercom device comprising:a user interface including an input mechanism;a memory; and receive a user-selectable entry from the input mechanism;', 'set an away mode of the intercom device in response to the user-selectable entry;', 'receive an audio stream from another intercom device;', 'receive a record signal indicative of a request to record the audio stream from the another intercom device; and', 'record the audio stream in the memory in response to receiving the record signal when operating in the away mode., 'an electronic processor communicatively coupled to the user interface and the memory, the electronic processor configured to'}2. The intercom device according to claim 1 , wherein the electronic processor is further configured to receive a call signal from the another intercom device prior to receiving the audio stream claim 1 , the call signal initiating communication with the intercom device.3. The intercom device according to claim 2 , wherein the electronic processor is further configured to send an away signal indicative of the away mode to the another intercom device when the away mode is set and the call ...

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Номер записи: 62
25-02-2021 дата публикации

ARRAY MICROPHONE AND SOUND COLLECTION METHOD

Номер: US20210058700A1
Автор: IKEGAYA Yuji, Yamakawa Takashi
Принадлежит:

A sound collection method and a microphone array estimate at least one sound source direction and form a plurality of sound collection beams in the estimated plurality of sound source direction, using sound collection signals of a plurality of microphones. The number of sound source directions estimated is smaller than the number of sound collection beams formed. 1. A microphone array comprising:a plurality of microphones;an estimator that estimates at least one sound source direction;a beam former that forms, using sound collection signals from the plurality of microphones, a plurality of sound collection beams larger in number than the number of the estimated at least one sound source direction but no more than a predetermined maximum number;a memory storing information indicating a beam direction of each of the plurality of sound collection beams;a determiner that determines whether or not a number of the plurality of sound collection beams reaches the predetermined maximum number; andan updater that updates at least one of the stored beam directions to the estimated at least one sound source direction upon the number of the plurality of sound collection beams being determined to reach the predetermined maximum number.2. The microphone array according to claim 1 , further comprising a mixing processor that mixes an audio signal corresponding to one sound collection beam claim 1 , among the plurality of sound collection beams claim 1 , by a gain according to volume of the one sound collection beam.3. The microphone array according to claim 1 , wherein the updater updates the direction of an earliest updated sound collection beam among the plurality of sound collection beams.4. The microphone array according to claim 1 , wherein the plurality of microphones are configured as a ceiling tile.5. The microphone array according to claim 1 , wherein the updater updates the direction of a sound collection beam with the direction thereof closest to the estimated at least ...

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Номер записи: 63
15-05-2014 дата публикации

METHOD AND APPARATUS FOR ACOUSTIC ECHO CANCELLATION IN VOIP TERMINAL

Номер: US20140133648A1
Автор: Ciarkowski Andrzej, Czyzewski Andrzej
Принадлежит:

A method of acoustic echo cancellation in the VoIP terminal using processing of the far-end signal with the digital adaptive filter in order to obtain the echo estimate that is subtracted from the microphone signal in which the far-end signal, before is is converted to the analog from and passed to the loudspeaker (), is marked by embedding an encoded digital signature obtained from the signature generator () and then detection of the digital signature is performed in the signal collected by the microphone () and converted to digital form, depending on the result of the digital signature detection, adaptation of the digital adaptive filter () is resumed or stopped. A circuit for acoustic echo cancellation in VoIP terminal contains the digital adaptive filter with the control block situated between the far-end speech signal path and the near-end speech signal path, and the double-talk detector () that comprises the signature generator () connected by the signature encoder (). 141479. A method of acoustic echo cancellation in a VoIP terminal in which a far-end speech signal is processed with a digital adaptive filter in order to obtain a echo estimate that is subtracted from a microphone signal and the result is used for adaptation of the digital adaptive filter , said adaptation is stopped while a double-talk is present characterized in that the far-end speech signal , before it is converted to an analog form and passed to a loudspeaker () , is marked by adding an encoded digital signature obtained from a signature generator () and then detection of the digital signature is performed in a signal collected by a microphone () and converted to a digital form and depending on detection of presence or absence of the digital signature in the signal , adaptation of a digital adaptive filter () is resumed or stopped , respectively.26547. A method as in characterized in that the digital signature is a sequence of bytes chosen so that the digital signature is suppressed by a ...

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Номер записи: 64
13-02-2020 дата публикации

METHOD FOR IMPROVING ECHO CANCELLATION EFFECT AND SYSTEM THEREOF

Номер: US20200053224A1
Автор: Zhang Henglizi
Принадлежит:

A method for improving an echo cancellation effect and a system thereof are disclosed. The method comprises includes: performing a non-linear compensation for a non-linear response portion of an excitation signal using an NLC algorithm to obtain a compensated excitation signal; outputting the compensated excitation signal to an echo cancellation system; and performing echo cancellation for the compensated excitation signal by the echo cancellation system. According to the present disclosure, using the NLC algorithm, non-linear compensation is performed for the non-linear portion of the excitation signal, non-linear outputs generated due to non-linear characteristics of the system are pre-compensated when being input to the echo cancellation system, such that the echo signal output by the echo cancellation system is minimized and the echo cancellation effect is improved. 1. A method for improving echo cancellation effect , which comprising the following steps of:Step S1, performing a non-linear compensation for a non-linear response portion of an excitation signal using an NLC algorithm to obtain a compensated excitation signal;Step S2, outputting the compensated excitation signal to the an echo cancellation system; andStep S3, performing echo cancellation for the compensated excitation signal by the echo cancellation system.2. The method according to claim 1 , wherein the compensated excitation signal comprises a linear response portion and a background noise portion of the excitation signal.3. The method according to claim 2 , wherein the non-linear response portion of the excitation signal is converted to at least one of a linear response portion or a smaller non-linear response portion upon the non-linear compensation.4. An echo cancellation system claim 2 , comprising:a non-linear compensation module; andan echo cancellation device;wherein the non-linear compensation module is configured to perform a non-linear compensation for an excitation signal using an NLC ...

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Номер записи: 65
15-05-2014 дата публикации

ECHO CANCELLATION FOR ULTRASOUND

Номер: US20140135077A1
Автор: Dagan Noam, Leviant Michael, Nelson Hui-ya Liao
Принадлежит: QUALCOMM INCORPORATED

A method includes accessing signal data descriptive of a transmission sequence and pre-determined values associated with the transmission sequence. The signal data and the pre-determined values may be stored in a memory. The method includes transmitting a signal from a speaker of an electronic device according to the transmission sequence. The method includes generating a frame based on one or more signals received at a microphone of the electronic device. The one or more signals include an echo signal associated with the transmitted signal. The method includes processing the frame using the pre-determined values to produce an output frame in which a contribution associated with the echo signal is reduced. 1. A method comprising:accessing signal data descriptive of a transmission sequence and pre-determined values associated with the transmission sequence, wherein the signal data and the pre-determined values are stored in a memory;transmitting a signal from a speaker of an electronic device according to the transmission sequence;generating a frame based on one or more signals received at a microphone of the electronic device, the one or more signals including an echo signal associated with the transmitted signal; andprocessing the frame using the pre-determined values to produce an output frame in which a contribution associated with the echo signal is reduced.2. The method of claim 1 , wherein the pre-determined values correspond to a fast Fourier transform (FFT) of the transmission sequence.3. The method of claim 2 , wherein processing the frame further comprises:performing an FFT on the frame to produce a first processed frame; andproviding the first processed frame to echo cancellation logic, wherein the echo cancellation logic is configured to process the first processed frame based on the pre-determined values.4. The method of claim 3 , wherein processing the first processed frame based on the pre-determined values comprises:multiplying the first processed ...

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Номер записи: 66
15-05-2014 дата публикации

Dynamic Speaker Management with Echo Cancellation

Номер: US20140135078A1
Автор: DOY Anthony S., Nigam Vivek P., Polleros Robert
Принадлежит: MAXIM INTEGRATED PRODUCTS, INC.

A system for echo cancellation includes a dynamic speaker management (DSM) module, a current/voltage sensing amplifier, a sound pressure level (SPL) model module and an echo canceller. The example DSM module is receptive to a far-end signal and is operative to develop a modified far-end signal and a plurality of parameter outputs. The example current/voltage sensing amplifier is coupled to the modified far-end signal and develops an amplifier output, a voltage (V) parameter output, and a current (I) parameter output. The example sound pressure level (SPL) model module is coupled to the plurality of parameter outputs of the of the DSM module and is operative to develop a predicted SPL. The example echo canceller module is responsive to the predicted SPL and to a near-end signal and operative to develop an echo-canceled output signal. 1. A system for echo cancellation comprising:a dynamic speaker management (DSM) module receptive to a far-end signal and operative to develop a modified far-end signal and a plurality of parameter outputs;a current/voltage sensing amplifier coupled to the modified far-end signal and having an amplifier output, a voltage (V) parameter output, and a current (I) parameter output;a sound pressure level (SPL) model module coupled to the plurality of parameter outputs of the of the DSM module and operative to develop a predicted SPL; andan echo canceller module responsive to the predicted SPL and to a near-end signal and operative to develop an echo-canceled output signal.2. A system for echo cancellation as recited in further comprising a speaker coupled to the amplifier output of the current/voltage sensing amplifier and a microphone providing the near-end signal.3. A system for echo cancellation as recited in wherein the DSM module includes a DSM control block and a speaker modeler.4. A system for echo cancellation as recited in wherein the speaker modeler uses the V and I parameter outputs of the current/voltage sensing amplifier to ...

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Номер записи: 67
21-02-2019 дата публикации

COMMUNICATION DISTURBANCE ANALYSIS DEVICE AND NON-TRANSITORY RECORDING MEDIUM STORING A COMPUTER READABLE PROGRAM

Номер: US20190058802A1
Автор: Ikeda Kazunori
Принадлежит: KONICA MINOLTA, INC.

Disclosed is a communication disturbance analysis device, including: a hardware processor that: obtains a first recorded data created by recording a first communication sound at a transmitter in one facsimile communication and a second recorded data created by recording a second communication sound at a receiver in the one facsimile communication, detects a silence section from each of the first recorded data and the second recorded data; and specifies an extracted section to be extracted as a recoded data to be analyzed from the first recorded data and the second recorded data in accordance with the detected silence section. 1. A communication disturbance analysis device , comprising: obtains a first recorded data created by recording a first communication sound at a transmitter in one facsimile communication and a second recorded data created by recording a second communication sound at a receiver in the one facsimile communication,', 'detects a silence section from each of the first recorded data and the second recorded data; and', 'specifies an extracted section to be extracted as a recoded data to be analyzed from the first recorded data and the second recorded data in accordance with the detected silence section., 'a hardware processor that2. The communication disturbance analysis device of claim 1 , wherein in case that after the hardware processor detects a specific control signal in a facsimile communication procedure claim 1 , the hardware processor detects a retransmission of the specific control signal claim 1 , the hardware processor detects a period from a detection of the specific control signal until a detection of the retransmission of the specific control signal claim 1 , as the silence section.3. The communication disturbance analysis device of claim 2 , further comprising: a signal level measurer and a signal frequency measurer claim 2 ,wherein in case that a frequency of a first signal having a predetermined level or more detected by the signal ...

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Номер записи: 68
02-03-2017 дата публикации

Nearend Speech Detector

Номер: US20170064087A1
Автор: Mani Senthil Kumar, Namani Gandhi
Принадлежит:

A nearend speech detector for classifying speech at a communication system receiving a microphone signal from a nearend microphone and a farend signal from a farend communication system, the nearend speech detector comprising: a signal processor configured to transform the microphone and farend signals into the frequency domain; a calculation unit configured to form: an estimate of a nearend signal representing nearend speech present in the microphone signal; and a measure of gain between the microphone signal and the nearend signal; and a signal classifier configured to classify speech at the communication system in dependence on a measure of variance of the gain and a measure of variance of the nearend signal. 1. A nearend speech detector for classifying speech at a communication system receiving a microphone signal from a nearend microphone and a farend signal from a farend communication system , the nearend speech detector comprising:a signal processor configured to transform the microphone and farend signals into the frequency domain; an estimate of a nearend signal representing nearend speech present in the microphone signal; and', 'a measure of gain between the microphone signal and the nearend signal; and, 'a calculation unit configured to forma signal classifier configured to classify speech at the communication system in dependence on a measure of variance of the gain and a measure of variance of the nearend signal.2. A nearend speech detector as claimed in claim 1 , wherein the calculation unit is configured to form the estimate of the nearend signal and the measure of gain in respect of each of a plurality of frequency bins claim 1 , and the measures of variance being measures of variance across the frequency bins.3. A nearend speech detector as claimed in claim 1 , wherein the signal processor is configured to transform the microphone and farend signals by performing Short Time Fourier Transform (STFT).4. A nearend speech detector as claimed in claim 1 ...

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Номер записи: 69
29-05-2014 дата публикации

Detecting Double Talk in Acoustic Echo Cancellation Using Zero-Crossing Rate

Номер: US20140146963A1
Автор: Muhammad Zubair Ikram
Принадлежит: Texas Instruments Inc

A method for acoustic echo cancellation in a communication device is provided that includes receiving a first near-end audio signal in the communication device, wherein the first near-end audio signal comprises acoustic echo of a far-end audio signal reproduced by the communication device, and performing echo cancellation on the first near-end audio signal to generate a second near-end audio signal with at least some of the acoustic echo removed, wherein the echo cancellation is performed responsive to presence or absence of double-talk (DT), and wherein a zero-crossing rate (ZCR) is used to detect the presence or absence of DT.

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Номер записи: 70
29-05-2014 дата публикации

Acoustic echo cancellation system

Номер: US20140146975A1
Автор: Kai-Ju Cheng, Kuo-Chun Huang, Rong-Quen Chen, Shih-Hsiang Lo
Принадлежит: QUANTA COMPUTER INC

An acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device.

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Номер записи: 71
27-02-2020 дата публикации

MICROELECTRONIC PACKAGE COMMUNICATION USING RADIO INTERFACES CONNECTED THROUGH WIRING

Номер: US20200065263A1
Автор: Chawla Gaurav, Elsherbini Adel A., Kamgaing Telesphor, Liff Shawna, Oster Sasha N.
Принадлежит:

Microelectronic package communication is described using radio interfaces connected through wiring. One example includes a system board, an integrated circuit chip, and a package substrate mounted to the system board to carry the integrated circuit chip, the package substrate having conductive connectors to connect the integrated circuit chip to external components. A radio on the package substrate is coupled to the integrated circuit chip to modulate the data onto a carrier and to transmit the modulated data. A radio on the system board receives the transmitted modulated data and demodulates the received data, and a cable interface is coupled to the system board radio to couple the received demodulated data to a cable. 1. An apparatus comprising:a system board;an integrated circuit chip;a package substrate mounted to the system board to carry the integrated circuit chip, the package substrate having conductive connectors to connect the integrated circuit chip to external components through the system board;a radio on the package substrate coupled to the integrated circuit chip to modulate the data onto a carrier and to transmit the modulated data;a radio on the system board to receive the transmitted modulated data and to demodulate the received data; anda cable interface coupled to the system board radio to couple the received demodulated data to a cable.2. The apparatus of claim 1 , wherein the cable is a multiple conductor flex cable.3. The apparatus of claim 1 , wherein the cable is at least one coaxial conductor.4. The apparatus of claim 1 , wherein the cable is an optical fiber.5. The apparatus of claim 4 , further comprising an optical fiber modulator coupled to the radio to receive the demodulated data and to modulate the received demodulated data onto an optical fiber.6. The apparatus of claim 1 , wherein the system board radio comprises:a plurality of radiating elements;a plurality of radio dies each coupled to a respective radiating element to demodulate ...

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Номер записи: 72
08-03-2018 дата публикации

IN-CAR COMMUNICATION HOWLING PREVENTION

Номер: US20180068672A1
Автор: Reuter Mike
Принадлежит:

Howling or oscillation in an in-vehicle communications system is prevented when the speaker gain or microphone gain is detected as giving rise to a loop gain equal to or greater than one for a particular frequency or band of frequencies or when the rate of gain of a frequency or band of frequencies increases at a rate that indicating howling will occur. Howling is prevented and not just stopped or suppressed, by adjusting the gain of all frequencies prior to the howling actually starting. 1. A method of preventing an in-vehicle audio communication system from howling , the in-vehicle audio communications system comprising an audio feedback loop , the loop comprising an in-vehicle microphone that detects in-vehicle sounds , an audio amplifier that receives signals from the microphone , and , a loudspeaker coupled to the audio amplifier and which transduces audio frequency signals from the amplifier into sound that is output into the vehicle , the signals from the loudspeaker being acoustically coupled to the microphone through the vehicle's interior , the amplifier having a changeable gain factor by which the audio signals from the microphone are amplified or suppressed , then output to the loudspeaker , the method comprising:determining a frequency-domain model of acoustic characteristics of the vehicle's interior, for a plurality of different audio frequency bands in order to detect changes in acoustic coupling between the loudspeaker and microphone in substantially real time;detecting audio frequency acoustic signals in the vehicle by the microphone;amplifying the detected audio frequency signals;monitoring magnitudes of audio signals detected by the microphone, in predetermined frequency bands of audio signals, and which were output from the loudspeaker;determining a loop gain for each of the plurality of different frequency bands;decreasing the loop gain of the audio amplifier for all audio frequencies, responsive to a determination that the loop gain for an ...

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Номер записи: 73
29-05-2014 дата публикации

FAR FIELD NOISE SUPPRESSION FOR TELEPHONY DEVICES

Номер: US20140148224A1
Автор: Chu Peter, GOESNAR Erwin, Potts Steve, Truong Kwan K.
Принадлежит: POLYCOM, INC.

Noise suppression systems and methods suppress far field noise in a microphone signal. A telephony system includes a main microphone and a reference microphone. In one example, the main microphone and the reference microphone can be located in the same device. In another example, the main microphone and the reference microphone can be located in two separate devices. A DSP can use the reference microphone signal to carry out suppression of far field noise in the main microphone signal. In one approach the DSP can determine an estimate of far field noise in the main microphone signal based on a noise estimate of the reference microphone signal and a reference and main microphone coupling estimate, and then subtract the far field noise estimate from the main microphone signal. Alternatively, the DSP can suppress the main microphone signal if it determines that a local talker is inactive. 1. A system comprising:a main device including a main microphone for generating a main microphone signal;a reference device including a reference microphone for generating a reference microphone signal;a communication link coupling the main device and the reference device; and receive the main microphone signal and the reference microphone signal, and', 'generate a noise suppressed main microphone signal by using the reference microphone signal to suppress far field noise in the main microphone signal,, 'a processor configured towherein the main device and the reference device, when not communicating with each other over the communication link, are configured to provide near end audio signals to a far end independently of each other.2. The system of claim 1 , wherein the distance between the main microphone and the reference microphone is variable.3. The system of claim 1 , wherein the processor is further configured to:for a consecutive plurality of time frames, split the main microphone signal and the reference microphone signal into a plurality of subbands;determine a coupling ...

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Номер записи: 74
10-03-2016 дата публикации

Method ad Apparatus for in-Ear Canal Sound Suppression

Номер: US20160072958A1
Автор: Jason McIntosh, John Usher, Marc Boillot, Steven Goldstein
Принадлежит: Personics Holdings Inc

A method and system of conferencing can include the steps of initiating a conference call at a communication device with two or more communication devices and selecting to suppress a voice communication of at least one communication device on the conference call where a modified electronic signal is generated with the selected at least one communication device so that the voice communication from the selected at least one communication device is inaudible. The method or system further includes sending the modified electronic signal to at least one other communication device on the conference call. Other embodiments are disclosed.

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Номер записи: 75
28-02-2019 дата публикации

SYSTEMS AND METHODS TO DISRUPT PHASE CANCELLATION EFFECTS WHEN USING HEADSET DEVICES

Номер: US20190068790A1
Автор: Sutton Chris
Принадлежит:

In applications where assisted listening headphones are worn inside of a theater, phase cancellation effects cause the headset wearer to perceive the audio as reduced in volume and distorted. These undesirable phase cancellation effects may be disrupted through preprocessing or real time processing of the headset audio track by summing acoustical noise with the original headset audio track and providing this altered audio track to the headset. The acoustical noise is modulated such that it is imperceptible to the headset wearer while at the same time disrupting undesirable phase cancellation effects, which would otherwise occur if the headset audio track was provided unaltered. Thus, the preprocessing of the headset audio preserves the integrity of the intended headset audio, as perceived by the headset wearer, in headsets worn in a theater environment. 1. A headset audio preprocessing method comprising:storing an audio track in memory of a mobile computing device;receiving in a microphone of the mobile computing device, contemporaneously played back audio;synchronizing playback of the stored audio track with the contemporaneously played back audio;acquiring acoustic noise; and,summing the acquired acoustic noise with the synchronized playback of the stored audio track in order reduce phase cancelation effects otherwise present in the synchronized playback of the stored audio track.2. The method of claim 1 , wherein the acoustic noise is pink noise.3. The method of claim 1 , wherein the acoustic noise is dithered noise.4. The method of claim 1 , wherein the contemporaneously played back audio is included as part of a motion picture played in a movie theater.5. The method of claim 1 , wherein the mobile computing device is a mobile phone.6. A computer program product for headset audio preprocessing claim 1 , the computer program product comprising:a non-transitory computer readable storage medium comprising a memory device having computer readable program code ...

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Номер записи: 76
11-03-2021 дата публикации

ADAPTIVE BEAMFORMING MICROPHONE METADATA TRANSMISSION TO COORDINATE ACOUSTIC ECHO CANCELLATION IN AN AUDIO CONFERENCING SYSTEM

Номер: US20210074311A1
Автор: Fink Dennis, King Matthew, LaBosco Mark, Parisi Alex, Walker Mitchell
Принадлежит: Crestron Electronics, Inc.

An audio processing device for use in a network connected audio conferencing system is provided, comprising: a network microphone array comprising two or more microphones (mics) and a beamforming circuit, wherein the network mic array is adapted to acquire acoustic audio signals, convert the same to electric audio signals, perform audio beamforming on the electric audio signals, and output a digital combined beamforming circuit output signal that comprises a first signal part and a second signal part, and wherein the first signal part comprises a first set of digital bits that comprises an active beam index, and wherein the active beam index encodes a selected beam position out of a possible N beam positions, and wherein the second signal part comprises a second set of digital bits that comprises a beamformed audio signal; a receiver adapted to receive the digital combined beamforming circuit output signal and split the same into the first signal part and the second signal part; a plurality of acoustic echo cancellation filter devices, each of which are adapted to receive the second signal part and a far end reference audio signal from a far end audio processing device, and perform acoustic echo cancellation on the beamformed audio signal in view of the far end audio signal; and an AEC filter circuit controller adapted to receive the first signal part, decipher the active beam index encoded in the first beamformed audio signal part to determine which of the N beam positions is active, and select a corresponding one of the plurality of acoustic echo cancellation filter devices based on the active one of N beam positions to generate an output audio signal from the audio processing device to be transmitted to the far end audio processing device. 1. An audio processing device for use in a network connected audio conferencing system , comprising: the first signal part comprises a first set of digital bits that comprises an active beam index, and wherein the active beam ...

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Номер записи: 77
19-03-2015 дата публикации

DYNAMIC NETWORK ROUTING DECISION PROCESSES AND SYSTEMS

Номер: US20150079999A1
Автор: TOKER M. Erol
Принадлежит:

Systems, processes, and computer readable media are described for dynamically routing calls over one of a plurality of possible networks to a client device. A process comprises determining, by a processor, at least one characteristic of a voice communication path to a client device across a plurality of different networks, where one network includes an Internet routing service and another network includes a mobile or cellular network service. The process further includes initiating voice communication between a telephony server and the client device via one of the at least two different networks based on the at least one characteristic. The process may further include accessing data preferences associated with the client device, where initiating the voice communications is further based on the data preferences associated with the client device. 1. A computer-implemented method for routing calls over one of a plurality of possible networks to a client device , the method comprising: determining at least one characteristic of a voice communication path to a client device across a plurality of different networks, wherein one network of the plurality of networks includes an Internet routing service and another network of the plurality of networks includes a cellular network service; and', 'initiating voice communication between a telephony server and the client device via one of the at least two different networks based on the at least one characteristic., 'at an electronic device having at least one processor and memory2. The computer-implemented method of claim 1 , further comprising accessing data preferences associated with the client device claim 1 , and wherein the initiating voice communications is further based on the data preferences associated with the client device.3. The computer-implemented method of claim 1 , wherein the at least one characteristic includes a Quality of Service metric.4. The computer-implemented method of claim 1 , wherein the at least one ...

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Номер записи: 78
15-03-2018 дата публикации

Dynamically increased noise suppression based on input noise characteristics

Номер: US20180075836A1
Автор: Bijal Joshi, Guillaume Lamy, Matthew R. Kirsch
Принадлежит: Continental Automotive Systems Inc

A maximum noise suppression level (G min ) is not a single constant value for an entire frequency range, but is allowed to vary across frequencies. The amount of variation is dynamically computed based on the input noise characteristics. For example, if there is excess noise in the lower frequency region, the maximum noise suppression level in that region will increase to suppress the noise in that frequency region. This feature can be enabled all the time, and will be active when the input conditions warrant extra noise suppression in a particular frequency region. Thus, the effort involved in manually tuning an audio system (e.g., hands-free telephony, voice-controlled automotive head unit, etc.) can be significantly reduced or eliminated.

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Номер записи: 79
07-03-2019 дата публикации

Communication system for communicating audio signals between a plurality of communication devices in a virtual sound environment

Номер: US20190075399A1
Автор: Andreas Jessen, Lars Elliot Nyegaard, Svend FELDT
Принадлежит: Sennheiser Communications AS

The present invention relates to a communication system for communication of a plurality of stereo audio signals between a plurality of communication devices, wherein the plurality of communication devices comprises a first communication device, a second communication device and at least a third communication device. Each communication device of the plurality of communication devices may comprise a signal processing unit, an audio interface configured to receive a local voice signal of a user of the communication device, a binaural rendering unit configured to render the local voice signal into a stereo local voice signal based on a first spatial information, an input communication interface configured to receive a first stereo audio signal and a second stereo audio signal of the plurality of stereo audio signals transmitted by the second communication device and the third communication device, respectively. The first stereo audio signal may comprise a second voice signal of a second user of the second communication device, and the second voice signal may include a second spatial information, and wherein the second stereo audio signal may comprise a third voice signal of a third user of the third communication device, and where the third voice signal may include a third spatial information. Furthermore, the communication device may comprise an output communication interface configured to transmit a third stereo audio signal of the plurality of stereo audio signals comprising the local voice signal provided with the first spatial information to the second communication device and the third communication device. The first stereo audio signal and the second stereo audio signal may be transmitted to the audio interface, and the user of the communication device experiences a virtual sound environment, wherein the second voice signal and the third voice signal is positioned in the virtual sound environment based on the second spatial information and the third spatial ...

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Номер записи: 80
17-03-2016 дата публикации

Sound Localization for an Electronic Call

Номер: US20160080577A1
Автор: Lyren Philip Scott, Norris Glen A.
Принадлежит:

A method moves a sound localization point (SLP) of a voice of a computer program during a voice exchange between the computer program and a person. 120.-. (canceled)21. A method executed by one or more electronic devices in an electronic system that move a sound localization point (SLP) of a voice of a computer program during a voice exchange between the computer program and a user , the method comprising:providing, through an electronic earphone being located at a head of the user and being in communication with a handheld portable electronic device (HPED), binaural sound to the user during the voice exchange between the computer program and the user such that an origin of the voice of the computer program occurs at the SLP at a first location in empty space that is away from and proximate to the user;receiving, from the user and at the HPED, a command to move the SLP of the voice of the computer program from the first location in empty space to a second location in empty space that is proximate to the first location and away from and proximate to the user;moving, based on the command and during the voice exchange between the computer program and the user, the SLP of the voice of the computer program from the first location in empty space to the second location in empty space that is proximate to the first location and away from and proximate to the user; andproviding, through the electronic earphone being located at the head of the user and being in communication with the HPED, the binaural sound to the user during the voice exchange between the computer program and the user such that the origin of the voice of the computer program occurs at the SLP at the second location in empty space that is proximate to the first location and away from and proximate to the user.22. The method of claim 21 , wherein the command is a voice command that instructs the one or more of the electronic devices in the electronic system to move the SLP of the voice of the computer program ...

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Номер записи: 81
15-03-2018 дата публикации

FULL DUPLEX VOICE COMMUNICATION SYSTEM AND METHOD

Номер: US20180077290A1
Автор: BASS Eric, ONOCHIE Francis Chukwuemeka, RITTER Dillon Reed, SILVA Rodolfo, ZARGAR Tanmay
Принадлежит:

A full duplex voice communication method constituted of: estimating an acoustic echo within a near-end signal; cancelling the estimated acoustic echo; detecting whether, or not, a change has occurred in a near-end acoustic echo path, the received near-end signal represents speech and the received far-end signal represents silence, wherein, responsive to the results thereof, the method is further constituted of: alternately attenuating frequency components of the echo cancelled near-end signal by a first frequency domain attenuation value and by a second greater frequency domain attenuation value; alternately attenuating a first function of the frequency component attenuated echo cancelled near-end signal by a first switchable attenuation value and by a second greater switchable attenuation value; and alternately attenuating a second function of the received far-end signal by a third switchable attenuation value and by a fourth greater switchable attenuation value. 1. A full duplex voice communication system comprising:a near-end input port arranged to receive a near-end signal;a far-end input port in communication with a far-end communication device and arranged to receive a far-end signal from said far-end communication device;an acoustic echo estimation functionality arranged, responsive to said received far-end signal, to estimate an acoustic echo within said received near-end signal;an acoustic echo cancellation functionality arranged to cancel said estimated acoustic echo from said received near-end signal;a frequency domain processing functionality;a first switchable attenuation functionality;a second switchable attenuation functionality;an echo path change detection functionality arranged to detect a change in a near-end acoustic echo path;a near-end speech detection functionality arranged to detect whether said received near-end signal represents speech;a far-end silence detection functionality arranged to detect whether said received far-end signal ...

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Номер записи: 82
16-03-2017 дата публикации

Automatic volume control of a voice signal provided to a captioning communication service

Номер: US20170078463A1
Автор: Brian Chevrier, Jeffrey C. Bullough, Shane A. ROYLANCE
Принадлежит: Sorenson IP Holdings LLC

Apparatuses and methods are disclosed for automatic volume control of an audio stream reproduced by a captioning communication service for use by a call assistant in generating a text transcription of a communication session between a hearing-impaired user and a far-end user. The automatic volume control automatically adjusts a volume of the audio stream reproduced by the captioning communication service responsive to a volume control command identifying which of the far-end voice signal and the near-end voice signal is active at a given time. The system further includes an echo modifier configured to add distortion to an echo portion of the far-end voice signal when generating the audio stream.

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Номер записи: 83
16-03-2017 дата публикации

NONLINEAR ACOUSTIC ECHO CANCELLATION BASED ON TRANSDUCER IMPEDANCE

Номер: US20170078489A1
Автор: Allen Justin, Lashkari Khosrow
Принадлежит:

An acoustic echo cancellation (AEC) system within an audio playback system of an electronic device, such as a mobile phone, may calculate an estimation of an acoustic echo based on parameters describing the transducer reproducing the audio playback signals. Those parameters may include, for example, a resistance and/or inductance of the transducer and a current through and/or a voltage across the transducer. The acoustic echo cancellation system may predict, for example, a coil velocity of the transducer based on the transducer impedance. Then, an echo may be estimated using the predicted coil velocity. That estimated echo may be output to the transducer to cancel echo in the playback signal. Additionally, that estimated echo may be used to predict nonlinearities in the transducer output and an appropriate signal generated to cancel nonlinear behavior. 1. An apparatus , comprising:a current input node for receiving a current signal that is indicative of a measured current into an audio speaker;a voltage input node for receiving a voltage signal that is indicative of a voltage value measured across the audio speaker; and calculating an impedance of the audio speaker based, at least in part, on the current signal received at the current input node and the voltage signal received at the voltage input node by using an adaptive filter; and', 'generating an acoustic echo cancellation signal based, at least in part, on the calculated impedance of the audio speaker., 'a processing circuit coupled to the current input node and to the voltage input node and configured to perform the steps of2. The apparatus of claim 1 , wherein the processing circuit is configured to generate the acoustic echo cancellation signal to cancel a nonlinear response of the audio speaker.3. The apparatus of claim 1 , wherein the processing circuit is configured to generate the acoustic echo cancellation signal by performing the step of calculating a back electromagnetic force (bemf) of the audio ...

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Номер записи: 84
23-03-2017 дата публикации

Method And Apparatus For Communication Enhanced Air Filtration Mask

Номер: US20170080262A1
Автор: Barocio Abiu, Freres Cameron
Принадлежит: Tech Tools, LLC

The inventive mask solves a major industry problem, which involves the challenge of effectuating clear communications in an environment requiring protective mask use, without the necessity of removing the protective mask. The inventive mask facilitates individual and workplace communications by employing a half-mask or full-mask face protector having a mounting bracket that facilitates incorporation of a basic, intermediate, or advanced level communications module with varying functionality, depending on user preference, workplace requirements, and similar consideration. The inventive mask also incorporates a replaceable selection panel for the communications module, which facilitates access to control mechanisms to enable mask-to-mask, wireless intercom communications and media connection capabilities, and contributes to user cost savings by obviating the prospect of full mask replacement if the replaceable selection panel becomes soiled or otherwise inoperable. The inventive mask also incorporates an intercom module to facilitate communications between inventive mask users and non-users of the inventive mask. 1. A communication enhanced protective mask to facilitate workplace communication comprising:(A) A face protector comprising a communication mount to facilitate user communication, a plurality of straps to help secure said mask to a user's face, a plurality of ventilation ports to facilitate respiration with air filtration, and an outer surface and an inner surface to provide a barrier between a mask user's face and airborne workplace contaminants, wherein said inner surface is configured to define a clean air compartment between said inner surface and the user's face to provide filtered, breathable air during use of said mask;(B) A communication module removeably attached to said communication mount of said face protector to facilitate communicate between other similar mask users and non-mask users without removing said mask from the user's face; and(C) A ...

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Номер записи: 85
26-03-2015 дата публикации

ECHO SUPPRESSOR USING PAST ECHO PATH CHARACTERISTICS FOR UPDATING

Номер: US20150086006A1
Автор: KAWABATA Naoya
Принадлежит: OKI ELECTRIC INDUSTRY CO., LTD.

In an echo suppressor, a frequency bin component detector compares a far-end signal amplitude spectrum with a threshold value for each frequency bin to determine whether or not each frequency bin includes a frequency component. A frequency bin echo path characteristic estimator uses the far-end signal amplitude spectrum in the frequency bins determined to have a frequency component by the frequency bin component detector and the near-end input signal amplitude spectrum of corresponding frequency bins to estimate the echo path characteristics of the frequency bins. An estimated echo signal calculation-and-echo suppressor section calculates estimated echo signals based on the echo path characteristic in each frequency bin and the far-end signal amplitude spectrum to suppress the estimated echo signals from the near-end input signal amplitude spectrum. 1. An echo suppressor apparatus comprising:a far-end signal amplitude spectrum calculator converting an incoming far-end signal into a frequency domain and calculating a far-end signal amplitude spectrum of the far-end signal;a near-end input signal amplitude spectrum calculator converting a near-end input signal into a frequency domain and calculating a near-end input signal amplitude spectrum of the near-end input signal;a frequency bin component detector using the far-end signal amplitude spectrum for a frequency bin to determine whether or not there is a frequency component in the frequency bin;a frequency bin echo path characteristic estimator using a far-end signal amplitude spectrum of the frequency bin determined to have a frequency component by said frequency bin component detector and a near-end input signal amplitude spectrum of a corresponding frequency bin of the near-end input signal to estimate an echo path characteristic of the frequency bin;an estimated-echo signal calculator calculating an estimated echo signal from the echo path characteristic for the frequency bin estimated by said frequency bin echo ...

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Номер записи: 86
05-03-2020 дата публикации

TELEPHONE SIGNAL PROCESSING

Номер: US20200076953A1
Автор: Baldwin Thomas, Tao Yufei
Принадлежит:

A method of processing a telephone signal comprising voice signals and data signals, the method comprising detecting the presence of an artefact in the telephone signal indicative of the presence of a data signal fragment associated with an earlier attenuation of a data signal and processing the telephone signal by further attenuating the telephone signal in the region of the artefact in order to remove the data signal fragment from the telephone signal. 1. A method of processing a telephone signal comprising voice signals and data signals , the method comprising:detecting the presence of an artefact in the telephone signal indicative of the presence of a data signal fragment associated with an earlier attenuation of a data signal; andprocessing the telephone signal by further attenuating the telephone signal in the region of the artefact in order to remove the data signal fragment from the telephone signal.2. A method according to claim 1 , wherein the data signal comprises at least one of:a) an acoustic signal,b) acoustic signal according to an acoustic data transmission protocol, andc) a DTMF tone.3. A method according to or claim 1 , wherein attenuating the telephone signal in the region of the artefact comprises at least one of:a) omitting or dropping or deleting a portion of the telephone signal,b) replacing a portion of the telephone signal, and/orc) modifying a portion of the telephone signal.4. A method according to any preceding claim claim 1 , further comprising further attenuating the telephone signal only when data signal fragments are expected to be present.5. A method according to any preceding claim claim 1 , wherein processing of the telephone signal occurs in the time domain.6. A method according to any preceding claim claim 1 , wherein the artefact comprises a spike in the telephone signal claim 1 , defined by the ratio of the maximum or peak amplitude of the telephone signal to the noise floor exceeding a threshold.7. A method according to claim ...

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Номер записи: 87
18-03-2021 дата публикации

COMPUTER-PROGRAMMED TELEPHONE-ENABLED DEVICES FOR PROCESSING AND MANAGING NUMEROUS SIMULTANEOUS VOICE CONVERSATIONS CONDUCTED BY AN INDIVIDUAL OVER A COMPUTER NETWORK AND COMPUTER METHODS OF IMPLEMENTING THEREOF

Номер: US20210084171A1
Автор: Detterman Eric, Tassone Anthony
Принадлежит: GREEN KEY TECHNOLOGIES, INC.

In some embodiments, the present invention provides for a computer-implemented method, including: causing, by a specifically programmed computer call management communication system, to transform, over a computer network, computing devices of users, into corresponding specialized call management devices, by having each computing device to execute a specialized call management client software application being in electronic communication with the specifically programmed computer call management communication system over the computer network by utilizing SIP; where the specialized call management client software application generates specialized graphical user interfaces configured to allow each user to concurrently initiate and maintain, over the computer network, a plurality of voice communications of distinct types with other users, by, for example, allowing each user to independently and dynamically divert, in real-time, any voice communication of any type to any audio device associated with a corresponding specialized call management device of such user. 1. A computer-implemented method , comprising:causing, by a programmed computer call management communication system, to transform, over a computer network, a plurality of computing devices of a plurality of users, into a corresponding plurality of call management devices, by having each computing device to execute a call management software application being in electronic communication with the programmed computer call management communication system over the computer network by utilizing at least a session initiation protocol (SIP);wherein the call management software application, upon the execution, generates a plurality of user interfaces configured to allow a plurality of users to concurrently initiate a plurality of voice communications of distinct types based, at least in part, on:maintaining each voice communication of the plurality of voice communications independent from another voice communication of ...

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Номер записи: 88
12-06-2014 дата публикации

Subband Domain Echo Masking for Improved Duplexity of Spectral Domain Echo Suppressors

Номер: US20140162731A1
Автор: Michiel Helsloot, Radolan Gavin
Принадлежит: Dialog Semiconductor BV

A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

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Номер записи: 89
24-03-2016 дата публикации

A METHOD AND APPARATUS FOR SUPPRESSION OF UNWANTED AUDIO SIGNALS

Номер: US20160086618A1
Автор: Nedjar Shmuel, NEORAN Itai, YAM ROY
Принадлежит:

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. 1. A device for removal of unwanted components in an audio signal , the device comprising:a processor operatively coupled to a memory, said processor configured to:receive a reference input and a processed input into the memory, where said processed input is a result of at least a reduction process of unwanted components of the audio signal;estimate envelope values for the processed input and for the reference input at a plurality of time and frequency instances;for each said 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 said second gain and said first gain; and,apply said output gain to the processed input; and,thereby producing a filtered output with unwanted components suppressed.2. The device of claim 1 , wherein said processed input and said reference input are in a time domain representation and said processor is configured to convert said processed input and said reference input ...

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Номер записи: 90
22-03-2018 дата публикации

Managing Telephony and Entertainment Audio in a Vehicle Audio Platform

Номер: US20180084344A1
Автор: Barksdale Tobe Z., Cheung Shiufun, Dublin Michael S., Holmi Douglas J., Maiguma Ryo, OSWALD Charles, Salvador Eduardo T., Song Zukui, VAUTIN Jeffery R.
Принадлежит: Bose Corporation

A signal input module receives at least one of an entertainment audio signal and a telephony audio signal from vehicle sound circuitry. A level control module executes gain level control logic to balance the entertainment audio signal and a telephony audio signal according to a ratio. A gain control signal is applied to at least one of the entertainment audio signal and the telephony audio signal. A routing module mixes the entertainment audio signal and the telephony audio signal and routes the mixed signal to an output channel associated with a speaker. 1. An apparatus comprising:a signal input module configured to receive an entertainment audio signal and a telephony audio signal from vehicle sound circuitry;a level control module configured to execute gain level control logic to balance the entertainment audio signal and the telephony audio signal relative to each other according to a predetermined ratio of gain levels, and to apply a gain control signal to at least one of the entertainment audio signal and the telephony audio signal; anda routing module configured to mix the entertainment audio signal and the telephony audio signal and route the mixed signal to an output channel associated with a speaker.2. The apparatus of claim 1 , wherein the predetermined ratio of gain levels is set to satisfy a Speech Transmission Index (STI) parameter claim 1 , so as to make the presentation of the entertainment audio signal and the telephony audio signal acceptable.3. The apparatus of any of the foregoing claims claim 1 , wherein the level control module is configured to balance the entertainment audio signal and the telephony audio signal in such a manner as that both the telephony and entertainment audio are audible and intelligible.4. The apparatus of any of the foregoing claims claim 1 , adapted to be used in an automobile passenger compartment including multiple car seats and speakers claim 1 , wherein the level control module is configured to balance the ...

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Номер записи: 91
19-06-2014 дата публикации

Correlation based filter adaptation

Номер: US20140169568A1
Автор: Qin Li, Vinod Prakash
Принадлежит: Microsoft Corp

Example apparatus and methods concern performing stereo acoustic echo cancellation using a correlation based filter adaptation control approach and without using stereo de-correlation. An embodiment includes a stereo adaptive filter that produces an echo removed microphone signal from received audio signals. The embodiment includes a mono adaptive filter that produces an echo removed microphone signal from the received audio signals. A correlation detector determines a level of correlation between the received signals and provides a signal to an adaptive filter controller. The adaptive filter controller controls how the stereo adaptive filter and the mono adaptive filter adapt audio echo cancellation as a function of the correlation between the received signals. A signal selector may select for output the signal from either the stereo adaptive filter or the mono adaptive filter based, for example, on the power level of the signals.

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Номер записи: 92
29-03-2018 дата публикации

EMERGENCY REPORT APPARATUS

Номер: US20180089987A1
Автор: HOSOI Toshihiro, YOKOI Shinichi
Принадлежит: Denso Corporation

In an emergency report apparatus, diagnostic data is modulated based on a preselected diagnostic modulation method by using a diagnostic carrier wave having a carrier wave frequency that is set within a range of frequencies detectable by a vehicle-mounted voice input instrument and is outside a voice band. A diagnostic electric signal is outputted to a voice output instrument. Demodulated data is generated by demodulating an input voice electric signal being a voice electric signal representing a voice detected by the voice input instrument based on a preselected diagnostic demodulation method. It is determines whether the demodulated data includes data that matches the diagnostic data. 1. An emergency report apparatus that is mounted in a vehicle to establish wireless data communication in an event of an emergency in the vehicle and enable an occupant of the vehicle to converse with an emergency report center , the vehicle including a voice input instrument and a voice output instrument ,the emergency report apparatus comprising: modulates preselected diagnostic data based on a preselected diagnostic modulation method by using a diagnostic carrier wave having a preselected carrier wave frequency that is within a range of frequencies detectable by the voice input instrument and is outside a voice band, and', 'outputs a diagnostic electric signal to the voice output instrument, the diagnostic electric signal being obtained by modulating the diagnostic data;, 'a modulation section that'}a demodulator that generates demodulated data by demodulating an input voice electric signal being a voice electric signal representing a voice detected by the voice input instrument based on a preselected diagnostic demodulation method; anda diagnostic data determination section that determines whether the demodulated data generated by the demodulator includes data that matches the diagnostic data.2. The emergency report apparatus according to claim 1 , further comprising:a plurality ...

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Номер записи: 93
25-03-2021 дата публикации

SPECTRAL BLENDING WITH INTERIOR MICROPHONE

Номер: US20210092233A1
Автор: Andersen Esge B., Hua Thanh P.
Принадлежит:

A headphone can include plurality of exterior microphones, that generates corresponding exterior microphone signals, an accelerometer that generates an accelerometer signal; and an interior microphone, not directly exposed to the environment, that generates an interior microphone signal. A processor of the headphone can be configured to generate an audio signal containing voice of a user, based on a) the accelerometer signal, b) the interior microphone signal, and c) the plurality of exterior microphone signals. 1. A headphone , comprising:a plurality of exterior microphones, directly exposed to an environment of the headphone, that generates corresponding exterior microphone signals;an accelerometer that generates an accelerometer signal;an interior microphone, not directly exposed to the environment, that generates an interior microphone signal; anda processor, configured togenerate an audio signal containing voice of a user, based on a) the accelerometer signal, b) the interior microphone signal, and c) the plurality of exterior microphone signals.2. The headphone of claim 1 , wherein the audio signal has a) a first frequency band generated based on the accelerometer signal claim 1 , b) a second frequency band generated based on the interior microphone signal claim 1 , and c) a third frequency band generated based on the exterior microphone signals claim 1 , the second frequency band having a higher frequency than the first frequency band claim 1 , and the third frequency band having a higher frequency than the second frequency band.3. The headphone of claim 1 , wherein generating the audio signal includes beamforming at least two of the exterior microphone signals to form an exterior beamformed signal used to generate the audio signal.4. The headphone of claim 1 , wherein the interior microphone is echo cancelled.5. The headphone of claim 1 , wherein the headphone is at least partially worn in the user's ear canal.6. The headphone of claim 1 , wherein the ...

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Номер записи: 94
31-03-2016 дата публикации

Detection of Acoustic Echo Cancellation

Номер: US20160094718A1
Автор: Akella Srinivas, Ghantasala Anupama, Mani Senthil Kumar
Принадлежит:

An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system, the echo cancellation detector comprising a comparison generator configured to compare the far-end signal with the near-end signal, a decision unit configured to make a determination about a first acoustic echo canceller based on that comparison and a controller configured to control an operation of a second acoustic echo canceller in dependence on the determination. 1. An echo cancellation detector for controlling an acoustic echo canceller that is configured to cancel an echo of a far-end signal in a near-end signal in a telephony system , the echo cancellation detector comprising:a comparison generator configured to compare the far-end signal with the near-end signal;a decision unit configured to make a determination about a first acoustic echo canceller based on a result of comparison by the comparison generator; anda controller configured to control an operation of a second acoustic echo canceller in dependence on the determination by the decision unit.2. An echo cancellation detector as claimed in claim 1 , wherein the decision unit is further configured to make a determination as to whether the first acoustic echo canceller is present in said telephony system or not.3. An echo cancellation detector as claimed in claim 2 , wherein the controller is further configured to:control the second acoustic echo canceller to be in a state in which it is not operating in response to a determination that the first acoustic echo canceller is present; andcontrol the second acoustic echo canceller to be in a state in which it is operating in response to a determination that the first acoustic echo canceller is not present.4. An echo cancellation detector as claimed in claim 2 , wherein the controller comprises:a monitoring unit configured to monitor whether the first acoustic echo canceller is ...

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Номер записи: 95
21-03-2019 дата публикации

Sound Signal Processing Device and Sound Signal Processing Method

Номер: US20190089839A1
Автор: INOUE Takayuki, UKAI Satoshi
Принадлежит:

A sound signal processing device includes: a microphone terminal to which a sound signal derived from sound received by a microphone is input; a loudspeaker terminal from which a sound signal directed to a loudspeaker is output; a first input terminal to which a sound signal from another proximal-end device is input; a first output terminal from which a sound signal directed to the other device is output; a distal-end input terminal to which a distal-end sound signal is input via a network; a distal-end output terminal from which a sound signal directed to the network is output; and at least one processor configured to execute stored instructions to establish at least one signal path from at least one of the microphone terminal, the first input terminal, or the distal-end input terminal, to at least one of the loudspeaker terminal, the first output terminal, or the distal-end output terminal. 1. A sound signal processing device comprising:a microphone terminal to which a sound signal derived from sound received by a microphone is input;a loudspeaker terminal from which a sound signal directed to a loudspeaker is output;a first input terminal to which a sound signal from another device at a proximal-end is input;a first output terminal from which a sound signal directed to the other device is output;a distal-end input terminal to which a distal-end sound signal is input via a network;a distal-end output terminal from which a sound signal directed to the network is output; andat least one processor configured to execute stored instructions to establish at least one signal path from at least one of the microphone terminal, the first input terminal, or the distal-end input terminal, to at least one of the loudspeaker terminal, the first output terminal, or the distal-end output terminal.2. The sound signal processing device according to claim 1 ,wherein the at least one processor is further configured to detect whether the subject device is connected to the network, ...

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Номер записи: 96
30-03-2017 дата публикации

Acoustic echo path change detection apparatus and method

Номер: US20170093460A1
Автор: Francis Chukwuemeka Onochie, Tanmay Zargar
Принадлежит: Microsemi Semiconductor US Inc

An acoustic echo path change detection apparatus constituted of: a time domain path change detection functionality arranged to: detect a change in a near-end acoustic echo path responsive to a time domain analysis of a near-end signal and a signal output by an acoustic echo canceller; and output an indication of the detected change, a frequency domain path change detection functionality arranged to: detect a change in the near-end acoustic echo path responsive to a frequency domain analysis of a far-end signal and the signal output by the acoustic echo canceller; and output an indication of the detected change, and a combination path change detection functionality arranged to: determine a first function of the output indication of the time domain path change detection functionality and the output indication of the frequency domain path change detection functionality; and output the determined first function.

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Номер записи: 97
26-06-2014 дата публикации

ECHO SUPPRESSION

Номер: US20140177822A1
Автор: Sorensen Karsten Vandborg
Принадлежит: MICROSOFT CORPORATION

Method, user device and computer program product for suppressing echo. An audio signal is output from a speaker. A microphone receives an audio signal, wherein the received audio signal includes an echo resulting from the outputted audio signal. A Finite Impulse Response filter estimate ĥ(n) is dynamically adapted in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) related to the echo in the received audio signal. The filter estimate ĥ(n) is used in an estimate of the echo power in the received audio signal, and the estimated echo power is used to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal. 1. A method of suppressing echo , the method comprising:outputting an audio signal;receiving an audio signal, wherein the received audio signal includes an echo resulting from said outputted audio signal;dynamically adapting a Finite Impulse Response filter estimate ĥ(n) in the time domain based on the outputted audio signal and the received audio signal to model an echo path h(n) of the echo in the received audio signal;using the filter estimate ĥ(n) in an estimate of the echo power in the received audio signal; andusing the estimated echo power to apply echo suppression to the received audio signal, thereby suppressing the echo in the received audio signal.2. The method of wherein said echo suppression is applied to the received audio signal without a prior step of applying echo cancellation to the received audio signal.3. The method of wherein the outputted audio signal is used in the estimate of the echo power in the received audio signal.4. The method of further comprising applying echo cancellation to the received audio signal claim 1 , wherein said echo suppression is applied after the echo cancellation in the processing of the received audio signal.5. The method of wherein said echo suppression which is applied to the received audio signal is ...

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Номер записи: 98
19-03-2020 дата публикации

ACOUSTIC ECHO CANCELLATION WITH ROOM CHANGE DETECTION

Номер: US20200091963A1
Автор: Christoph Markus, Kandade Rajan Vasudev
Принадлежит: Harman Becker Automotive Systems GmbH

Acoustic echo cancelling includes receiving a source signal and a sink signal; providing a first error signal representative of an echo-free residual signal based on a first set of coefficients based on the source signal and the sink signal, the first error signal forming an output signal of the controller; providing a second error signal based on a second set of coefficients based on the source signal and the sink signal; detecting a room change if the evaluated first second error signal is greater than a sum or product of the evaluated second first error signal and a first threshold; copying one of sets of reference coefficients stored in a memory to the second acoustic echo canceller; and copying the first set of coefficients from the first acoustic echo canceller as a set of reference coefficients into at least one of the second acoustic echo canceller and the memory. 1. An acoustic echo cancelling controller configured to receive a source signal representative of sound broadcast at a first position in a room and a sink signal representative of sound picked up at a second position in the room , the sound picked up at the second position being transferred from the first position according to a transfer characteristic , the controller comprising:a first acoustic echo canceller configured to receive the source signal and the sink signal, and to model the transfer function in an adaptive manner based on a first set of coefficients, the first acoustic echo canceller being further configured to provide a first error signal representative of an echo-free residual signal, the first error signal forming an output signal of the controller;a second acoustic echo canceller configured to receive the source signal and the sink signal, and to model the transfer function in a non-adaptive manner based on a second set of coefficients, the second acoustic echo canceller being further configured to provide a second error signal;a memory operatively coupled with the first acoustic ...

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Номер записи: 99
28-03-2019 дата публикации

Frequency domain noise attenuation utilizing two transducers

Номер: US20190096421A1
Автор: Jean Laroche
Принадлежит: CREATIVE TECHNOLOGY LTD

Embodiments may find applications to ambient noise attenuation in cell phones, for example, where a second microphone is placed at a distance from the voice microphone so that ambient noise is present at both the voice microphone and the second microphone, but where the user's voice is primarily picked up at the voice microphone. Frequency domain filtering is employed on the voice signal, so that those frequency components representing mainly ambient noise are de-emphasized relative to the other frequency components. Other embodiments are described and claimed.

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Номер записи: 100