67results about "Echo suppressors/cancellers" patented technology

This document relates to distributed devices teleconferencing. Some implementations can employ adaptive microphone selection based on signal characteristics such as signal-to-noise ratios or speech quality, and / or based on a microphone affinity approach. The selected microphone signals can be synchronized and mixed to generate a playback signal that is sent to a remote device. Further implementations can perform proximity-based mixing, where microphone signals received from devices in a particular room can be omitted from playback signals transmitted to other devices in the same room. These techniques can allow enhanced call quality for teleconferencing sessions where co-located users can employ their own devices to participate in a call with other users.

A conference system (400) comprises a central server (401), a plurality of conference units (411, 412, 413; 511), each having a microphone (421, 422, 423) and a loudspeaker (431, 432, 433), and a network (103) coupling the conference units (411, 412, 413; 511) to the central server (401). The central server (401) generates a mix signal (450) comprising contributions of speaker signals (441, 443) from a subset of conference units (411, 413) with open microphones (421, 423), and downstream transmits the mix signal (450) to the conference units (411, 412, 413). At least one conference unit (411, 413) is configured to subtract from the received mix signal (451, 453) an estimated contribution of its own speaker signal (441, 443) to thereby generate a loudspeaker signal (471, 473) comprising contributions of speaker signals (441, 443; 541) from the subset of conference units (411, 413) minus itself for output by its loudspeaker (431, 433).

A conference room device located in a conference room uses a conference room microphone to capture audio for one or more in-person conference participants located in the conference room during a conference. During the conference, audio is sampled from the microphone used by the conference room device and from microphones of one or more user devices determined to be located in the conference room. Based on a score calculated for the audio sampled from a user device being higher than a score calculated for audio sampled from the conference room device, the microphone of the user device is selected for audio capture for one or more of the in-person conference participants. The audio is output through conferencing software to which the conference room device is connected via a user interface tile for the conference room without displaying a new user interface tile.

Presented herein are techniques for seamless teleconferencing. A method can include selecting a first microphone for an audio call, the first microphone associated with a first device of multiple devices that are in proximity to one another and are each connected to the audio call. The method can further include selecting a first speaker for the audio call, the first speaker associated with one of the multiple devices. The method can further include inhibiting audio associated with all microphones and speakers associated with the multiple devices except for the first microphone and the first speaker. The method can further include switching to a second microphone associated with a second device of the multiple devices after a period of time by aligning audio streams of the first microphone and of the second microphone, inhibiting audio associated with the first microphone after the aligning, and enabling audio associated with the second microphone.

A method for a communications system between at least three devices, the method comprising: determining at least one device of the at least three devices is an ambient transmitting device, the ambient transmitting device configured with a controllable acoustic echo canceller to reduce echoes, wherein the at least one ambient transmitting device is configured to generate at least one ambient audio signal, and wherein the remaining at least two devices are non-ambient devices; controlling the at least one ambient transmitting device to reduce the effect of the controllable acoustic echo canceller; and applying at least one delay difference to at least one communications path between non-ambient devices.

A conference system (400) comprises a central server (401), a plurality of conference units (411, 412, 413; 511), each having a microphone (421, 422, 423) and a loudspeaker (431, 432, 433), and a network (103) coupling the conference units (411, 412, 413; 511) to the central server (401). The central server (401) generates a mix signal (450) comprising contributions of speaker signals (441, 443) from a subset of conference units (411, 413) with open microphones (421, 423), and downstream transmits the mix signal (450) to the conference units (411, 412, 413). At least one conference unit (411, 413) is configured to subtract from the received mix signal (451, 453) an estimated contribution of its own speaker signal (441, 443) to thereby generate a loudspeaker signal (471, 473) comprising contributions of speaker signals (441, 443; 541) from the subset of conference units (411, 413) minus itself for output by its loudspeaker (431, 433).

Disclosed is an audio device with sidetone processing, the audio device comprising: a plurality of microphones; an output transducer configured to output a proximal audio output signal; and a processor circuit comprising a signal processor module configured to provide a far-end audio output signal and a sidetone module configured to provide a sidetone audio output signal. The signal processor module is configured to process a first audio input signal and a second audio input signal to provide a plurality of filter parameters, the sidetone module is configured to obtain first data indicative of a plurality of filter parameters and to process the first audio input signal and the second audio input signal using one or more filters based on the first data to provide a sidetone audio output signal, and wherein the sidetone audio output signal comprises a sidetone signal, the near-end audio output signal is based on the sidetone audio output signal and the far-end audio input signal.

A device includes a memory configured to buffer an audio input from an audio source. The audio input corresponds to a voice call. The device also includes an audio processor. The audio processor is configured to select, based on a power indicator, a noise reducer mode from multiple noise reducer modes. The audio processor is also configured to generate, based on the selected noise reducer mode, an audio output based on the audio input. The audio processor is further configured to output the audio output.

A method, a system, and a server for reducing noise in a workspace are disclosed. The workspace may have a plurality of terminals connected to a server via a communication network. The method may include detecting a noise level, which is above a predetermined threshold value, in or adjacent to at least one of the terminals. The noise level in or adjacent to a terminal may be detected by monitoring noise emitted from or in the vicinity of that terminal by a microphone associated with that terminal. The method may also include identifying the terminal at which the noise level exceeds the predetermined threshold. Further, the method may include initiating a measure for reducing the noise level.

The present disclosure relates to a method for determining the perceptual effect of the amount of echo or reverberation in a degraded audio signal on the perceived quality of the degraded audio signal, wherein the degraded audio signal is obtained by transmitting a reference audio signal through the audio transmission system to provide the degraded audio signal. The method includes performing a windowing operation on the degraded digital audio samples and the reference digital audio samples by multiplying the degraded digital audio samples and the reference digital audio samples by a window function to obtain the degraded digital audio samples and the reference digital audio samples. A local estimate of the amount of echo or reverberation is determined based on these samples.

A method of facilitating audio conferencing comprises predicting or determining whether a first user (76) of a first user device (71) is able to hear a second user (77) of a second user device (72) directly and enabling the first user device to reproduce audio captured by the second user device while the first user device reproduces audio captured by a third user device (73) if the first user is predicted or determined not to be able to hear the second user directly or preventing that the first user device reproduces audio captured by the second user device while the first user device reproduces audio captured by the third user device if the first user is predicted or determined to be able to hear the second user directly.

An information processing system including a processor is provided. The processor is configured to, in a case where a first information processing device and a second information processing device participate in a same web conference, acquire position information of the first information processing device and the second information processing device, measure a distance between the first information processing device and the second information processing device based on the acquired position information, and in response to the measured distance being shorter than a predetermined distance and in response to a sound output unit of the first information processing device being active, control a sound-pick-up unit of the second information processing device to be inactive and control a display screen of the second information processing device to display a message indicating that a howl is likely to occur.

Systems, methods, and computer-readable storage devices are disclosed for optimizing speech enhancement components to use in speech communication systems using non-intrusive speech quality assessment. One method including: receiving audio data, the audio data including speech, and the audio data having been processed by at least one speech enhancement component; detecting a first quality of the speech of the audio data using a trained non-intrusive speech quality assessment (NISQA) model, the trained NISQA model trained to detect quality of speech automatically; and changing one or more of the at least one speech enhancement component based on the detected first quality of the speech.

A method of compensating for environmental noise during a teleconference may involve: estimating, by a control system, a current speech spectrum corresponding to speech of remote teleconference participants; estimating, by the control system, a current noise spectrum corresponding to environmental noise in a local environment in which a local teleconference participant is located; calculating, by the control system, a current speech intelligibility index (SII) based, at least in part, on the current speech spectrum and the current noise spectrum; determining, by the control system and based at least in part on the current SII, whether to make an adjustment of a local audio system used by the local teleconference participant, wherein the determining involves evaluating the current SII according to one or more target SII parameters; and updating at least one of the one or more target SII parameters responsive to user input corresponding to a playback volume change.

This document relates to distributed teleconferencing. Some implementations can employ personalized enhancement models to enhance microphone signals for participants in a call. Further implementations can perform proximity-based mixing, where microphone signals received from devices in a particular room can be omitted from playback signals transmitted to other devices in the same room. These techniques can allow enhanced call quality for teleconferencing sessions where co-located users can employ their own devices to participate in a call with other users.

Systems, methods, and computer-readable storage devices are disclosed for optimizing speech enhancement components for use in a speech communication system using non-invasive speech quality evaluation. A method comprising: receiving audio data, the audio data comprising speech and having been processed by at least one speech enhancement component; detecting a first quality of the speech of the audio data using a trained non-invasive speech quality assessment (NISQA) model, the trained NISQA model being trained to automatically detect a quality of the speech; and altering one or more of the at least one speech enhancement component based on the detected first quality of the speech.

A participant computing device connected to a teleconferencing session receives mixed audio data from a teleconference computing system orchestrating the session. The mixed audio data comprises a mix of audio captured by other devices connected to the session. The mixed audio and audio captured by the other devices prior to receipt of the mixed audio data are compared to determine that the participant computing device is co-located with at least one of the other devices. Information indicating that some of the other devices are featured devices that most actively contribute to the mixed audio is obtained. Co-location information for each of the featured devices is received. The co-location information is indicative of whether a featured device is co-located with the participant computing device. Based on the co-location information, a determination is made that one of the featured devices is co-located with the participant computing device.

Echo cancellation for bidirectional audio communication includes receiving, at an AEC system, audio signals from one or more microphones based at least in part on a near-end signal and a reproduced far-end signal. One or more loudspeakers reproduce the far-end signal. The AEC system operates at least in part with one or more filters to update an estimate of coefficients of a channel from the one or more loudspeakers to the one or more microphones. One or more control parameters that affect operation of the AEC system are configurable and are set to one or more values in a range of values based on an accuracy of the estimate of the coefficients of the estimated channel and characteristics of the near-end signal. The AEC system controls the one or more filters with different values of the one or more control parameters at different times.

The present disclosure is directed towards a semi-global mute system and method. In one example, a method for muting communication devices includes receiving a message indicating the semi-global mute switch has been activated. The method further includes upon receiving the message: preventing a co-located communication device from receiving the audio signals transmitted from the first audio communication device, permitting the first audio communication device to transmit signals to the remote communication device, and permitting the first audio communication device and the co-located communication device to receive audio signals transmitted from the remote communication device.

A conference room device located in a conference room uses a conference room microphone to capture audio for one or more in-person conference participants located in the conference room during a conference. During the conference, audio is sampled from the microphone used by the conference room device and from microphones of one or more user devices determined to be located in the conference room. Based on a score calculated for the audio sampled from a user device being higher than a score calculated for audio sampled from the conference room device, the microphone of the user device is selected for audio capture for one or more of the in-person conference participants. The audio is output through conferencing software to which the conference room device is connected via a user interface tile for the conference room without displaying a new user interface tile.

The invention relates to a method of determining a perceptual impact of an amount of echo or an amount of reverberation in a degraded audio signal on a perceptual quality of the degraded audio signal, wherein the degraded audio signal is received from an audio transmission system, the degraded audio signal being obtained by a transmission of a reference audio signal by the audio transmission system to provide the degraded audio signal. The method comprises performing a windowing operation on the degraded audio signal and the reference audio signal by multiplying the degraded audio signal and the reference audio signal with a window function to produce degraded digital audio samples and reference digital audio samples. A local estimate of the amount of echo or the amount of reverberation is determined based on the degraded digital audio samples and the reference digital audio samples.

The present disclosure is directed towards a semi-global mute system and method. In one example, a method for muting communication devices includes receiving a message indicating the semi-global mute switch has been activated. The method further includes upon receiving the message: preventing a co-located communication device from receiving the audio signals transmitted from the first audio communication device, permitting the first audio communication device to transmit signals to the remote communication device, and permitting the first audio communication device and the co-located communication device to receive audio signals transmitted from the remote communication device.

A headset for generating a non-delayed sidetone for a user, the headset including an input for receiving a playback signal, the playback signal including respective delayed audio signals from one or more users mixed together as the playback signal, one or more users including a first user and the mixed audio signal from the first user being a delayed input audio signal; a signal remover configured to generate a first processed signal by removing the delayed audio signal for the first user from the playback signal; and a signal mixer coupled to the signal remover, the signal mixer configured to: generate a second processed signal by mixing the first processed signal with a non-delayed input audio signal from the first user, and output the second processed signal, the non-delayed input audio signal acting as a sidetone for the first user.

Systems and methods for providing voice feedback for video conferences are provided. A computer-implemented method includes joining, by a client device, a video conference hosted by a video conference provider, the video conference having a plurality of participants using a plurality of client devices. With voice feedback enabled, the client device may receive a first audio stream from the video conference provider, the first audio stream associated with a first participant of the plurality of participants and a second audio stream from a user of the client device, the second audio stream comprising a voice of the user of the client device. The client device can play the first audio stream on one or more channels of one or more audio devices connected to the client device and play the second audio stream on the one or more channels of the one or more audio devices.

A method of compensating for environmental noise during a teleconference may involve: estimating, by a control system, a current speech spectrum corresponding to speech of remote teleconference participants; estimating, by the control system, a current noise spectrum corresponding to environmental noise in a local environment in which a local teleconference participant is located; calculating, by the control system, a current speech intelligibility index (SII) based, at least in part, on the current speech spectrum and the current noise spectrum; determining, by the control system and based at least in part on the current SII, whether to make an adjustment of a local audio system used by the local teleconference participant, wherein the determining involves evaluating the current SII according to one or more target SII parameters; and updating at least one of the one or more target SII parameters responsive to user input corresponding to a playback volume change.

A device includes a memory configured to buffer an audio input from an audio source. The audio input corresponds to a voice call. The device also includes an audio processor. The audio processor is configured to select, based on a power indicator, a noise reducer mode from multiple noise reducer modes. The audio processor is also configured to generate, based on the selected noise reducer mode, an audio output based on the audio input. The audio processor is further configured to output the audio output.

A method of compensating for ambient noise during a teleconference may involve estimating, by a control system, a current speech spectrum corresponding to a speech of a remote teleconference participant; estimating, by the control system, a current noise spectrum corresponding to ambient noise in a local environment in which a local teleconference participant is located; calculating, by the control system, a current speech intelligibility index (SII) based at least in part on the current speech spectrum and the current noise spectrum; determining, by the control system and based at least in part on the current SII, whether to adjust a local audio system used by the local teleconference participant, where the determining involves evaluating the current SII according to one or more target SII parameters; and updating at least one of the one or more target SII parameters in response to a user input corresponding to the playback volume change.

A conference system (400) comprising: a central server (401); a plurality of conference units (411, 412, 413, 511), each conference unit having a microphone (421, 422, 423) and a loudspeaker (431, 432, 433); and a network (103) connecting the conference units (411, 412, 413; 511) to a central server (401). A central server (401) generates a mixed signal (450) containing contributions of speaker signals (441, 443) from a subset of conference units (411, 413) with microphones (421, 423) in an on state, and the central server (401) is configured to downlink the mixed signal (450) to the conference units (411, 412, 413). The at least one conference unit (411, 413) is configured to subtract the estimated contribution (591) of its own speaker signal (441, 443) from the received mixed signal (451, 453) to generate a speaker signal (471, 473) for its speaker output, the speaker signal (471, 473) containing the contribution of the speaker signal (441, 443; 541) from a subset of the conference units (411, 413) but subtracting the own speaker signal of the at least one conference unit.

Systems, methods, and computer-readable storage devices are disclosed for optimizing speech enhancement components to use in speech communication systems using non-intrusive speech quality assessment. One method including: receiving, from a computing device over a network, audio data, the audio data including speech; detecting a first quality of the speech of the audio data using a trained non-intrusive speech quality assessment (NISQA) model, the trained NISQA model trained to detect quality of speech automatically; determining whether the computing device is a low-quality endpoint based on the first quality of speech of the audio data; and transferring, from the computing device over the network, at least one speech enhancement component to at least one server device when the computing device is determined to be a low-quality endpoint.

Systems, methods, and computer-readable storage devices are disclosed for optimizing speech enhancement components to be used in a speech communication system using non-invasive speech quality evaluation. A method includes receiving audio data from a computing device over a network, the audio data including speech; detecting a first quality of the speech of the audio data using a trained non-invasive speech quality assessment (NISQA) model, the trained NISQA model being trained to automatically detect a quality of the speech; determining whether the computing device is a low quality endpoint based on a first quality of the speech of the audio data; and transmitting at least one speech enhancement component from the computing device to at least one server device over the network when the computing device is determined to be a low quality endpoint.