Apple Inc.

Aol Inc.

Facebook, Inc.

Oath Inc.

Other

Aol Llc, a Delaware Limited Liability Company

Aol Llc, a Deleware Limited Liability Company

Callahan Cellular LLC

David L Biderman

Roger N Pantos

William B May, Jr

Christopher M Garrido

Scott K Brown

James David Batson

Ming Jin

Alan Tseng

Stephen E Loomis

Gurtej Singh G Chandok

Karthick Santhanam

John Y Su

Luciano Miguel Verger

Patrick Miauton

David S Bill

Conor Patrick Cahill

Michael Lester Wise

Eric L Chien

David Wexelblat

David F Pare

Hsien-Po Shiang

Hsi-Jung Wu

Xiaosong Zhou

Ajoy K Singh

Dazhong Zhang

Najeeb M Abdulrahiman

Srinivasan Vasudevan

John Samuel Bushell

Christopher L Flick

Eric A Allamanche

Anil G Naik

Zhenheng Li

Deepak Chitlur Lakshman

Joshua L Hare

Xiaoxiao Zheng

Keith Peters

Komei Harada

Yeping Su

Hyeonkuk Jeong

Chris Yoochang Chung

Padmavathy Bhooma

Douglas Scott Price

Ian J Baird

Christopher L Oklota

Christoph Paasch

Erik Vladimir Ortega Gonzalez

Kyle William Horn

Gurtej Singh Chandok

Thomas Pepper

Qian Sun

David Alan Pound

Brian Thomas Toombs

Wenhui Xu

Jordan B Schneider

Gabriel Nava Marino

Jeffrey L Robbin

Taylor Gerard Carrigan

Daniel Ben Pollack

Eric Taylor Seymour

Yan Yang

Cahya Adiansyah Masputra

Darin Benjamin Adler

Keith Walter Rauenbuehler

Rainer Brodersen

Richard Michael Powell

Joseph M Williams

Thomas Michael Madden

Hang Yuan

Daniel C Klingler

Mohammed Zubair Visharam

Thomas P Devanneaux

Nicholas M Fraioli

Yang Yu

Adam E Kriegel

Jose A Lozano Hinojosa

Courtney Ann Kennedy

Bradley F Patterson

Sudeng Hu

Kevin Arthur Robertson

Vu H Chiem

Hassan A Shojania

Chieh Lu

Andrew M Dickson

Bess C Chan

Brajesh K Dave

Michel A Rynderman

Yichao Shen

Sasanka T Vemuri

Simon Ezriel Chaim Goldrei

John Kevin Calhoun

Shuang Liu

Simon Gibson

Jinbo Qiu

Ashwin Ramesh

Jinbo Qui

Igor Kobzar

Santosh Vamaraju

Eric H Zhang

Erik Vladimir Ortega Gonzales

Spencer E Lewson

A method performed by a first device, which includes performing an audio call with a second device by transmitting a microphone signal as an uplink signal and receiving a downlink signal for driving a first speaker and while performing the audio call, performing a joint media playback session in which both devices independently stream a piece of media content for synchronous playback such that both devices receive an audio signal of the piece of media content for driving respective speakers at the same time, determining that a voice activity detection (VAD) signal indicates that the downlink signal includes speech, in response to determining that the VAD signal indicates that the downlink signal includes speech, processing the audio signal of the piece of media content by applying a scalar gain, and driving the first speaker with a mix of the downlink signal and the audio signal.

Method and system for processing remote active speech during a call

The subject technology receives, at a local device, a requested camera format based on specifications of a display associated with a remote device. The remote device and the local device are devices participating in a video conference. The requested camera format includes a first resolution. Camera formats supported by a camera associated with the local device are determined. If a second resolution of a first camera format matches among the supported camera formats matches with the first resolution, the first camera format is selected for capturing the video stream by the camera. Otherwise, a second camera format among the supported camera formats is determined for capturing the video stream so as to maximize a field of view of the video stream relative to other camera formats supported by the camera.

Camera format selection

The embodiments described herein can dynamically adjust timing of network bandwidth estimations by adjusting a target frequency for sending probing sequences to one or more receivers of content. The receivers receive these probing sequences from a transmitter and respond to the transmitter with network bandwidth estimations. In one embodiment, the probing sequences can use the content itself to create the probing sequences. The embodiments can be used in video conferencing applications to control how a transmitter of content can adjust transmissions based upon the network bandwidth estimations. In one embodiment, a policy can be used at a transmitter to begin a video transmission with a high frequency target for sending probing sequences (with, e.g., smaller length probing sequences) and transition to a lower frequency target (with, e.g., longer length probing sequences) when network bandwidth deteriorates and then return to the higher frequency target when the network bandwidth improves.

High frequency probing for network bandwidth estimation using video data in real-time video conference

A method and apparatus of a device that manages a video telephony call is described. In an exemplary embodiment, the device receives a network event from a network service of a device. The device further determines that the network event that is due to a local disruption of a network component of the device. In addition, and in response to the determination, the device restricts a local dynamic control of the video telephony call.

Systems and methods of enhanced adaptability in real-time communications based on events and metrics from network interfaces

A method and apparatus of a device that uses a video jitter buffer to buffer the video frames for a received video stream is described. The device uses the video jitter buffer to estimate the delay variability of the frames and pick a target delay that will help harmonize the playback with minimal loss and delay. This is achieved by estimating the delay spread, which is the result of performing statistical analysis of the measured lags of the video frames received. The video jitter buffer provides target delay recommendation and reference frame information to the video player in order to anchor the playback and schedule the frames presentation time.

Systems and methods of video jitter estimation

Participant systems in an audiovisual (AV) conferencing can, in one embodiment, use methods to ensure that all participants have a common platform to support AV conferencing, such as a common codec, and also attempt to provide higher quality AV conferencing (e.g., better than the common codec) based on available bandwidth and other criteria. The participant systems can use the common codec as a fallback platform when bandwidth or other criteria dictate a reduction in the quality of the AV conferencing. Other embodiments are also disclosed.

Optimizing media experience in conferencing with diverse participants

In an example method, a system receives a plurality of frames of a video, and generates a data structure representing the video and representing a plurality of temporal layers. Generating the data structure includes: (i) determining a plurality of quality levels for presenting the video, where each of the quality levels corresponds to a different respective sampling period for sampling the frames of the video, (ii) assigning, based on the sampling periods, each of the frames to a respective one of the temporal layers of the data structure, and (iii) indicating, in the data structure, one or more relationships between (a) at least one the frames assigned to at least one of the temporal layers of the data structure, and (b) at least another one of the frames assigned to at least another one of the temporal layers of the data structure. Further, the system outputs the data structure.

Encoding and decoding video content

A video playback buffer adjustment system is disclosed. The video playback buffer has a target size, measured in units of time of content, and includes a compressed frames queue and an uncompressed frames queue. The uncompressed frames queue has a minimum threshold queue size. If the size of the uncompressed frames queue falls below the minimum threshold size, then a next encoded frame in the compressed frames queue is removed from the compressed frames queue, decoded, and enqueued in the uncompressed frames queue. If a current size of the playback queue exceeds the target size plus &#x394; (a hysteresis value), then a playback frame rate is increased, and a presentation time of one or more frames in the uncompressed frames queue is adjusted according to the increased playback frame rate. If the playback queue size is less than the target size minus &#x394;, the playback speed can be reduced.

Video playback buffer adjustment

Techniques to use an embedded passcode within an audio ringtone to establish a secure connection for arbitrary phone relay are described. The use of an embedded passcode enables encrypted ad-hoc connections for the relay of audio of an incoming telephone call to a secondary device, such as a virtual assistant enabled smart speaker device.

Embedded audio passcode in ringtone to establish secure connection for arbitrary phone relay

Methods and apparatuses for real-time or near real-time streaming of content using transfer protocols such as an HTTP compliant protocol. In one embodiment, a method includes dividing a stream of data, representing the contiguous time based content of a program (e.g. a live video broadcast), into a plurality of distinct media files, and generating a playlist file having a plurality of tags and Universal Resource Indicators (URIs) indicating an order of presentation of the plurality of distinct media files. The plurality of media files and the playlist file can be made available for transmission to a client device which can retrieve the media files using the playlist file.

Playlists for real-time or near real-time streaming

A device implementing a system for audio-video conferencing using multiple stream identifiers includes a processor configured to receive, from a sending device, indication of a first content stream and a first stream identifier, and indication of a second content stream and a second stream identifier associated. The first content stream and the second content stream correspond to different bit rates of streaming content. The processor is configured to receive, from a receiving device, a request to subscribe to the second content stream, the request including the second stream identifier, and receive, from the sending device, an indication that the second stream identifier has been associated with the first content stream. The processor is configured to forward, to the receiving device, the first content stream based on the request to subscribe to the second content stream and on the indication that the second stream identifier has been associated with the first content stream.

Multiway audio-video conferencing

Methods and apparatus to manage communication sessions to handover between a direct connection at a secondary wireless device and a relayed connection to the secondary wireless device via a primary wireless device. A connection manager of a secondary wireless device can trigger transfer of a communication session based on measurements of performance metrics for the communication session. Upon detection of performance degradation in a local connection or a backhaul connection or both, the connection manager of the secondary wireless device can determine proximity of and/or capabilities for connections of the primary wireless device and instigate transfer of the communication session between different connection types, such as between a direct connection and a relayed connection. The transfer of the communication session can occur without user intervention or in response to input from the user without interrupting or reestablishing the communication session.

Active communication session handover between direct and relayed connections

This document describes a protocol for transferring unbounded streams of multimedia data. It specifies the data format of the files and the actions to be taken by the server (sender) and the clients (receivers) of the streams. It describes version 2 of this protocol.

Real-time or near real-time streaming

Growing community of inventors

Los Gatos, CA, United States of America

David L Biderman