Abstract: Cooperative interaction of a message processor and an intermediation processor. The message processor processes messages received from a network and to be sent over a network. The intermediation processor intermediates between the message processor and the network using a shared memory that includes a memory portion that is within the address space of the message processor. The message processor writes messages into the memory portion over a physical channel and reads message from the memory portion over the physical channel. The intermediation processor provides messages received from a network into the memory portion so that the received messages can be read by the message processor over the physical channel. The intermediation processor dispatches messages written to the memory portion by the message processor over the network.

Abstract: Cooperative interaction of a message processor and an intermediation processor. The message processor processes messages received from a network and to be sent over a network. The intermediation processor intermediates between the message processor and the network using a shared memory that includes a memory portion that is within the address space of the message processor. The message processor writes messages into the memory portion over a physical channel and reads message from the memory portion over the physical channel. The intermediation processor provides messages received from a network into the memory portion so that the received messages can be read by the message processor over the physical channel. The intermediation processor dispatches messages written to the memory portion by the message processor over the network.

Abstract: The present disclosure relates to playback of video/audio streaming media data. The media stream is available from the network at multiple bit rates. When a seek operation is performed, a playback device requests a lower bit rate media stream in order to quickly fill a playback buffer so that playback can commence more readily. After a seek start-up period is complete, the playback device can return to downloading the media stream associated with higher bit rates in order to increase the quality of the playback.

Abstract: Estimation of available bandwidth on a network uses packet pairs and spatially filtering. Packet pairs are transmitted over the network. The dispersion of the packet pairs is used to generate samples of the available bandwidth, which are then classified into bins to generate a histogram. The bins can have uniform bin widths, and the histogram data can be aged so that older samples are given less weight in the estimation. The histogram data is then spatially filtered. Kernel density algorithms can be used to spatially filter the histogram data. The network available bandwidth is estimated using the spatially filtered histogram data. Alternatively, the spatially filtered histogram data can be temporally filtered before the available bandwidth is estimated.

Abstract: The present disclosure relates to playback of video/audio streaming media data. The media stream is available from the network at multiple bit rates. When a seek operation is performed, a playback device requests a lower bit rate media stream in order to quickly fill a playback buffer so that playback can commence more readily. After a seek start-up period is complete, the playback device can return to downloading the media stream associated with higher bit rates in order to increase the quality of the playback.

Abstract: Estimation of available bandwidth on a network uses packet pairs and spatially filtering. Packet pairs are transmitted over the network. The dispersion of the packet pairs is used to generate samples of the available bandwidth, which are then classified into bins to generate a histogram. The bins can have uniform bin widths, and the histogram data can be aged so that older samples are given less weight in the estimation. The histogram data is then spatially filtered. Kernel density algorithms can be used to spatially filter the histogram data. The network available bandwidth is estimated using the spatially filtered histogram data. Alternatively, the spatially filtered histogram data can be temporally filtered before the available bandwidth is estimated.

Abstract: Estimation of available bandwidth on a network uses packet pairs and spatially filtering. Packet pairs are transmitted over the network. The dispersion of the packet pairs is used to generate samples of the available bandwidth, which are then classified into bins to generate a histogram. The bins can have uniform bin widths, and the histogram data can be aged so that older samples are given less weight in the estimation. The histogram data is then spatially filtered. Kernel density algorithms can be used to spatially filter the histogram data. The network available bandwidth is estimated using the spatially filtered histogram data. Alternatively, the spatially filtered histogram data can be temporally filtered before the available bandwidth is estimated.

Abstract: The present disclosure relates to playback of video/audio streaming media data to provide a glitch-free experience. The system adapts the media stream to the user connection in order to provide the glitch-free experience. Stream selection can be made using a heuristics module located on the playback device that analyzes various aspects of the playback to make intelligent decisions about which media stream to download from a network.

Abstract: Systems and/or methods (“tools”) are described that provide a buffer passing mechanism, and other techniques. Some of these tools may be enabled using a computer-implemented data structure that includes a plurality of master buffers, and a plurality of slave buffers that store data elements. Some of the slave buffers are referenced by more than one master buffer. Some of the slave buffers are referenced by one of the master buffers, but not by another master buffer.

Abstract: A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple corrections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.

Abstract: The present disclosure relates to playback of video/audio streaming media data to provide a glitch-free experience. The system adapts the media stream to the user connection in order to provide the glitch-free experience. Stream selection can be made using a heuristics module located on the playback device that analyzes various aspects of the playback to make intelligent decisions about which media stream to download from a network.

Abstract: The present disclosure relates to playback of video/audio streaming media data. The media stream is available from the network at multiple bit rates. When a seek operation is performed, a playback device requests a lower bit rate media stream in order to quickly fill a playback buffer so that playback can commence more readily. After a seek start-up period is complete, the playback device can return to downloading the media stream associated with higher bit rates in order to increase the quality of the playback.

Abstract: A method and system for dynamically altering the transmission settings of one or more computing devices engaged in a real-time communication session is presented. The devices exchange meaningful and dummy control packets according to a standard control protocol. The approximate bandwidth available on the network is then calculated based on the difference in arrival times between at least one of the dummy control packets and at least one of the meaningful control packets. Once the approximate bandwidth available on the network is computed, the one or more devices adjust outgoing audio and video data streams using a quality control mechanism. The quality control mechanism enables the one or more devices to transmit data in a way that maximizes the user experience during the real-time communication session.

Abstract: A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple corrections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.

Abstract: A technique for reducing information reception delays is provided. The technique reduces delays that may be caused by protocols that guarantee order and delivery, such as TCP/IP. The technique creates multiple connections between a sender and recipient computing devices and sends messages from the sender to the recipient on the multiple connections redundantly. The recipient can then use the first arriving message and ignore the subsequently arriving redundant messages. The recipient can also wait for a period of time before determining which of the arrived messages to use. The technique may dynamically add connections if messages are not consistently received in a timely manner on multiple connections. Conversely, the technique may remove connections if messages are consistently received in a timely manner on multiple connections. The technique can accordingly be used with applications that are intolerant of data reception delays such as Voice over IP, real-time streaming audio, or real-time streaming video.

Abstract: Systems and/or methods (“tools”) are described that provide a buffer passing mechanism, and other techniques. Some of these tools may be enabled using a computer-implemented data structure that includes a plurality of master buffers, and a plurality of slave buffers that store data elements. Some of the slave buffers are referenced by more than one master buffer. Some of the slave buffers are referenced by one of the master buffers, but not by another master buffer.

Abstract: Encoded signal reuse implementations are described. In one implementation, an already encoded signal may be selected, in a favor of encoding a signal, if a parameter associated with the already encoded signal substantially matches an evaluation parameter used to select an encoding procedure. The implementations may also enable selection of an already encoded signal if encoding a signal would produce an encoded signal that is substantially the same as the already encoded signal.

Abstract: A codec selection process is independent of a codec source. A codec table may be used that includes indexed codec references related to codecs stored in a codec source. The codec table and the codec source may be modified without affecting a codec selection process. This feature of an exemplary implementation makes it fairly straightforward to add, change, or otherwise modify codecs stored in the codec source and referenced in the codec table without having to change the codec selection process.

Abstract: A method and system for dynamically altering the transmission settings of one or more computing devices engaged in a real-time communication session is presented. The devices exchange meaningful and dummy control packets according to a standard control protocol. The approximate bandwidth available on the network is then calculated based on the difference in arrival times between at least one of the dummy control packets and at least one of the meaningful control packets. Once the approximate bandwidth available on the network is computed, the one or more devices adjust outgoing audio and video data streams using a quality control mechanism. The quality control mechanism enables the one or more devices to transmit data in a way that maximizes the user experience during the real-time communication session.

Abstract: A method and system for dynamically altering the transmission settings of one or more computing devices engaged in a real-time communication session is presented. The devices exchange meaningful and dummy control packets according to a standard control protocol. The approximate bandwidth available on the network is then calculated based on the difference in arrival times between at least one of the dummy control packets and at least one of the meaningful control packets. Once the approximate bandwidth available on the network is computed, the one or more devices adjust outgoing audio and video data streams using a quality control mechanism. The quality control mechanism enables the one or more devices to transmit data in a way that maximizes the user experience during the real-time communication session.