A video signal output apparatus includes: an outputter that outputs a video signal and a first audio signal to each of output apparatuses; a receiver that receives a second audio signal generated as a result of each of the output apparatuses performing audio signal processing on the first audio signal; an synchronization information obtainer that obtains synchronization information indicating a time difference between a time required for video signal processing and a time required for the audio signal processing by each of the output apparatuses; a delay calculator that calculates a delay time for reducing a time difference between respective timings of displaying video by the output apparatuses, using a timing at which the receiver receives the second audio signal from each of the output apparatuses and the synchronization information obtained; and a delayer that delays the video signal and the first audio signal, by the delay time calculated.

Novel techniques are described for lock-free sharing of a circular buffer. Embodiments can provide shared, lock-free, constant-bitrate access by multiple consumer systems to a live stream of audiovisual information being recorded to a circular buffer by a producer. For example, when a producer system writes a data stream to the circular buffer, the producer system records shared metadata. When a consumer system desires to begin reading from the shared buffer at a particular time, the shared metadata is used to compute a predicted write pointer location and corresponding dirty region around the write pointer at the desired read time. A read pointer of the consumer system can be set to avoid the dirty region, thereby permitting read access to a stable region of the circular buffer without relying on a buffer lock.

Systems and methods for loss protection of audio streams that may obtain an audio source such as from a microphone device, determine a sampling rate for encoding the audio source, and sampling the audio source to identify a plurality of samples (e.g., according to the sampling rate). A first audio output stream may be generated by encoding the plurality of samples. A second audio output stream may be generated by compressing each sample of the plurality of samples to collectively generate a second audio output stream. The first and second audio output streams may be transmitted (e.g., over a network subject to loss) to a recipient computer system that is able to utilize portions of the second audio output stream in place of lost portions first audio output stream.

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.

This invention concerns the transmitting and receiving of digital media packets, such as audio and video channels and lighting instructions. The network (104) is comprised of at least a transmitter device (110) and a receiving device (112). The controllers (122) and (126) of these devices handle the exchanging of configuration messages between the devices (110) and (112). Using the invention, the user is not required to manually configure the processor to receive media packet streams. Instead, a controller (126) of a receiving device (112) operates to receive information on a user selection of media channels and automatically configure the processor of the transmitter device. Further, the receiving device (112) is able to receive media channels using both unicast and multicast protocols. Media channels can be given textual labels which are unique on the unique (104) and easily identify to the user the actual source of the media channel. Media channels of different formats to be sent on the same network simultaneous. Further, redundant media channels are easily accommodated.

Method of recording a multimedia content broadcast streamwise, comprising the following steps: initialize a current throughput of recording of the stream in a mass memory; acquire a current portion of the multimedia content at the current recording throughput, and store it temporarily in the buffer memory; evaluate a rate of fill of the buffer memory; if the fill rate is greater than a predetermined high threshold of fill, decrease the current recording throughput; if the fill rate is less than or equal to a predetermined low threshold, increase the current recording throughput.

Techniques of playing back a looping video file involve providing multiple video codecs for decoding a video file. Each video codec performs its own decoding operation on the looping video file, resulting in multiple buffers of video frames and a buffer of audio frames. Then, as the GPU renders the ending video frames from a first buffer, it begins rendering the beginning video frames from a second buffer. In this way, the beginning of a next video cycle is already rendered for viewing by the time a current video cycle is ending.

Embodiments included herein generally relate to processing data samples. More particularly, embodiments relate to processing a plurality of data samples using a multi-agent ring-buffer and a plurality of agents configured to communicate with the ring-buffer to process each of the plurality of data samples.

A buffer size of an electronic media accessory, such as a wireless audio or video player, is dynamically adjusted based on machine learning optimizations balancing low latency and high media quality. A host device that is wirelessly paired with the electronic media accessory may detect various environmental conditions, and communicate a recommended buffer size to the electronic accessory based on such conditions. The electronic accessory adjusts its buffer in accordance with the recommended size, thereby achieving a latency of approximately 100 ms or less in good RF conditions.

A control device of media playback system receives a user command to initiate playback of video content and associated audio content and transmits a BLUETOOTH message based on the command to a video playback device. The video playback device receives the BLUETOOTH message, obtains the media content via a local area network (LAN), and determines an indication of when to begin playback of the media content. The video playback device transmits the indication to an audio playback device via the LAN and outputs the video content in lip-synchrony with playback of the corresponding audio content by the audio playback device. While outputting the video content in lip-synchrony with playback of the corresponding audio content by the audio playback device, the video playback device transmits timing information via the LAN to the audio playback device to maintain lip-synchrony with playback of the corresponding audio content by the audio playback device.