Abstract: The present invention involves a system and method for optimizing video encoding. For each candidate motion vector, encoding distortion is determined between a macroblock and a reconstructed macroblock by determining discrete cosine transform coefficients of the macroblock and quantizing the discrete cosine transform coefficients. An estimate unit estimates the length of the bit stream that would be required to encode the quantized discrete cosine transform coefficients along with the mode information bits including mode and motion vector information. The reconstructed macroblock is determined based on the quantized discrete cosine transform coefficients. A bit-rate term based on the length of the bit-rate stream is determined and included in the encoding distortion. The candidate motion vector which minimizes the encoding distortion of the macroblock is chosen to be the motion vector for the macroblock.

Abstract: Two instruction executions circuits C1 and C2, possibly pipelined, share a write port to write instruction results to their destinations. When both circuits have results available for writing in the same clock cycle, the write port is given to circuit C1. Circuit C2 gets the write port only when there is a bubble in the write back stage of circuit C1. Circuit C2 executes instructions that occur infrequently in an average program. Examples are division, reciprocal square root, and power computation instructions. Circuit C1 executes instructions that occur more frequently. Circuits C1 and C2 are part of a functional unit of a VLIW processor.

Abstract: A method and apparatus for processing a digital data stream digital including a plurality of samples representing an audio signal. The plurality of samples are requantized. The plurality of requantized samples are filtered using a fast Fourier transform, wherein pulse code modulated samples are created from the plurality of requantized samples.

Abstract: A method and system for efficiently compressing a digital audio signal, wherein the digital audio signal includes a plurality of samples. Each of the samples are separated into a plurality of subbands. A signal to mask ratio (SMR) for each the subbands is predicted utilizing a model of relationships between energy values within each of the subbands and SMR values based on a predetermined psychoacoustic model. A number of are allocated bits in response to the predicted SMR and a preselected bit-rate. Then, each of the subbands are quantized based on the number of bits allocated, wherein the digital audio signal may be efficiently compressed.