Abstract: Video streams are generated using a combination of Multiple Bit Rate (MBR) encoding and Scalable Video Coding (SVC). Capabilities and requests of the clients are used in determining the video streams to generate as well as what video streams to deliver to the clients. The clients are placed into groups based on a resolution capability of the client. For each resolution grouping, MBR is used for generating spatial streams and SVC is used for generating temporal and quality streams.

Abstract: The generation and delivery of key frames to clients of a video conference are performed in response to a need for a synchronization point. Instead of automatically sending a key frame periodically to each of the clients in the video conference, a key frame is sent to one or more clients upon the occurrence of an event in the video conference. For example, a key frame may be sent to a client when the client joins the video conference. A key frame may also be sent to a client that has packet loss, upon the request of a client, a speaker change within the video conference, when a new stream is added by the client, and the like. The clients in the video conference that are not affected by the event continue to receive predicted frames.

Abstract: A LED drive circuit according to the present invention comprises a controller and a programmable signal. The controller generates a switching signal coupled to switch a magnetic device for generating an output current to drive a plurality of LEDs. The programmable signal is coupled to regulate a current-control signal of the controller. The switching signal is modulated in response to the current-control signal for regulating the output current, and the level of the output current is correlated to the current-control signal.

Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

Abstract: The invention includes several techniques and tools, which can be used in combination or separately. For example, an audio encoder can encode information directly using coding processes that include a windowed overlapped transform, a selective multi-channel transform, scalar quantization and entropy encoding. The audio encoder can also encode information parametrically according to a parametric compression mode that accounts for audibility of distortion according to an auditory model. A corresponding audio decoder can decode first information directly and second information according to the parametric mode.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: Video encoding computations are optimized by dynamically adjusting slice patterns of video frames based on complexity of each frame and allocating multi-core threading based on the slices. The complexity may be based on predefined parameters such as color, motion, and comparable ones for each slice. Allocation is determined based on capacity and queue of each processing core such that overall computation performance for video encoding is improved.

Abstract: Subscriptions in a video conference may be provided using multiple bit rate streams. A video conference server may receive video streams from each client in a video conference and may receive subscription requests from each client. The subscription requests may include requests to see video streams from specific other clients at a given resolution and/or frame rate. The video conference server may match up the received video streams with the subscription requests in order to send the subscribing clients their desired video streams. The server may also be able to request different versions of video streams from participants (e.g. different resolutions) and/or alter the video streams in order to better comply with the subscription request.

Abstract: Video conference rate matching may be provided. A video conference server may receive video source streams from clients on a video conference. The server may analyze each client's capabilities and choose a video stream to send to each client based on those capabilities. For example, a client capable of encoding and decoding a high definition video stream may provide three source video streams—a high definition stream, a medium resolution stream, and a low resolution stream. The server may send only the low resolution stream to a client with a low amount of available bandwidth. The server may send the medium resolution stream to another client with sufficient bandwidth for the high definition stream, but which lacks the ability to decode the high definition stream.

Abstract: An improved loss recovery method for coding streaming media classifies each data unit in the media stream as an independent data unit (I unit), a remotely predicted unit (R unit) or a predicted data unit (P unit). Each of these units is organized into independent segments having an I unit, multiple P units and R units interspersed among the P units. The beginning of each segment is the start of a random access point, while each R unit provides a loss recovery point that can be placed independently of the I unit. This approach separates the random access point from the loss recovery points provided by the R units, and makes the stream more impervious to data losses without substantially impacting coding efficiency. The most important data units are transmitted with the most reliability to ensure that the majority of the data received by the client is usable. The I units are the least sensitive to transmission losses because they are coded using only their own data.

Abstract: An improved loss recovery method for coding streaming media classifies each data unit in the media stream as an independent data unit (I unit), a remotely predicted unit (R unit) or a predicted data unit (P unit). Each of these units is organized into independent segments having an I unit, multiple P units and R units interspersed among the P units. The beginning of each segment is the start of a random access point, while each R unit provides a loss recovery point that can be placed independently of the I unit. This approach separates the random access point from the loss recovery points provided by the R units, and makes the stream more impervious to data losses without substantially impacting coding efficiency. The most important data units are transmitted with the most reliability to ensure that the majority of the data received by the client is usable. The I units are the least sensitive to transmission losses because they are coded using only their own data.

Abstract: An improved loss recovery method for coding streaming media classifies each data unit in the media stream as an independent data unit (I unit), a remotely predicted unit (R unit) or a predicted data unit (P unit). Each of these units is organized into independent segments having an I unit, multiple P units and R units interspersed among the P units. The beginning of each segment is the start of a random access point, while each R unit provides a loss recovery point that can be placed independently of the I unit. This approach separates the random access point from the loss recovery points provided by the R units, and makes the stream more impervious to data losses without substantially impacting coding efficiency. The most important data units are transmitted with the most reliability to ensure that the majority of the data received by the client is usable. The I units are the least sensitive to transmission losses because they are coded using only their own data.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: With intelligent differential quantization, a video codec intelligently quantizes video at differing strength levels within a frame, such as on a macroblock (MB) or a group of MB basis. This allows the codec to control bit usage on a finer granularity than a frame to meet hardware constraints, as well as providing perceptual optimization by coarsely quantizing unimportant regions, while finely quantizing important regions within a frame. The intelligent differential quantization uses motion information gathered from encoding and analysis of the video to classify the importance of different regions of the image, and quantizes the regions accordingly. In addition, the intelligent differential quantization include efficient signaling of information as to the differential quantization strengths in the compressed bit stream.

Abstract: An audio processing tool measures the quality of reconstructed audio data. For example, an audio encoder measures the quality of a block of reconstructed frequency coefficient data in a quantization loop. The invention includes several techniques and tools, which can be used in combination or separately. First, before measuring quality, the tool normalizes the block to account for variation in block sizes. Second, for the quality measurement, the tool processes the reconstructed data by critical bands, which can differ from the quantization bands used to compress the data. Third, the tool accounts for the masking effect of the reconstructed data, not just the masking effect of the original data. Fourth, the tool band weights the quality measurement, which can be used to account for noise substitution or band truncation. Finally, the tool changes quality measurement techniques depending on the channel coding mode.

Abstract: An audio processing tool measures the quality of reconstructed audio data. For example, an audio encoder measures the quality of a block of reconstructed frequency coefficient data in a quantization loop. The invention includes several techniques and tools, which can be used in combination or separately. First, before measuring quality, the tool normalizes the block to account for variation in block sizes. Second, for the quality measurement, the tool processes the reconstructed data by critical bands, which can differ from the quantization bands used to compress the data. Third, the tool accounts for the masking effect of the reconstructed data, not just the masking effect of the original data. Fourth, the tool band weights the quality measurement, which can be used to account for noise substitution or band truncation. Finally, the tool changes quality measurement techniques depending on the channel coding mode.

Abstract: A connector is disclosed that includes one or more pins allowing power to be provided to a connector board attached externally to the television set. In one embodiment, the connector has one or more USB pins to allow serial communication between the television set and the connector board. In another embodiment, the connector has one or more pins to allow communication of television setup information to the connector board. In yet another embodiment, the connector board may use the setup information provided from the television to perform audio processing and deliver enhanced audio sound to the speaker system associated with the television. The connector board may also wirelessly communicate with a personal computer, thereby coupling the personal computer to the television set.

Abstract: A transform coder adaptively configures window sizes for transform coding in a two-pass process to maximize coding efficiency, while achieving necessary time resolution to avoid pre-echo. In a first pass, the coder places small size windows over detected transient regions of an input signal in an open-loop window configuration process. In a second pass, the coder adjusts the window size configuration according to measurements of the achieved quality in a closed-loop window configuration process. Where quality measurement shows unacceptable quantization noise, the coder increases window size. Where pre-echo is detected, the coder reduces window size within coding bit rate constraints.

Abstract: Quality settings established by an encoder are adjusted based on information associated with regions of interest (“ROIs”). For example, quantization step sizes can be reduced (to improve quality) or increased (to reduce bit rate). ROIs can be identified and quality settings can be adjusted based on input received from a user interface. An overlap setting can be determined for a portion of a picture that corresponds to an ROI overlap area. For example, an overlap setting is chosen from step sizes corresponding to a first overlapping ROI and a second overlapping ROI, or from relative reductions in step size corresponding to the first ROI and the second ROI. ROIs can be parameterized by information (e.g., using data structures) that indicates spatial dimensions of the ROIs and quality adjustment information (e.g., dead zone information, step size information, and quantization mode information).

Abstract: An encoder compresses a set of still images and parameterizes (with effect parameters) additional images synthesized from the set of still images. A decoder decompresses the set of still images and synthesizes the additional images using the effect parameters.