Apple Inc.

Audience, Inc.

Knowles Electronics, Inc.

Brian D Clark

Tom-Davy W Saux

Dana C Massie

Vladan Bajic

Baptiste Pierre Paquier

Hanchi Chen

Thomas M Jensen

Leonardo Rub

Carlo Murgia

David P Rossum

Jean Laroche

Scott Charles Grinker

Ludger Solbach

Andrew P Bright

Lloyd Watts

Peter Santos

Richard Thomas Witek

Peter C Eastty

Victoria Chiu

David Pearce

A method performed by an in-ear headphone that includes a speaker and an internal microphone. The method receives a microphone signal from the internal microphone that indicates a current sound pressure level (SPL) in an ear canal of a user. The current SPL is a result of a control leak from the in-ear headphone into an ambient environment that reduces a SPL in the ear canal between 2 dB and 25 dB at a frequency within a frequency range than if otherwise not present. The method determines an active noise cancellation (ANC) filter based on the microphone signal and generates an anti-noise signal using the ANC filter. The method drives the speaker using the anti-noise signal to reduce the current SPL in the ear canal of the user as much as 25 dB at a frequency within the frequency range.

Feedback acoustic noise cancellation tuning

A system and method to encode and decode multiple audio signals to provide independent control of the audio signals is provided. A host device may encode the audio signals to enable a complete separation of the constituent audio signals when the mixed stream is decoded on a playback device. The gains of the audio signals may be independently controlled before they are mixed to increase the intelligibility of one audio signal relative to another audio signal at the playback device. The ability to separate the constituent audio signals from the mixed signals at the playback device allows the processing operations performed on the constituent audio signals and the associated path latencies to be independently chosen. In addition, in applications where the mixed stream is transmitted from a single host device to multiple playback devices, the constituent audio signals may be selectively masked on a playback device to increase user privacy.

Sending notification and multi-channel audio over channel limited link for independent gain control

An audio system has an ambient sound enhancement (ASE) function, in which an against-the-ear audio device having a speaker converts a digitally processed version of an input audio signal into amplified sound. The amplification may be in accordance with a stored hearing profile of the user. The audio system also has an acoustic noise cancellation (ANC) function that may be combined in various ways with the ASE function, and that may be responsive to voice activity detection. Other aspects are also described and claimed.

Headphone acoustic noise cancellation and speaker protection

A method for audio signal processing of microphone signals of a headphone. An audio signal from a first microphone of a headphone is filtered by an ANC system to produce a first filtered signal. Dynamic range control is performed upon the first filtered signal to produce a first dynamic range adjusted signal that drives a speaker of the headphone. Other aspects are also described and claimed.

Headphone acoustic noise cancellation and speaker protection or dynamic user experience processing

A multi-core audio processor includes a data protocol interface configured to receive a stream of audio data, a plurality of data processing cores including a single sample processing core and a block data processing core, an audio fabric block configured to route samples of the stream between the data protocol interface and the plurality of data processing cores. The single sample processing core includes an execution unit configured to execute one or more low latency instructions for performing computations for the samples.

Multi-core audio processor with low-latency sample processing core

In various embodiments, a floating-point arithmetic circuit includes a range exception detection circuit and an output circuit. The range exception detection circuit may generate a selection signal that indicates whether a floating-point arithmetic result generated within the floating-point arithmetic circuit is within a specified range. The output circuit may output the floating-point arithmetic result in response to the selection signal indicating the floating-point arithmetic result is within a specified range. The output circuit may output a corresponding specified value in response to the selection signal indicating the floating-point arithmetic result is not within the specified range. Accordingly, floating-point arithmetic operations may be performed in combination with an operation that limits a range of an output to a specified range.

Floating-point arithmetic operation range exception override circuit

Provided are methods and systems for passive training for automatic speech recognition. An example method includes utilizing a first, speaker-independent model to detect a spoken keyword or a key phrase in spoken utterances. While utilizing the first model, a second model is passively trained to detect the spoken keyword or the key phrase in the spoken utterances using at least partially the spoken utterances. The second, speaker dependent model may utilize deep neural network (DNN) or convolutional neural network (CNN) techniques. In response to completion of the training, a switch is made from utilizing the first model to utilizing the second model to detect the spoken keyword or the key phrase in spoken utterances. While utilizing the second model, parameters associated therewith are updated using the spoken utterances in response to detecting the keyword or the key phrase in the spoken utterances. User authentication functionality may be provided.

Passive training for automatic speech recognition

A semi-digital finite impulse response, FIR, filter is configured as a sparse FIR filter and as a minimum phase lag FIR filter. The FIR filter has a delay line composed of a number of sets of delay units sequentially coupled to each other, and where some of the sets of delay units have one or more untapped delay units as part of a cascade of two or more single-sample delay units. An analog summing node is coupled to the taps and produces at its output an analog version of a digital input signal that is fed to an input of the delay line. Other embodiments are also described and claimed.

Digital to analog conversion using semi-digital FIR filter

A system for audio processing of an acoustic signal captured by at least one microphone based on analysis of a video signal associated with the acoustic signal is provided. The video signal may be dynamically processed and analyzed in real time by the system to locate one or more sound sources and to determine distances between the sound sources and the at least one microphone. This information may be used to perform selective noise suppression, noise cancellation, or selective adjustment of acoustic signal components or energy levels acoustic signal components such that noise or unrelated acoustic signal components associated with sound sources not associated with located sound sources are suppressed or filtered. The video signal analysis may track sound source movement for dynamic steering of an acoustic beam towards the sound source.

Directional audio capture adaptation based on alternative sensory input

Digital methods and systems for signal processing and filtering are provided. The methods and corresponding systems provide asynchronous conversion of sampling rate frequencies and utilize advanced multistage phasor filters for converting an input signal having a first sampling rate into an output signal sampled in an arbitrary sequence of sampling times. The conversion process provides a sequence of sets of complex numbers representing a filtered version of the input signal. More specifically, the conversion process includes the calculation of values of the output signal by multiplying (e.g., scaling) the sets of complex numbers by a corresponding set of complex phasors, the complex phasors corresponding to the timing of the arbitrary time sequence to obtain a corresponding set of real results with the value of the output signal being the sum of the real results.

Sample rate conversion using infinite impulse response filters

Provided are methods and systems for digital signal processing by applying an improved SDFT technique for eliminating ripple in side lobes of a resulting spectrum. An exemplary method may comprise receiving an input signal which includes a number of discrete samples taken at regular time intervals. The input signal may then be filtered by a modified comb filter according to the sliding DFT technique to generate first and second filtered signals, the second filtered signal being filtered using a different filter coefficient. The first and second filtered signals may be processed by a first and second plurality of resonators respectively, according to the sliding DFT technique, to generate respective first and second SDFT output signals. The first and second SDFT output signals may then be selectively summed to generate a resulting output signal.

Sliding DFT windowing techniques for monotonically decreasing spectral leakage

Background audio can be provided during telephonic communication. Telephonic communication can be established via a network, such as between a user of a telephony device and a communication partner having a second telephony device. A voice signal may be received from the user via a microphone integral with the telephony device. An audio track can be retrieved, for example, from memory integral with the telephony device or from a third-party service provider via the communications network. Noise reduction is performed on the voice signal to produce a clean voice signal. The clean voice signal may be combined with the audio track to produce a combined signal, such that the audio track provides background audio to the clean voice signal. The combined signal can then be transmitted from the telephony device to the second telephony device via the communications network.

Providing background audio during telephonic communication

Systems and methods for asynchronous sample rate conversion are provided that allow time-varying arbitrary sample ratios. An uncorrected ratio between two arbitrary sample rates is corrected and subsequently used to perform an efficient sample rate conversion on the samples in a data stream. Coefficients of a (polyphase) finite impulse response filter are interpolated based on a current time register value. Additional computational efficiency (and a smaller finite impulse response filter) may be used due to oversampling the input signal to the finite impulse response filter.

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San Jose, CA, United States of America

Brian D Clark