Qualcomm Incorporated

Dinesh Ramakrishnan

Song Wang

Eddie L T Choy

Ian Ernan Liu

Erik Visser

Lae-Hoon Kim

Samir Kumar Gupta

Pei Xiang

Asif Mohammad

Deepak Kumar Challa

Naga Chandan Babu Gudivada

Kwokleung Chan

Ren Li

Hyun Jin Park

Vishnu Vardhan Kasilya Sudarsan

Brian Lee Momeyer

Wade Lyle Heimbigner

Sang-Uk Ryu

Ricardo De Jesus Bernal Castillo

Sanghyun Chi

Andre Schevciw

Roy Benjamin Silverstein

Eddie Choy

Rameshkumar Karuppusamy

Jimeng Zheng

Andre Gustavo Pucci Schevciw

Homayoun Shahri

Andrew Kostic

Ravi Satyanarayanan

Krishna Kumar Malthakal

Luis D Oliveira

Andre Gustavo Schevciw

Eddie LT Choy

Systems and techniques are provided for performing spatial audio recording. For instance, a process can include detecting an occlusion for at least one audio frame of one or more audio frames associated with a spatial audio recording. During the spatial audio recording, the process can further include selecting, based on detection of the occlusion, at least one of an occluded spatial filter for the one or more audio frames or a non-occluded spatial filter for the one or more audio frames.

Generating restored spatial audio signals for occluded microphones

System and method for detecting and identifying noises in a sound signal occurring during a call on a mobile device and selectively filtering and suppressing the noises in the sound signal are provided. In the mobile device, a processor is configured to receive a sound signal, detect noises in the received sound signal, identify the noises in the received sound signal, display the identified noises in a user interface (UI), receive a selection of the displayed identified noises from the UI, and filter the received selection of the displayed identified noises from the received sound signal. The processor may use a machine learning module with a neural network to detect and identify the noises in the received sound signal.

Selective suppression of noises in a sound signal

Aspects of the disclosure relate to using a display as a sound emitter and may relate to an electronic device including a display. In particular a vibration sensor such as an accelerometer is physically coupled to the display and senses display vibration to provide a high accuracy feedback loop with respect to representing actual audio output from the display. The electronic device includes an actuator physically coupled to the display and configured to cause vibration of the display in response to an audio signal. The electronic device further includes a vibration sensor physically coupled to the display and configured to output a vibration sensor signal proportional to the vibration of the display due to the actuator.

Feedback control for display as sound emitter

Methods, systems, computer-readable media, and apparatuses for acoustic echo cancellation during playback of encoded audio are presented. In some embodiments, a decoder is arranged to decode an encoded media signal to produce an echo reference signal, and an echo canceler is arranged to perform an acoustic echo cancellation operation, based on the echo reference signal, on an input voice signal to produce an echo-cancelled voice signal. The echo canceler may be configured to reduce, relative to an energy of a voice component of the input voice signal, an energy of a signal component of the input voice signal that is based on audio content from the encoded media signal.

Acoustic echo cancellation during playback of encoded audio

A method for audio signal processing is described. The method includes decomposing a recorded auditory scene into a first category of localizable sources and a second category of ambient sound. The method also includes recording an indication of the directions of each of the localizable sources. The method may be performed with a device having a microphone array.

Three-dimensional sound capturing and reproducing with multi-microphones

In general, techniques are described for limiting active noise cancellation output. As one example, an apparatus comprising one or more processors may perform the techniques. The one or more processors may be configured to, when an estimated noise level increases, dynamically lowering application of active noise cancellation to at least a portion of an audio signal to obtain at least a portion of an active noise cancelled version of the audio signal.

Limiting active noise cancellation output

A method includes receiving, at a processor, a first data frame at a first time from a first microphone. The method also includes receiving a second data frame at the first time from a second microphone. The method further includes calculating a power ratio of the first microphone and the second microphone based on the first data frame and the second data frame in response to determining that the first data frame and the second data frame are noise data frames.

Automated gain matching for multiple microphones

To reduce the risk of brown outs and resets in a mobile station using a far-field speaker, a voltage level of a battery and a level of an input audio signal are monitored. When the level of the audio signal increases past a threshold level, and the voltage level of the battery falls below a threshold voltage, an attenuation is determined and applied to the audio signal. The applied attenuation reduces the volume of the audio signal, which reduces the risk of a brown out or a reset due to insufficient voltage in the mobile station.

Battery power monitoring and audio signal attenuation

Signal processing solutions take advantage of microphones located on different devices and improve the quality of transmitted voice signals in a communication system. With usage of various devices such as Bluetooth headsets, wired headsets and the like in conjunction with mobile handsets, multiple microphones located on different devices are exploited for improving performance and/or voice quality in a communication system. Audio signals are recorded by microphones on different devices and processed to produce various benefits, such as improved voice quality, background noise reduction, voice activity detection and the like.

Speech enhancement using multiple microphones on multiple devices

A method of orientation-sensitive recording control includes indicating, within a portable device and at a first time, that the portable device has a first orientation relative to a gravitational axis and, based on the indication, selecting a first pair among at least three microphone channels of the portable device. This method also includes indicating, within the portable device and at a second time that is different than the first time, that the portable device has a second orientation relative to the gravitational axis that is different than the first orientation and, based on the indication, selecting a second pair among the at least three microphone channels that is different than the first pair. In this method, each the at least three microphone channels is based on a signal produced by a corresponding one of at least three microphones of the portable device.

Systems, methods, apparatus, and computer-readable media for orientation-sensitive recording control

A method for echo cancellation and noise suppression is disclosed. Linear echo cancellation (LEC) is performed for a primary microphone channel on an entire frequency band or in a range of frequencies where echo is audible. LEC is performed on one or more secondary microphone channels only on a lower frequency range over which spatial processing is effective. The microphone channels are spatially processed over the lower frequency range after LEC. Non-linear noise suppression post-processing is performed on the entire frequency band. Echo post-processing is performed on the entire frequency band.

Integrated echo cancellation and noise suppression

A method for suppressing ambient noise using multiple audio signals may include providing at least two audio signals captured by at least two electro-acoustic transducers. The at least two audio signals may include desired audio and ambient noise. The method may also include performing beamforming on the at least two audio signals in order to obtain a desired audio reference signal that is separate from a noise reference signal.

Methods and apparatus for suppressing ambient noise using multiple audio signals

In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system.

Resolving buffer underflow/overflow in a digital system

An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

Suppressing noise in an audio signal

Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal.

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San Diego, CA, United States of America

Dinesh Ramakrishnan