Abstract: A method of rate-control for a sequence of scalably coded images having transform coefficients partitioned into coding units coded in a plurality of quality increments having respective significance values. The method defines subsets each having one or more coding units, at least one image contributing at least one coding unit to two or more subsets. A list of requirements (LOR) is set having a least one entry associated with each subset. The significance values are used to select quality increments to construct an admissible codestream that satisfies the LOR on the subsets. The quality increments may be selected to achieve high quality for different subsets subject to size requirements in the LOR. For certain requirements, the codestream will also exhibit approximately constant reconstructed image quality. The quality increments may also be selected to achieve small compressed sizes for different subsets subject to quality requirements in the LOR.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: Neural networks provide efficient, robust and precise filtering techniques for compensating linear and non-linear distortion of an audio transducer such as a speaker, amplified broadcast antenna or perhaps a microphone. These techniques include both a method of characterizing the audio transducer to compute the inverse transfer functions and a method of implementing those inverse transfer functions for reproduction. The inverse transfer functions are preferably extracted using time domain calculations such as provided by linear and non-linear neural networks, which more accurately represent the properties of audio signals and the audio transducer than conventional frequency domain or modeling based approaches. Although the preferred approach is to compensate for both linear and non-linear distortion, the neural network filtering techniques may be applied independently.

Abstract: Scalable image compression is exploited to facilitate the creative process in the post-production of motion pictures. Specifically, digital intermediate (DI) processing of motion pictures is enabled by dynamically rendering proxies in response to client requests. A DI application is designed to enhance the efficiency of post-processing and the quality of the work product of the editors, colorists and other creative people. The DI application also provides a method for efficiently formatting the product for film, digital cinema, DVD and other video applications.

Abstract: The present invention provides a method of decoding two-channel matrix encoded audio to reconstruct multichannel audio that more closely approximates a discrete surround-sound presentation. This is accomplished by subband filtering the two-channel matrix encoded audio, mapping each of the subband signals into an expanded sound field to produce multichannel subband signals, and synthesizing those subband signals to reconstruct multichannel audio. By steering the subbands separately about an expanded sound field, various sounds can be simultaneously positioned about the sound field at different points allowing for more accurate placement and more distinct definition of each sound element.

Abstract: DTS Interactive provides low cost fully interactive immersive digital surround sound environment suitable for 3D gaming and other high fidelity audio applications, which can be configured to maintain compatibility with the existing infrastructure of Digital Surround Sound decoders. The component audio is stored and mixed in a compressed and simplified format that reduces memory requirements and processor utilization and increases the number of components that can be mixed without degrading audio quality. Techniques are also provided for “looping” compressed audio, which is an important and standard feature in gaming applications that manipulate PCM audio. In addition, decoder sync is ensured by transmitting frames of “silence” whenever mixed audio is not present either due to processing latency or the gaming application.