Dolby Laboratories Licensing Corporation

Stephen Decker Vernon

Mark Franklin Davis

Robert Loring Andersen

Michael Mead Truman

Grant Allen Davidson

Louis Dunn Fielder

Robin Atkins

Brett Graham Crockett

Charles Quito Robinson

Michael John Smithers

Craig C Todd

Kenneth James Gundry

James W Fosgate

Dossym Nurmukhanov

Matthew Aubrey Watson

Ian Eric Esten

Joel Robert Kustka

Todd Jeffrey Heller Hager

Philip Anthony Williams

Gabriel Grant Cory

Sophia Genung Poirier

Methods for mapping an image from one dynamic range to another dynamic range are presented. The mapping is based on a sigmoid-like function determined by three anchor points. The first anchor point is determined using the black point levels of a reference and a target display, the second anchor point is determined using the white point levels of the reference and target displays, and the third anchor point is determined using mid-tones information data for the input image and the mid-tones level of the target display. The mid-tones level of the target display is computed adaptively based on a desired range of constant brightness and a mapping function. The mapping function mapping may be piece-wise linear (in the log domain), piece-wise continuous, or a sigmoid. Examples for mapping high dynamic range images to standard dynamic range displays are presented.

Display mapping for high dynamic range images

Control data templates are generated independent of a plurality of audio elements based on user input. The user input relates to parameter values and control inputs for operations. In response to receiving audio elements after the control data templates are generated, audio objects are generated to store audio sample data representing the audio elements. Control data is generated based on the parameter values and the control inputs for the operations in the control data templates. The control data specifies the operations to be performed while rendering the audio objects. The control data is then stored separately from the audio sample data in the audio objects. The audio objects can be communicated to downstream recipient devices for rendering and/or remixing.

Interactive audio content generation, delivery, playback and sharing

A process that allocates bits for quantizing spectral components in a perceptual coding system is performed more efficiently by obtaining an accurate estimate of the optimal value for one or more coding parameters that are used in the bit allocation process. In one implementation for a perceptual audio coding system, an accurate estimate of an offset from a calculated psychoacoustic masking curve is derived by selecting an initial value for the offset, calculating the number of bits that would be allocated if the initial offset were used for coding, and estimating the optimum value of the offset from a difference between this calculated number and the number of bits that are actually available for allocation.

Reduced computational complexity of bit allocation for perceptual coding

A method of modifying the operation of the encoder function and/or the decoder function of a perceptual coding system in accordance with supplemental information, such as a watermark, so that the supplemental information may be detectable in the output of the decoder function. One or more parameters are modulated in the encoder function and/or the decoder function in response to the supplemental information.

Modulating one or more parameters of an audio or video perceptual coding system in response to supplemental information

Audio sources in typical computer systems provide different numbers of channels of audio signals to a mixing component of the operating system. This conventional arrangement usually prevents the audio signals from all sources from being played back through all output channels. Novel arrangements of upmixing and mixing components are disclosed that allow audio signals to be delivered to all output channels regardless of the configuration of the audio sources and the number of channels that are provided by those audio sources.

Method for expanding an audio mix to fill all available output channels

An audio encoder discards spectral components of an input signal and uses channel coupling to reduce the information capacity requirements of an encoded signal. Channel coupling represents selected spectral components of multiple channels of signals in a composite form. An audio decoder synthesizes spectral components to replace the discarded spectral components and generates spectral components for individual channel signals from the coupled-channel signal. The encoder provides scale factors in the encoded signal that improve the efficiency of the decoder to generate output signals that substantially preserve the spectral energy of the original input signals.

Audio coding systems and methods using spectral component coupling and spectral component regeneration

A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent direction lying on a second axis, the first and second aces being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom, The processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three output signals are produced by matrix multiplying the two input signals by the matrix coefficients.

Method for apparatus for audio matrix decoding

Codeword-position-caused encoder latency is reduced by avoiding the requirement for knowledge of the message prior to generating an error detecting or concealing codeword associated with the message. A pseudo error detecting or concealing codeword is inserted in place of the normal error detecting or concealing codeword appropriate for the segment of information to which the error detecting or concealing codeword relates. In order to satisfy the requirement of conventional decoders, the pseudo error detecting or concealing information must match or be appropriate for the segment so that the decoder sees the codeword and message segment as valid or error free. This is accomplished by modifying or perturbing at least a portion of the segment to which the pseudo codeword relates. The invention is particularly useful for maintaining the backward compatibility of audio data encoding formats in which the minimum latency is too long (e.g., computer games, where the player performs some operation leading to a sound, and that sound must not be perceptibly delayed with respect to the operation).

Low latency data encoder

Scalable coding of audio into a core layer in response to a desired noise spectrum established according to psychoacoustic principles supports coding augmentation data into augmentation layers in response to various criteria including offset of such desired noise spectrum. Compatible decoding provides a plurality of decoded resolutions from a single signal. Coding is preferably performed on subband signals generated according to spectral transform, quadrature mirror filtering, or other conventional processing of audio input. A scalable data structure for audio transmission includes core and augmentation layers, the former for carrying a first coding of an audio signal that places post decode noise beneath a desired noise spectrum, the later for carrying offset data regarding the desired noise spectrum and data about coding of the audio signal that places post decode noise beneath the desired noise spectrum shifted by the offset data.

Scalable coding method for high quality audio

Any of several information processing techniques may be used in various information storage and transmission applications to prevent the occurrence of certain &ldquo;forbidden&rdquo; bit patterns. According to an encoding technique, a reversible coding process is used to generate an encoded representation of an information stream that cannot contain any forbidden data patterns. This may be accomplished by partitioning the information stream into segments and encoding each segment according to a respective encoding key that is selected such that the results of the coding process cannot contain a forbidden data pattern. According to one substitution technique, all occurrences of forbidden data patterns are replaced with permissible data patterns that do not otherwise occur in the information stream. This may be accomplished by partitioning the information stream into segments, identifying an unused data pattern in a respective segment, and carrying out the replacement of all occurrences of the forbidden data pattern in that segment. According to another substitution technique, all occurrences of a forbidden data pattern are replaced by any permissible data pattern. This may be accomplished by partitioning the information stream into segments, identifying occurrences of the substitution data pattern and the forbidden data pattern in a respective segment, constructing a flag for each occurrence, and replacing all occurrences of the forbidden data pattern in that segment with the substitution data pattern. Decoding keys, substitution data patterns, substitution flags or any other information needed to recover the original information is assembled with the modified information in a form that does not equal the forbidden data pattern.

Avoiding forbidden data patterns in coded audio data

Accurate measures of estimated spectral power and phase are derived from two spectral representations of each of one or more input signals generated by single-sideband filter banks including critically-sampled perfect-reconstruction filter banks implemented by generalized forms of Time Domain Aliasing Cancellation transforms. The derived measures of spectral magnitude and phase may be used in various applications including adaptive bit allocation in perceptual coding systems. An efficient implementation of one particular single-sideband critically-sampled filter bank comprises folding and shuffling blocks of real-valued input signal samples into blocks of complex-valued samples, rotating the complex-valued samples in the complex plane, transforming the rotated samples into transform coefficients, and rotating and shuffling the transform coefficients to form the spectral representation.

Method and apparatus for efficient implementation of single-sideband

The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to the suppression of quantizing noise which results from low bit-rate encoding and decoding of signal events such as transients. An encoder identifies intervals of an input signal in which a decoder is likely to generate audible artifacts caused by quantizing errors. Indications of these intervals are passed in the encoded signal to the decoder. The decoder uses these indications to eliminate quantizing artifacts by suppressing corresponding intervals in the output signal.

Quantization noise suppression for encoder/decoder system

In a system in which a low-bit rate encoder and decoder carries matrixed audio signals, an adaptive rematrix rematrixes matrixed signals from an unmodified 4:2 matrix encoder to separate and isolate quiet components from loud ones, thereby avoiding the corruption of quiet signals with the low-bit-rate coding quantization noise of loud signals. The decoder is similarly equipped with a rematrix, which tracks the encoder rematrix and restores the signals to the form required by the unmodified 2:4 matrix decoder. The encoder adaptive rematrix selects the matrix output signals or the amplitude weighted sum and difference of the matrix output signals. The choice of whether the matrix output signals or the sum and difference of the matrix output signals are selected is based on a determination of which results in fewer undesirable artifacts when the output audio signals are recovered in the decoder. The adaptive rematrix may operate on frequency component representations of signals rather than the time-domain signals themselves.

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Hillsborough, CA, United States of America

Stephen Decker Vernon