Dolby Laboratories Licensing Corporation

Lake Dsp Pty Limited

Lake Technology Limited

Andrew Peter Reilly

David S McGrath

Richard James Cartwright

Adam Richard McKeag

Jeffrey Charles Riedmiller

Michael John Smithers

Timothy Alan Port

Benjamin Alexander Jancovich

Glen Norman Dickins

Glenn N Dickins

Michael William Mason

Nicholas L Engel

Niall Battson

Marcus Altman

Zhongjin Wang

Adam Richard Mckeag

David Stanley Mcgrath

Glenn Norman Dickins

Some noise compensation methods involve receiving microphone signals corresponding to ambient noise from a noise source location in or near an audio environment, determining or estimating a listener position in the audio environment and estimating at least one critical distance, which is a distance from the noise source location at which directly propagated sound pressure is equal to diffuse field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance and implementing a noise compensation method for the ambient noise based, at least in part, on an estimate of whether the listener position is within the critical distance.

Methods for reducing error in environmental noise compensation systems

Noise compensation method comprising: (a) receiving a content stream including content audio data; (b) receiving first microphone signals from a first device; (c) detecting ambient noise from a noise source location in or near the audio environment; (d) causing a first wireless signal to be transmitted from the first device to a second device, the first wireless signal including instructions for the second device to record an audio segment (e) receiving a second wireless signal from the second device; (f) determining a content stream audio segment time interval for a content stream audio segment; (g) receiving a third wireless signal from the second device, including a recorded audio segment captured via a second device microphone; (h) determining a second device ambient noise signal at the second device location; and (i) implementing a noise compensation method for the content audio data based, at least in part, on the second device ambient noise signal.

An apparatus and method of loudspeaker equalization. The method combines default tunings (generated based on a default listening environment) and room tunings (generated based on an end user listening environment) to result in combined tunings that account for differences between the end user listening environment and the default listening environment.

Method and apparatus for audio processing

The processing of audio signals during playback is provided, so that audio signals that fall below a specified threshold loudness level are processed to avoid making unwanted background noise audible. N-channel audio is received from a playback volume controller/leveler (). The level of the audio is compared with a threshold level. If the level is greater than the threshold level, the audio is processed with a first amount of gain in accordance with a first dynamic range control (DRC) compression curve that is tuned for professionally produced audio. If the level is less than or equal to the threshold level, the audio is processed with a second amount of gain in accordance with a second DRC compression curve that is designed to avoid boosting unwanted background noise. After applying the gain to the audio, the audio is sent to a downstream device.

Compressor target curve to avoid boosting noise

In some embodiments, a method for benchmarking an audio processing algorithm ('APA') while the APA is executed in a manner simulating expected real time execution by a deployed system. Other embodiments include a method including steps of determining a synthetic APA which corresponds to a counterpart APA (intended for real use by a first deployed system), and benchmarking the synthetic APA while it is executed in a manner simulating expected real time execution of the synthetic APA by a contemplated deployed system. Other aspects include a system or device configured to implement any embodiment of the inventive method, or including a memory which stores data indicative of at least one synthetic APA determined in accordance with, or a benchmark generated by, an embodiment of the inventive method or steps thereof, and a computer readable medium which stores code for implementing any embodiment of the inventive method or steps thereof.

Audio benchmarking with simulated real time processing of audio

Systems and methods for leveling loudness variation in an audio signal are described. Embodiments use both a perceptual leveling algorithm and a standards-based loudness measure together to minimize audio process artifacts and ensure that the measured loudness of the processed audio is close to a required measure, according to a particular standard measurement of loudness. These systems and methods can be used either offline or in real-time.

System and method for leveling loudness variation in an audio signal

A method, an apparatus, and a software product to process a plurality of input audio signals. The apparatus accepts a plurality of input signals and includes a multi-input, multi-output reverberator arranged to generate a set of output signals including delayed reverberation components simulating the reverberations a listener is likely to hear in a listening environment. The apparatus further includes a multi-input, two-output filter accepting the outputs of the reverberator and the plurality of input terminals, providing outputs for the left and right ears, and configured to implement a set of head related transfer functions corresponding to a listening environment and a set of directions of a listener in the listening environment. The apparatus is such that a listener listening to the outputs through headphones has the sensation of listening to the plurality of input audio signals as if they are emanating from a plurality of loudspeakers spatially located in the listening environment at a corresponding plurality of directions.

Processing audio signals with head related transfer function filters and a reverberator

An apparatus for creating, utilizing a pair of oppositely opposed headphone speakers, the sensation of a sound source being spatially distant from the area between the pair of headphones, the apparatus comprising: (a) a series of audio inputs representing audio signals being projected from an idealised sound source located at a spatial location relative to the idealised listener; (b) a first mixing matrix means interconnected to the audio inputs and a series of feedback inputs for outputting a predetermined combination of the audio inputs as intermediate output signals; (c) a filter system of filtering the intermediate output signals and outputting filtered intermediate output signals and the series of feedback inputs, the filter system including separate filters for filtering the direct response and short time response and an approximation to the reverberant response, in addition to the feedback response filtering for producing the feedback inputs; and (d) a second matrix mixing means combining the filtered intermediate output signals to produce left and right channel stereo outputs.

Utilization of filtering effects in stereo headphone devices to enhance spatialization of source around a listener

A method for applying a computation utilizing a processor () capable of accessing an on-chip memory () and data from an off-chip source (), the method comprising the iterative steps of retrieving at the on-chip memory () successive frames of input data from the off-chip source (); computing from a current input frame of data () available in the on-chip memory current result elements for completing a current output frame of results (), and pre-computing from the current frame of input data future result elements for contributing to at least one future output frame of results.

Low latency computation in real time utilizing a DSP processor

A method of processing a series of input audio signals representing a series of virtual audio sound sources placed at predetermined positions around a listener to produce a reduced set of audio output signals for playback over speaker devices placed around a listener, the method comprising the steps of: (a) for each of the input audio signals and for each of the audio output signals: (i) convolving the input audio signals with an initial head portion of a corresponding impulse response mapping substantially the initial sound and early reflections for an impulse response of a corresponding virtual audio source to a corresponding speaker device so as to form a series of initial responses; (b) for each of the input audio signals and for each of the audio output signals: (i) forming a combined mix from the audio input signals; and (ii) forming a combined convolution tail from the tails of the corresponding impulse responses; (iii) convolving the combined mix with the combined convolution tail to form a combined tail response; (c)for each of the audio output signals: (i) combining a corresponding series of initial responses and a corresponding combined tail response to form the audio output signal.

Audio signal processing method and apparatus

A method of performing a convolution of an input data stream, the method comprising the steps of: (a) transforming an initial series of input data values into the frequency domain to form initial frequency domain data values; (b) multiplying the frequency domain data values with a set of coefficients so as to form convolved frequency data; (c) transforming the convolved frequency data into a spatial domain so as to form an initial series of output data values; (d) performing steps (a) to (c) on a subsequent series of input data values of the input data stream so as to form a subsequent series of output data values; the subsequent series of input data values overlapping the initial series of input data values; and (e) cross fading overlapping portions of the initial and subsequent output data values.

Variants on low-latency convolution-smooth updating of filter response using crossfades

A method of compressing an audio signal comprising a series of values is disclosed, the method comprising: utilising a predetermined maximum number of bits B for representing values; dividing the audio signal into a series of blocks of a predetermined length N; and for each of the blocks: computing the maximum number of bits b required to represent the values within a block; packing an indicator of b bits in an output stream; and for each value within the block: where the value exceeds 2 writing only the b most significant bits to the output stream; and where the value does not exceed 2 writing the value to the output stream. The aforementioned methods are well suited to compressing impulse response functions. There is also disclosed a method of decoding an audio signal encoded by the aforementioned methods.

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Hurlstone Park, Australia

Andrew Peter Reilly