Dts, Inc.

Other

University of Washington

Jeffrey Kenneth Thompson

Aaron Kube Warner

Robert W Reams

Zoran Fejzo

Brandon John Smith

Jean-Marc Jot

Antonius A Kalker

Elias Nemer

Les Atlas

Antonius Kalker

Aaron Warner

Systems, devices, and methods are described herein for adjusting a relationship between dialog and non-dialog signals in an audio program. In an example, information about a long-term dialog balance for an audio program can be received. The long-term loudness dialog balance can indicate a dialog-to-non-dialog loudness relationship of the audio program. A dialog loudness preference can be received, such as from a user, from a database, or from another source. A desired long-term gain or attenuation can be determined according to a difference between the received long-term dialog balance for the audio program and the received dialog balance preference. The long-term gain or attenuation can be applied to at least one of the dialog signal and the non-dialog signal of the audio program to render an audio program that is enhanced according to the loudness preference.

Dialog audio signal balancing in an object-based audio program

A transform-based codec and method with energy smoothing for mitigating vector quantization errors (such as 'birdies') during the encoding process. Embodiments of the codec and method use an encoder to apply in combination an orthogonal transformation and a vector permutation to frequency transform coefficients. In some embodiments the transformation is performed first followed by the permutation and in other embodiments the order is reversed. The order used is reversed at the decoder. A smoothing parameter containing the level of energy smoothing to be applied is passed from the encoder to the decoder and used by both to compute a transform matrix and an inverse transform matrix. In some embodiments the transform matrix is a fraction Hadamard matrix that is invertible, energy preserving, controllable, and stable.

Transform-based audio codec and method with subband energy smoothing

A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.

Matrix decoder with constant-power pairwise panning

A multiplet-based spatial matrixing codec and method for reducing channel counts (and thus bitrates) of high-channel count (seven or more channels) multichannel audio, optimizing audio quality by enabling tradeoffs between spatial accuracy and basic audio quality, and converting audio signal formats to playback environment configurations. An initial N channel count is reduced to M channels by spatial matrix mixing to a lower number of channels using multiplet pan laws. The multiplet pan laws include doublet, triplet, and quadruplet pan laws. For example, using a quadruplet pan law one of the N channels can be downmixed to four of the M channels to create a quadruplet channel. Spatial information as well and audio content is contained in the multiplet channels. During upmixing the downmixed channel is extracted from the multiplet channels using the corresponding multiplet pan law. The extracted channel then is rendered at any location within a playback environment.

Multiplet-based matrix mixing for high-channel count multichannel audio

A system for generating parametric stereo data from phase modulated stereo data is provided. A phase difference system receives left channel data and right channel data and determines a phase difference between the left channel data and the right channel data. A phase difference weighting system receives the phase difference data and generates weighting data to adjust left channel amplitude data and right channel amplitude data based on the phase difference data. A magnitude modification system adjusts the left channel amplitude data and the right channel amplitude data using the weighting data to eliminate phase data in the left channel data and the right channel data.

Parametric stereo conversion system and method

A system for identifying audio data is provided. The system includes a transform system receiving left channel audio data and right channel audio data and generating a plurality of frequency bins of left channel magnitude data, left channel phase data, right channel magnitude data and right channel phase data. A watermarking system receives watermarking data and modifies predetermined frequency bins of the left channel phase data and the right channel phase data to encode the watermarking data. A magnitude system receives the left channel magnitude data and the right channel magnitude data and increases the left channel magnitude data and the right channel magnitude data for one or more of the predetermined frequency bins to a threshold level if the left channel magnitude data and the right channel magnitude data for the corresponding frequency bin is less than the threshold level.

Temporally accurate watermarking system and method of operation

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.

Audio spatial environment engine

An audio spatial environment engine for flexible and scalable up-mixing from an M channel audio system to an N channel audio system, where M and N are integers and N is greater than M, is provided. The input M channel audio is provided to an analysis filter bank which converts the time domain signals into frequency domain signals. Relevant inter-channel spatial cues are extracted from the frequency domain signals on a sub-band basis and are used as parameters to generate adaptive N channel filters which control the spatial placement of a frequency band element in the up-mixed sound field. The N channel filters are smoothed across both time and frequency to limit filter variability which could cause annoying fluctuation effects. The smoothed N channel filters are then applied to adaptive combinations of the frequency domain input signals and are provided to a synthesis filter bank which generates the N channel time domain output signals.

The speech of two or more simultaneous speakers (or other simultaneous sounds) conveyed in a single channel are distinguished. Joint acoustic/modulation frequency analysis and display tools are used to localize and separate sonorant portions of multiple-speakers' speech into distinct regions using invertible transform functions. For example, the regions representing one of the speakers are set to zero, and the inverted modified display maintains only the speech of the other speaker. A combined audio signal is manipulated using a base acoustic transform, followed by a second modulation transform, which separates the combined signals into distinguishable components. The components corresponding to the undesired speaker are masked, leaving only the second modulation transform of the desired speaker's audio signal. An inverse second modulation transform of the desired signal is performed, followed by an inverse base acoustic transform of the desired signal, providing an audio signal for only the desired speaker.

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Bothell, WA, United States of America

Jeffrey Kenneth Thompson