Qualcomm Incorporated

Intel Corporation

Via Telecom Co., Ltd.

Sanghyun Chi

Meoung-Jin Lim

Erik Visser

Lae-Hoon Kim

Shankar Thagadur Shivappa

S M Akramus Salehin

Soumya Das

Arungundram C Mahendran

Song Wang

Shuhua Zhang

Karthika Paladugu

Yinyi Guo

Dinesh Ramakrishnan

Soham Sheth

Isaac Garcia Munoz

Nils Gunther Peters

Vasudev Nayak

Hannes Pessentheiner

Wendeline Ruth Avis

Krishna Kumar Malthakal

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

An apparatus includes one or more processors configured to receive orientation data and to select, based on the orientation data, a particular filter from among multiple filters. The one or more processors are configured to perform signal processing operations associated with three-dimensional (3D) sound data based on the particular filter.

3D sound orientation adaptability

A device includes one or more processors configured to obtain audio signals representing sound captured by at least three microphones and determine spatial audio data based on the audio signals. The one or more processors are further configured to determine a metric indicative of wind noise in the audio signals. The metric is based on a comparison of a first value and a second value. The first value corresponds to an aggregate signal based on the spatial audio data, and the second value corresponds to a differential signal based on the spatial audio data.

Spatial audio wind noise detection

In an aspect, a lens is zoomed in to create a zoomed lens. Lens data associated with the lens includes a direction of the lens relative to an object in a field-of-view of the zoomed lens and a magnification of the object resulting from the zoomed lens. An array of microphones capture audio signals including audio produced by the object and interference produced by other objects. The audio signals are processed to identify a directional component associated with the audio produced by the object and three orthogonal components associated with the interference produced by the other objects. Stereo beamforming is used to increase a magnitude of the directional component (relative to the interference) while retaining a binaural nature of the audio signals. The increase in magnitude of the directional component is based on an amount of the magnification provided by the zoomed lens to the object.

Spatial audio zoom

Systems and methods are described herein for providing a virtual conference using a master device implemented with a personal communication device (PCD), including determining, by the master device, a latency for each of a plurality of satellite devices connected to the master device. The master device then determines an uplink buffer duration based on a difference between a highest latency and a lowest latency among the plurality of satellite devices. The master device determines a processing time for an uplink data packet, the processing time being determined based, at least in part, on the uplink buffer duration. The master device then performs signal processing at the processing time for the received uplink data packets.

Methods and systems for virtual conference system using personal communication devices

Systems and methods are described herein for providing a virtual conference with a master device connected to a plurality of satellite devices, including: receiving, by the master device, uplink data packets from a plurality of channels, each of the plurality of channels is associated with one of the plurality of satellite devices, and dividing, by the master device, the plurality of channels into two or more groups based on a conversation captured in the uplink data packets of each of the plurality of channels. The master device selects one group from the two or more groups for output. The master device also transmits downlink data packets corresponding to the selected group for the plurality of satellite devices.

A method for speech restoration by an electronic device is described. The method includes obtaining a noisy speech signal. The method also includes suppressing noise in the noisy speech signal to produce a noise-suppressed speech signal. The noise-suppressed speech signal has a bandwidth that includes at least three subbands. The method further includes iteratively restoring each of the at least three subbands. Each of the at least three subbands is restored based on all previously restored subbands of the at least three subbands.

Systems and methods for speech restoration

An acoustic processing apparatus is provided. The apparatus includes a pre-processing component, a filter and a first signal processing component. The pre-processing component compensates a non-linearity of a reference signal to generate an input signal. The filter coupled to the pre-processing component, the filter executes filtering on the input signal to generate an output signal. The first signal processing component, coupled to the pre-processing component, the reference signal obtains a gain from the first signal processing component to generate a first signal, and the first signal processing component passes the gain to the pre-processing component.

Pre-process (amplitude distortion) and post-process (phase synchronization) for linear AEC system

An automatic audio recovery system in a mobile device is provided. The system includes one or more status generators and an audio path recovery handler. The one or more status generators are configured to monitor operation of audio path elements within the mobile device, and are configured to detect and report one or more abnormal audio operations to an audio status monitor, where, responsive to one or more reports corresponding to the one or more abnormal audio operations, the audio status monitor requests automatic recovery of the audio path elements. The audio path recovery handler is configured to receive an automatic recovery request from the audio status monitor, and is configured to responsively direct reset elements corresponding to the audio path elements to automatically initialize the audio path elements to a stable state.

Apparatus and method for automatic audio system and recovery from unexpected behaviors

An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.

Acoustic shock protection device and method thereof

A method for detecting a double-talk condition and an echo cancellation system using the same are introduced herein. According to an exemplary embodiment, the method may be adapted to an double-talk detector of an echo cancellation system and includes the following steps: obtaining an input signal with a first power; computing a second power, wherein the second power is a power combination of at least one estimated signal; determining whether the double-talk condition occurs according to a relationship between the first power and the second power, when the first power is larger than the second power, the double-talk condition occurs.

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San Diego, CA, United States of America

Sanghyun Chi