Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: An analysis/synthesis system uses existing analysis and synthesis filterbanks in an audio coding system to implement a phase shift filter that requires very little if any additional processing. One implementation using a single processing path can obtain a phase shift of either zero or ninety degrees. Another implementation that uses two processing paths can obtain a phase shift of essentially any desired angle.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.