Abstract: A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point.

Abstract: An audio spatial environment engine for converting from an N channel audio system to an M channel audio system, where N is an integer greater than M, is provided. The audio spatial environment engine includes one or more correlators receiving two of the N channels of audio data and eliminating delays between the channels that are irrelevant to an average human listener. One or more Hilbert transform systems each perform a Hilbert transform on one or more of the correlated channels of audio data. One or more summers receive at least one of the correlated channels of audio data and at least one of the Hilbert transformed correlated channels of audio data and generate one of the M channels of audio data.

Abstract: There are provided methods and an apparatus for processing audio signals. According to one aspect of the present invention there is included a method for processing audio signals having the steps of receiving at least one audio signal having at least a center channel signal, a right side channel signal, and a left side channel signal; processing the right and left side channel signals with a first virtualizer processor, thereby creating a right virtualized channel signal and a left virtualized channel signal; processing the center channel signal with a spatial extensor to produce distinct right and left outputs, thereby expanding the center channel with a pseudo-stereo effect; and summing the right and left outputs with the right and left virtualized channel signals to produce at least one modified side channel output.

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: 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: 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: A data transmission device includes a memory cache table (4) composed of a DRAM memory, a standard 2.5? hard disk (5), a control CPU (7), a FPGA (6) (or ASIC), a disk interface (3) and a backup battery. The device is unitized in the same external shape as a standard 3.5? hard disk and connected to the external computer (2) via the disk interface (3), and the FPGA (6) (or ASIC) manages the memory in the memory cache table (4) based on control actions by the CPU (7).

Abstract: A method of mixing audio channels is effective at rebalancing the audio without introducing unwanted artifacts or overly softening the discrete presentation of the original audio. This is accomplished between any two or more input channels by processing the audio channels to generate one or more “correlated” audio signals for each pair of input channels. The in-phase correlated signal representing content in both channels that is the same or very similar with little or no phase or time delay is mixed with the input channels. The present approach may also generate an out-of-phase correlated signal (same or similar signals with appreciable time or phase delay) that is typically discarded and a pair of independent signals (signals not present in the other input channel) that may be mixed with the input channels. The provision of both the in-phase correlated signal and the pair of independent signals makes the present approach well suited for the downmixing of audio channels as well.

Abstract: A multichannel audio format provides a truly discrete as well as a backward compatible mix for surround-sound, front or other discrete audio channels in cinema, home theater, or music environments. The additional discrete audio signals are mixed with the existing discrete audio channels into a predetermined format such as the 5.1 audio format. In addition these additional discrete audio channels are encoded and appended to the predetermined format as extension bits in the bitstream. The existing base of multichannel decoders can be used in combination with a mix decoder to reproduce truly discrete N.1 multichannel audio.

Abstract: The invention relates to a map panning directional indicator for a navigation system. A display device of the navigation system displays the map panning indicator, which indicates both a current heading between a map panning cursor and a current vehicle location relative to a map database and a distance between the map panning cursor and the current vehicle location. The map panning cursor is displayed in the center of a map section as a user pans over map sections of the map database, which are displayed on the display device.

Abstract: A blast supression device for use in explosive hardening in a relatively confined enclosed area. The device absorbs and dissipates the explosive force by utilizing a containment frame covered with overlapping multiple flexible resilient flaps that dissipate the explosive force by yielding during the blast within the device.