Abstract: 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.

Abstract: 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.

Abstract: Methods and systems are for receiving network data representing a quality of a communication link in a network, the network data comprising a bandwidth value representing an available bandwidth for the communication link; detecting, based on the network data, that the available bandwidth has changed or will change from a first bandwidth level to a second bandwidth level; determining a probability that the network is congested; when the probability satisfies a threshold value, adjusting a bitrate for the communications link to a first value that enables network congestion to clear; when the probability does not satisfy the threshold value, adjusting the bitrate for the communications link to a second value that fully utilizes the second bandwidth level.

Abstract: Aspects of the subject technology may provide low latency audio for group communication sessions. Low latency audio may be provided, some examples, by an electronic device using a lowest audio block size that is lower than a lowest audio block size that is available to one or more other electronic devices in an group communication session.

Abstract: 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.

Abstract: 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.

Abstract: A method and apparatus of a device that manages a video stream is described. In an exemplary embodiment, the device receives a plurality of packets for a video stream from a transmitting device via a server. The device may additionally store a first packet of the plurality of packets in a first buffer when the first packet is on-time and store a second packet of the plurality of packets in a second buffer when the second packet is late. The device may also further forward a frame from the second buffer to the first buffer when frame is complete.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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 ? (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 ?, the playback speed can be reduced.

Abstract: 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 ? (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 ?, the playback speed can be reduced.

Abstract: 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.

Abstract: 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.

Abstract: 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.