Disclosed are a video stream playing method, a system, a terminal and a storage medium. The method includes following steps: receiving audio and video data sent by a server in real time and storing the received audio and video data in a buffer area; moving audio and video data firstly stored in the buffer area to an area to be decoded in real time; and extracting audio and video data firstly stored in the area to be decoded in real time, and decoding and playing the audio and video data firstly stored in the area to be decoded.

This invention concerns the transmitting and receiving of digital media packets, such as audio and video channels and lighting instructions. In particular, the invention concerns the transmitting and receiving of redundant media packet streams. Samples are extracted from a first and second media packet stream. The extracted samples are written to a buffer based on the output time of each sample. Extracted samples having the same output time are written to the same location in the buffer. Both media packet streams are simply processed all the way to the buffer without any particular knowledge that one of the packet streams is actually redundant. This simplifies the management of the redundant packet streams, such as eliminating the need for a “fail-over” switch and the concept of an “active stream”, The location is the storage space allocated to store one sample. The extracted sample written to the location may be written over another extracted sample from a different packet stream previously written to the location. These extracted samples written to the same location may be identical.

The present invention discloses a synchronized audio playing method, apparatus, and system for playing audio synchronously. By outputting designated data using a designated data buffer area before arrival of the starting time point of synchronous playing so as to make an audio link in a stable state, the method shuns from the impact of factors like system process scheduling interruption and system hardware on synchronous playing, thereby enhancing the precision of synchronous playing; moreover, upon arrival of the starting time point of synchronous playing, by computing a tuning amount of the audio data in the local synchronization buffer area, adjusting the audio data in the local synchronization buffer area according to the tuning amount, and meanwhile performing output playing on the tuned audio data, the method achieves audio synchronization play. Compared with the prior art, the present invention needn't add playing timestamp information to an audio data stream, and the playing terminal needn't process timestamp information during the entire playing procedure, thereby effectively increasing the flexibility of synchronous playing.

Provided are a method and device for audio post-processing. The method may comprise receiving, at a first processor, an audio signal, detecting, at the first processor, a plurality of post-processing modules for altering the audio signal, and creating, based on information identifying functions of the plurality of post-processing modules, an optimized acceleration module. It may further comprise sending, through the optimized acceleration module, a buffer packet of the audio signal along a single data path to a second processor and post-processing, at the second processor, the buffer packet of the audio signal through each of a plurality of associated post-processing modules that correspond to the post-processing modules on the first processor, controlling each associated post-processing module via control paths from each corresponding post-processing module, and receiving, at the first processor, a post-processed buffer packet of the audio signal via a single return data path.

A data processing method of a multimedia device is disclosed. The multimedia device pauses playback of multimedia data in response to a pause signal. The method includes: buffering the multimedia data before the pause signal is received to obtain prerecorded multimedia data; writing the prerecorded multimedia data into a storage unit in response to the pause signal; reading the prerecorded multimedia data from the storage unit in response to a playback signal; and playing the prerecorded multimedia data.

One embodiment of the present invention is a method for playing a portion of a media work which includes steps of: (a) playing the media work; (b) receiving input from a user; (c) analyzing parameters to determine the portion of the media work to play; (d) altering at least a part of the portion; and (e) playing the portion.

It is provided a method, performed by a MOS, Mean Opinion Score, estimator, for predicting a multimedia session MOS. The multimedia comprises a video and an audio, wherein video quality is represented by a list of per time unit scores of a video quality, an initial buffering event and rebuffering events in the video, and wherein audio quality is represented by a list of per time unit scores of audio quality. The method comprises: generating video features from the list of per time unit scores of the video quality; generating audio features from the list of per time unit scores of the audio quality; generating buffering features from the initial buffering event and rebuffering events in the video; and estimating a multimedia session MOS from the generated video features, generated audio features and generated buffering features by using machine learning technique.

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.

This invention concerns the transmitting and receiving of digital media packets, such as audio and video channels and lighting instructions. In particular, the invention concerns the transmitting and receiving of redundant media packet streams. Samples are extracted from a first and second media packet stream. The extracted samples are written to a buffer based on the output time of each sample. Extracted samples having the same output in time are written to the same location in the buffer. Both media packet streams are simply processed all the way to the bugger without any particular knowledge that one of the packet streams is actually redundant. This simplifies the management of the redundant packet streams, such as eliminating the need for a “fail-over” switch and the concept of an “active stream”, the location is the storage space allocated to store one sample. The extracted sample written to the location may be written over another extracted sample from a different packet stream previously written to the location. These extracted samples written to the same location may be identical.

The present disclosure provides systems and methods for adjusting the playback speed of accessories based on the buffer level of the received content. The accessories may receive content from a host device at a certain speed. The incoming speed of the content may be different than the playback speed of the content. Therefore, a buffer level of the received content may be determined using a linear least square (“LLS”) fit of the buffer level, the average of the buffer level, or a phased locked loop (“PLL”) approach. Based on a difference between the buffer level and the playback speed, a speed adjustment may be determined. Instructions may be transmitted from a primary accessory to a secondary accessory to adjust the playback speed by a certain amount at a certain time. This may ensure that the accessories remain in sync. The accessories may then adjust their respective playback speeds.