Methods and systems are provided for adjusting a size of a buffer based on a probability that a rewind request will be received during a currently playing segment of media. When the system determines that receiving a rewind request is likely, the system will increase the size of a buffer such that the rewind request can be accommodated using data from the buffer. When the system determines that receiving a rewind request is unlikely, the system will decrease the size of the buffer to free resources for other system components.

Techniques have been developed to facilitate the livestreaming of group audiovisual performances. Audiovisual performances including vocal music are captured and coordinated with performances of other users in ways that can create compelling user and listener experiences. For example, in some cases or embodiments, duets with a host performer may be supported in a sing-with-the-artist style audiovisual livestream in which aspiring vocalists request or queue particular songs for a live radio show entertainment format. The developed techniques provide a communications latency-tolerant mechanism for synchronizing vocal performances captured at geographically-separated devices (e.g., at globally-distributed, but network-connected mobile phones or tablets or at audiovisual capture devices geographically separated from a live studio).

A system and method are provided for an AV device for use with a video player, one or more speakers, and encoded AV data. The encoded AV data includes multiplexed encoded video data and encoded audio data. The AV device is connected to the speakers via wireless channels. The AV device is able to determine channel delays associated with each wireless channel; synchronize program clocks of the video player and speakers; determine and modify buffer levels of each speaker; demultiplex the encoded AV data to obtain encode video data and encoded audio data; and provide prefetched portions of encoded audio data based on buffer levels.

Methods, systems, and apparatuses for content deliver, buffer management and synchronization are described herein. Content being played back on a playback platform is analyzed and based on the analysis, asynchronous playback of the content may be determined. A source buffer flush may be performed to correct the asynchronous playback. After the source buffer flush is performed, content segments may be written to the source buffer so that playback of the content on the playback platform resumes with synchronized audio and video.

In one aspect, a playback device is provided that is configured to identify a trigger event indicating a request to associate the playback device with another playback device. Based on identifying the trigger event, the playback device is configured to create a first sound code based on a first sound specimen detected by the playback device. After identifying the trigger event, the playback device is configured to receive from the other playback device a sound object and based on receiving the sound object, identify a second sound code. The playback device is also configured to, based on the first sound code and the second sound code, determine that it and the other playback device have a spatial relationship. Based on that determination, the playback device is configured to cause it and the other playback device to be associated in accordance with the indicated request.

The various implementations described herein include methods and systems for synchronous audio playback. In one aspect, a method is performed at each of a plurality of electronic devices, each having an audio system, an internal clock, processors and memory storing programs for execution by the processors. Each device is configured for two-way communications with a server and associated with a user account. The device receives an identification of a first device as a common clock device that has a first internal clock being designated as a master clock. The device receives a synchronized audio playback command that includes audio data to be output and a future playback time. In response to receiving the audio data, the device determines a synchronized audio playback time. If the determined synchronized audio playback time has not yet occurred, the electronic device outputs the audio data based on the determined synchronized audio playback time.

A method for playing a streaming media file and a display apparatus are provided. The method includes: receiving a command for replaying at least one portion of a streaming media file that has been played from a user; determining a first timestamp associating with replaying of the at least one portion of the streaming media file that has been played; calculating audio data capacity between the first timestamp and a second timestamp corresponding to a current play time point of the streaming media file when the command is received; in response to the audio data capacity being not greater than capacity of a cache space for a browser of a streaming application, sending audio data after the first timestamp in the cache space directly to a decoder for decoding without sending the audio data to a decryption module for decryption; outputting audio data decoded by the decoder to a loudspeaker.

A video audio transmission system, transmission method, sending device, and reception device capable of avoiding buffer overflow and buffer depletion in a decoding device and realizing Group of Pictures (GOP) synchronization in encoding devices by eliminating clock deviation among devices. In the video audio transmission system, the sending devices supply clocks generated from common time point information to cameras as genlock signals. The reception devices supply clocks generated from the common time point information to the decoding devices as genlock signals. Therefore, clock deviation between the devices can be eliminated, and the buffer overflow and the buffer depletion in the decoding device can be avoided. Frame periods of video signals output by a plurality of dispersed cameras can be aligned, and reliable GOP synchronization can be realized by the encoding devices on a latter stage with respect to the cameras.

A system and method provide an enhanced listening experience for a user of a radio receiver or other device that receives broadcast or streamed content having a plurality of program channels. Thu method and system buffer designated channels at the receiver and, when switching among channels, play back the buffered designated channels during reception from a selected point therein depending on the channel or other criteria. When switching to a different channel during reception, a user is able to hear content in a buffered program channel with music from the start of a song, for example, whereas content in a channel with news, talk radio, or live sports is played back from live reception, even though the content segments are transmitted at different start times relative to their selected times for playback following a channel change. User controls allow navigation among buffered designated channels during reception.

An electronic device and method are disclosed. The electronic device includes: a communication circuit for transceiving data, a charging circuit for receiving power via an electrical connection, and controlling a communication channel operative through the electrical connection, a memory, and a processor. The processor implements the method, including: receiving, via the charging circuit, and through the electrical connection, connection information of a first external electronic device for communicatively connecting to the first external electronic device, from a second external electronic device into which the electronic device is inserted and receiving power via the charging circuit, transmitting, via the charging circuit, and through the electrical connection, connection information of the electronic device for communicatively connecting to the first external electronic device, to the second external electronic device, and storing the connection information of the first external electronic device in the memory.