The disclosure relates to techniques for video source rate control for video telephony (VT) applications. The source video encoding rate may controlled using a dual-buffer based estimation of a frame budget that defines a number of encoding bits available for a frame of the video. The dual-buffer based estimation technique may track the fullness of a physical video buffer and the fullness of the virtual video buffer. The source video encoding rate is then controlled based on the resulting frame budget. The contents of the virtual buffer depend on constraints imposed by a target encoding rate, while the contents of the physical buffer depend on constraints imposed by varying channel conditions. Consideration of physical video buffer fullness permits the video source rate control technique to be channel-adaptive. Consideration of virtual video buffer fullness permits the video source rate control technique to avoid encoding excessive video that could overwhelm the channel.

Video source rate control for video telephony

Methods and systems for a content server to select sets of video streams having different encoding parameters for transmitting the sets of video streams to a media device are disclosed herein. In some embodiments, a method for transmitting video streams for a media program from a server to a media device includes: selecting, by the server, first encoding parameters including a first bitrate for a first set of video streams for the media program based on a first estimated bandwidth capacity for a network linking the server and the media device, transmitting the first set of video streams from the server to the media device, determining, by the server, second encoding parameters including a second bitrate for a second set of video streams for the media program, and transmitting the second set of video streams from the server to the media device.

Methods and devices for efficient adaptive bitrate streaming

Embodiments described herein relate to providing adaptive encoding of real-time information in packet-switched wireless communication systems. In an embodiment, a rate-adaptation unit may be configured to receive local as well as end-to-end feedback information associated with data transmission (such as data delay, packet loss, transmit power headroom, channel condition, sector loading, the amount of buffered data, etc.) from a wireless access module in communication with wireless/wired networks, and adapt the real-time information encoding in accordance with such feedback information.

Methods and systems for adaptive encoding of real-time information in packet-switched wireless communication systems

A display encoding system is disclosed. The display encoding system includes at least one processor, an encoding circuit, the encoding circuit having a structure separate from the structure of the processor(s), a communication connection to a computer network, a drawing memory, a traffic manager, and a memory access circuit. The processor(s) are configured to execute drawing commands, access the drawing memory via the memory access circuit to store image representation(s) in the drawing memory. The encoding circuit is configured to access the drawing memory to encode at least a portion of the image representation(s) stored in the drawing memory based on the executed drawing commands. The traffic manager is configured to transmit the encoded image over the communication connection to at least one display device.

Methods and apparatus for encoding a shared drawing memory

The disclosure relates to techniques for video packet shaping for video telephony (VT). The techniques can be used to prioritize audio packets to reduce audio delay. Channel conditions, excessive video content, or both can cause delays in audio transmission. When reverse link (RL) throughput is reduced, video packet size can overwhelm the RL and increase audio delay. The video packet may consume an excessive number of MAC RLP packets, resulting in delays between successive audio packets. The size of each video packet is adjusted so that audio packets are prioritized for transmission without substantial delay. The video packet size may be controlled based on channel conditions. The audio can be conveyed without substantial delay, even though the video may suffer from delay due to channel conditions. Although video may be compromised by channel conditions, video packet shaping ensures that the VT parties are able to smoothly carry on verbal conversation

Video packet shaping for video telephony

Adaptive encoding of digital multimedia information may be performed by measuring link parameters, such as a received signal strength, a bit error rate, or a rate of received acknowledgement signals, in order to determine an available transmission rate. A maximum encoding rate may then be determined based on the available transmission rate by, for example, dividing the available transmission rate by an overhead factor. If the encoding rate of the digital multimedia information exceeds the calculated maximum encoding rate, adaptive encoding of the digital multimedia information may be performed in order to conform the encoding rate of the digital multimedia information to the calculated maximum encoding rate. This process may involve compressing selected frames within a frame sequence, deleting high frequency components within selected frames, deleting I-frame components within selected frames, or mapping values within selected frames to corresponding values having coarser quantization.

Hartmut Wiesenthal

Papers

Adaptive encoding of digital multimedia information