Dolby Laboratories Licensing Corporation

Matthew Conrad Fellers

Grant Allen Davidson

Mark Stuart Vinton

Michael Mead Truman

Mark Franklin Davis

Louis Dunn Fielder

Joshua Brandon Lando

Vinay Melkote

Kuan-Chieh Yen

C Phillip Brown

Alan Jeffrey Seefeldt

Brett Graham Crockett

Charles Quito Robinson

Regunathan Radhakrishnan

Sripal S Mehta

Rhonda J Wilson

Claus Bauer

Kenneth James Gundry

Stewart Murrie

Rongshan Yu

Brad Basler

Vivek Kumar

S Spencer Hooks

Matthew Aubrey Watson

Douglas Walter Hansen

Eric Benjamin

Eric M Benjamin

A method of calibrating a feedback-based noise cancellation system of an ear device may involve obtaining a measured plant response of the ear device, obtaining a reference plant response value and determining a plant response variation between the reference plant response value and a value corresponding to the measured plant response. The method may involve obtaining a measured a coupler response of the ear device, obtaining a reference coupler response value and determining a coupler response variation between the reference coupler response value and a value corresponding to the measured coupler response. The method may involve determining, based at least in part on the plant response variation and the coupler response variation, a microphone signal gain correction factor and applying the microphone signal gain correction factor to ear device microphone signals that are input to a feedback loop of the feedback-based noise cancellation system.

Auto calibration of an active noise control system

The computational resources that are needed to apply a transform-based filterbank to a limited-bandwidth audio signals are reduced by performing an integrated process of combining real-valued input data into complex-valued data and applying a short transform to the complex-valued data, applying a bank of very short transforms to the output of the integrated process, and deriving a sequence of real-valued output data from the outputs of the bank of very short transforms.

Reduced complexity transform for a low-frequency-effects channel

A method for creating a head-related impulse response (HRIR) for use in rendering audio for playback through headphones comprises receiving location parameters for a sound including azimuth, elevation, and range relative to a head of a listener, applying a spherical head model to the azimuth, elevation, and range input parameters to generate binaural HRIR values, computing a pinna model using the azimuth and elevation parameters to apply to the binaural HRIR values to pinna modeled HRIR values, computing a torso model using the azimuth and elevation parameters to apply to the pinna modeled HRIR values to generate pinna and torso modeled HRIR values, and computing a near-field model using the azimuth and range parameters to apply to the pinna and torso modeled HRIR values to generate pinna, torso and near-field modeled HRIR values.

Structural modeling of the head related impulse response

Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.

Signal decorrelation in an audio processing system

Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence ('IDC') between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence ('ICC') between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data.

Methods for controlling the inter-channel coherence of upmixed audio signals

Embodiments are directed to an interconnect for coupling components in an object-based rendering system comprising: a first network channel coupling a renderer to an array of individually addressable drivers projecting sound in a listening environment and transmitting audio signals and control data from the renderer to the array, and a second network channel coupling a microphone placed in the listening environment to a calibration component of the renderer and transmitting calibration control signals for acoustic information generated by the microphone to the calibration component. The interconnect is suitable for use in a system for rendering spatial audio content comprising channel-based and object-based audio components.

Bi-directional interconnect for communication between a renderer and an array of individually addressable drivers

Embodiments are described for a system of rendering spatial audio content in a listening environment. The system includes a rendering component configured to generate a plurality of audio channels including information specifying a playback location in a listening area, an upmixer component receiving the plurality of audio channels and generating, for each audio channel, at least one reflected sub-channel configured to cause a majority of driver energy to reflect off of one or more surfaces of the listening area, and at least one direct sub-channel configured to cause a majority of driver energy to propagate directly to the playback location.

Reflected and direct rendering of upmixed content to individually addressable drivers

Received audio data may include a first set of frequency coefficients and a second set of frequency coefficients. Spatial parameters for at least part of the second set of frequency coefficients may be estimated, based at least in part on the first set of frequency coefficients. The estimated spatial parameters may be applied to the second set of frequency coefficients to generate a modified second set of frequency coefficients. The first set of frequency coefficients may correspond to a first frequency range (for example, an individual channel frequency range) and the second set of frequency coefficients may correspond to a second frequency range (for example, a coupled channel frequency range). Combined frequency coefficients of a composite coupling channel may be based on frequency coefficients of two or more channels. Cross-correlation coefficients, between frequency coefficients of a first channel and the combined frequency coefficients, may be computed.

Audio signal enhancement using estimated spatial parameters

A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.

Method and apparatus for reducing the effect of environmental noise on listeners

An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.

Method for adaptive control and equalization of electroacoustic channels

A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal including temporal shape. Spectral components are synthesized having the one or more assessed characteristics, integrated with the subband signals and passed through a synthesis filterbank to generate an output signal.

Audio coding system using temporal shape of a decoded signal to adapt synthesized spectral components

Blocks of audio information are arranged in groups that share encoding control parameters to reduce the amount of side information needed to convey the control parameters in an encoded signal. The configuration of groups that reduces the distortion of the encoded audio information may be determined by any of several techniques that search for an optimal or near optimal solution. The techniques include an exhaustive search, a fast optimal search and a greed merge, which allow the search technique to tradeoff the reduction in distortion against the bit rate of the encoded signal and/or the computational complexity of the search technique.

Audio coding based on block grouping

Audio coding processes like quantization can cause spectral components of an encoded audio signal to be set to zero, creating spectral holes in the signal. These spectral holes can degrade the perceived quality of audio signals that are reproduced by audio coding systems. An improved decoder avoids or reduces the degradation by filling the spectral holes with synthesized spectral components. An improved encoder may also be used to realize further improvements in the decoder.

Audio coding system using spectral hole filling

A receiver in an audio coding system receives a signal conveying frequency subband signals representing an audio signal. The subband signals are examined to assess one or more characteristics of the audio signal. Spectral components are synthesized having the assessed characteristics. The synthesized spectral components are integrated with the subband signals and passed through a synthesis filterbank to generate an output signal. In one implementation, the assessed characteristic is temporal shape and noise-like spectral components are synthesized having the temporal shape of the audio signal.

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San Francisco, CA, United States of America

Matthew Conrad Fellers