Hewlett-Packard Development Company, L.p.

Sunil Bharitkar

Madhu Sudan Athreya

Srikanth Kuthuru

Wensen Liu

Rafael A Ballagas

Richard F Sweet

David Murphy

In example implementations, an apparatus is provided. The apparatus includes an adaptive filter and a double talk detector in communication with the adaptive filter. The adaptive filter is to calculate a transfer function with coefficients for a particular time that is applied to an output signal of a microphone to cancel echoes caused by a reference signal in the output signal of the microphone. The double talk detector is to determine a peak of the coefficients, detect double talk based on a location of the peak of the coefficients, and transmit a pause signal to the adaptive filter in response to detection of the double talk, wherein the pause signal is to pause a calculation of updates to the coefficients by the adaptive filter.

Double talk detectors

Acoustic echo cancellation for a video conference system is described. A location of a person in a room can be determined. An audio signal received from the location of the person can be captured using beamforming. An acoustic echo cancellation parameter can be determined based in part on the audio signal captured from the location of the person. Acoustic echo cancellation can be performed on the audio signal using the acoustic echo cancellation parameter.

Acoustic echo cancellation

An example non-transitory computer-readable storage medium comprises instructions that, when executed by a processing resource of a computing device, cause the processing resource to, responsive to identifying a spatial area of attention of a gaze of a user, enhance audio associated with the spatial area of attention. The instructions further cause the processing resource to modify the enhancing based at least in part on a physiological observation of the user while hearing the audio.

Modify audio based on physiological observations

According to examples, an apparatus may include a processor and a non-transitory computer readable medium on which is stored instructions that the processor may execute to access an audio signal captured by a microphone of a user's speech while the microphone is in a muted state. The processor may also execute the instructions to analyze a spectral or frequency content of the accessed audio signal to determine whether the user was facing the microphone while the user spoke. In addition, based on a determination that the user was facing the microphone while the user spoke, the processor may execute the instructions to unmute the microphone.

Microphone control based on speech direction

In some examples, an audio control system can include a first set of resources, a second set of resources and a controller. The first set of resources can generate a frequency energy band representation of a multi-channel source audio input. Additionally, the second set of resources can determine at least a value representing a strength of correlation between multiple channels of the multi-channel source audio input. Moreover, the audio output controller can determine a set of control parameters for tuning sound creation from an audio signal generator to reflect a set of spatial characteristics of the source audio input, based on the frequency energy band representation and the first value.

Spatial characteristics of multi-channel source audio

In some examples, immersive audio rendering may include determining whether an audio signal includes a first content format including stereo content, or a second content format including multichannel or object-based content. In response to a determination that the audio signal includes the first content format, the audio signal may be routed to a first block that includes a low-frequency extension and a stereo to multichannel upmix to generate a resulting audio signal. Alternatively, the audio signal may be routed to another low-frequency extension to generate the resulting audio signal. The audio signal may be further processed by performing spatial synthesis on the resulting audio signal, and crosstalk cancellation on the spatial synthesized audio signal. Further, multiband-range compression may be performed on the crosstalk cancelled audio signal, and an output stereo signal may be generated based on the multiband-range compressed audio signal.

Immersive audio rendering

In some examples, adaptive speech intelligibility control for speech privacy may include determining, based on background noise at a near-end of a speaker, a noise estimate associated with speech emitted from the speaker, and comparing, by using a specified factor, the noise estimate to a speech level estimate for the speech emitted from the speaker. Adaptive speech intelligibility control for speech privacy may further include determining, based on the comparison, a gain value to be applied to the speaker to produce the speech at a specified level to maintain on-axis intelligibility with respect to the speaker, and applying the gain value to the speaker.

Adaptive speech intelligibility control for speech privacy

A communication system may include, in an example, a first computing device communicatively coupled, via a network, to at least a second computing device maintained at a geographically distinct location than the first computing device; the first computing device including: an array of audio output devices and a processor to receive transmitted speech data and metadata describing an estimated direction of arrival (DOA) of speech from a plurality of speakers at an array of microphones at the second computing device and render audio at the array of audio output devices associated with the first computing device by eliminating spatial collision during rendering; said spatial collision arising due to the low angular separation of the estimated DOA of a plurality of speakers.

Eliminating spatial collisions due to estimated directions of arrival of speech

In some examples, loudness enhancement based on multiband range compression may include determining, based on variations in compression parameters that include compression thresholds and compression ratios, corresponding variations in loudness levels for a specified loudness standard. A learning model may be trained based on the variations in the compression parameters and the corresponding variations in the loudness levels. A specified loudness level for a device may be ascertained, for example, from a user of the device. The compression parameters for the specified loudness level may be determined based on the trained learning model. Further, sub-band compression of an input audio signal may be performed, based on the determined compression parameters, by processing the input audio signal using a perfect reconstruction filterbank.

Loudness enhancement based on multiband range compression

In some examples, crosstalk cancellation for speaker-based spatial rendering may include perceptually smoothing head-related transfer functions (HRTFs) corresponding to ipsilateral and contralateral transfer paths of sound emitted from first and second speakers to corresponding first and second destinations. The crosstalk cancellation may further include inserting an inter-aural time difference in the perceptually smoothed HRTFs corresponding to the contralateral transfer paths. A crosstalk canceller may be generated by inverting the perceptually smoothed HRTFs corresponding to the ipsilateral transfer paths and the perceptually smoothed HRTFs corresponding to the contralateral transfer paths including the inserted inter-aural time difference.

Crosstalk cancellation for speaker-based spatial rendering

Described herein is a technology related to a mobile device with a pair of stereo speakers. The mobile device has an orientation detection system that detects the orientation of the mobile device and a crosstalk cancellation system that performs crosstalk cancellation with complementary cardioid beams in response to the detected orientation of the mobile device. The mobile device also has an audio system that emits complementary cardioid sound beams from the speakers.

Crosstalk cancellation for stereo speakers of mobile devices

According to examples, an apparatus may include a processor and a memory on which is stored machine readable instructions that are to cause the processor to determine a reference frame from a plurality of frames received at multiple different times, in which each of the plurality of frames includes audio signal data, and in which the reference frame includes audio signal data that identifies a highest audio signal level among audio signals identified in the plurality of frames. The reference frame may be time-aligned with each of the plurality of frames other than the reference frame to obtain respective time-aligned frames. The audio signals identified in each of the respective time-aligned frames may be added together to generate respective added audio signals. The respective added audio signals may be combined together to obtain a combined audio signal and the combined audio signal may be outputted.

Combined audio signal output

In some examples, matrix decomposition of audio signal processing filters for spatial rendering may include determining first and second spatial synthesis filters respectively as a sum and a difference of ipsilateral and contralateral spatial synthesis filters, and determining first and second crosstalk cancellation filters respectively as a sum and a difference of ipsilateral and contralateral crosstalk cancellation filters. A combined spatial synthesizer and crosstalk canceller that includes a first combined filter and a second combined filter may be determined based on application of matrix decomposition to the first and second spatial synthesis filters and the first and second crosstalk cancellation filters. Further, spatial synthesis and crosstalk cancellation on first and second input audio signals may be performed based on application of the combined spatial synthesizer and crosstalk canceller.

Matrix decomposition of audio signal processing filters for spatial rendering

An example system includes a filter bank of sub-octave filters to separate a lower frequency portion of an audio input stream into a number of sub-bands. A detector bank of detectors coupled with the filter bank determines an audio power level in each of the sub-bands. A sub-band selection engine coupled with the detector bank determines a dominant sub-band. A first filter engine isolates the dominant sub-band from the audio input stream and a harmonic engine coupled with the first filter generates harmonics of the dominant sub-band. A second filter engine coupled with the harmonic engine selects a sub-set of the harmonics to combine with a higher frequency portion of the audio input stream.

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Palo Alto, CA, United States of America

Sunil Bharitkar