Abstract: An Auto Setup Program (“ASP”) may be implemented in the sound system of a vehicle to accomplish the goal of a dramatically improved audio quality achievable in any vehicle, regardless of the speaker components and head-unit utilized. The ASP performs automated time alignment measurements and equalizer processes using an external microphone. The ASP eliminates the need for expensive and time consuming professional tuning operations. The on-board ASP can be run multiple times as desired. For example, the ASP can be executed after changes to the system, such as upgrading speakers.

Abstract: Audio amplification used in security systems need to be robust and have failsafe capability, they also need to be compact and energy efficient. A means of providing this by combining class D amplifiers in series is provided along with means to disconnect the amplifiers in a failure mode so as to provide ongoing operation should 1 of the amplifiers malfunction or another part of the system associated one of the amplifiers malfunction. The invention comprises an audio output stage which may be further integrated into an audio system having a supervisory controller to manage the transition from normal operation to failure state operation.

Abstract: An integrated circuit for digital signal routing. The integrated circuit has analog and digital inputs and outputs, including digital interfaces for connection to other integrated circuits. Inputs, including the digital interfaces, act as data sources. Outputs, including the digital interfaces, act as data destinations. The integrated circuit also includes signal processing blocks, which can act as data sources and data destinations. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. Data from a data source is buffered for an entire period of a data sample clock so that the multiply-accumulate block can retrieve the data at any point in the period, and output data of the multiply-accumulate block is buffered for an entire period of the data sample clock so that the data destination can retrieve the data at any point in the period.

Abstract: An example audio decoding device includes processing circuitry and a memory device coupled to the processing circuitry. The processing circuitry is configured to receive, in a bitstream, encoded representations of audio objects of a three-dimensional (3D) soundfield, to receive metadata associated with the bitstream, to obtain, from the received metadata, one or more transmission factors associated with one or more of the audio objects, and to apply the transmission factors to the one or more audio objects to obtain parallax-adjusted audio objects of the 3D soundfield. The memory device is configured to store at least a portion of the received bitstream, the received metadata, or the parallax-adjusted audio objects of the 3D soundfield.

Abstract: A system for providing spatially projected audio communication between members of a group, the system mounted onto a respective user of the group, the system comprising: a detection unit; a communication unit; a processing unit; and an audio interface unit, configured to audibly present the synthesized spatially resolved audio signal to the user.

Abstract: The technology described in this document can be embodied in a computer-implemented method of controlling a wearable audio device configured to provide an audio output. The method includes receiving data indicating the wearable audio device is proximate a geographic location associated with a plurality of localized audio messages, and presenting, using one or more user-interfaces, a representation of the plurality of localized audio messages. The method also includes receiving information indicative of a selection of a portion of the one or more user interfaces, and initiating playback of one or more of the plurality of localized audio messages that correspond to the selection, on the wearable audio device.

Abstract: An audio amplifier has an analog current source amplifier and a class D amplifier. The analog current source amplifier is active in a first mode to drive a speaker when an audio signal has smaller amplitude. The class D amplifier is active in a second mode to drive the speaker when the audio signal has larger amplitude. Other aspects are also described and claimed.

Abstract: Systems and methods for low latency adaptive noise cancellation include an audio sensor to sense environmental noise and generate a noise signal, an audio processing path to receive an audio signal, process the audio signal through an interpolation filter, and generate a primary audio signal having a first sample frequency, an adaptive noise cancellation processor to receive the noise signal and generate an anti-noise signal, a direct interpolator to receive the anti-noise signal and generate an anti-noise signal having the first sample frequency, and a limiter to provide clipping to reduce a number of bits in the anti-noise signal, an adder operable to combine the primary audio signal and the anti-noise signal and generate a combined output signal, and a low latency filter to process the combined output signal.

Abstract: A playback device is configured to receive, via a network interface, a source stream of audio including first and second channel streams of audio, and to produce, via respective first and second speaker drivers, a first channel audio output and a second channel audio output. The playback device is also configured to receive, via one or more microphones, a captured stream of audio including first and second portions corresponding to the respective first and second channel audio outputs. The playback device is also configured to combine at least the first channel stream of audio and the second channel stream of audio into a compound audio signal and perform acoustic echo cancellation on the compound audio signal and thereby produce an acoustic echo cancellation output, then to apply the acoustic echo cancellation output to the captured stream of audio and thereby increase a signal-to noise ratio of the captured stream of audio.

Abstract: A microphone assembly includes a transducer element and a processing circuit. The processing circuit includes an analog-to-digital converter (ADC) configured to receive, sample and quantize a microphone signal generated by the transducer element to generate a corresponding digital microphone signal. The processing circuit includes a feedback path including a digital loop filter configured to receive and filter the digital microphone signal to provide a first digital feedback signal and a digital-to-analog converter (DAC) configured to convert the first digital feedback signal into a corresponding analog feedback signal. The processing circuit additionally includes a summing node at the transducer output configured to combine the microphone signal and the analog feedback signal.

Abstract: A system comprises an ear-worn electronic device configured to be worn by a wearer. The ear-worn electronic device comprises a processor and memory coupled to the processor. The memory is configured to store an annoying sound dictionary representative of a plurality of annoying sounds pre-identified by the wearer. A microphone is coupled to the processor and configured to monitor an acoustic environment of the wearer. A speaker or a receiver is coupled to the processor. The processor is configured to identify different background noises present in the acoustic environment, determine which of the background noises correspond to one or more of the plurality of annoying sounds, and attenuate the one or more annoying sounds in an output signal provided to the speaker or receiver.

Abstract: An audio processing device comprises a) at least one input unit for providing a time-frequency representation Y(k,n) of an electric input signal representing sound consisting of target speech and noise signal components, where k and n are frequency band and time frame indices, respectively, b) a noise reduction system configured to b1) determine a first signal to noise ratio estimate ?(k,n) of said electric input signal, and to b2) determine a second signal to noise signal ratio estimate ?(k,n) of said electric input signal from said first signal to noise ratio estimate ?(k,n) based on a recursive algorithm providing non-linear smoothing, and wherein a determination of said one or more bias and/or smoothing parameters comprises the use of supervised learning, e.g. one or more neural networks. The invention may be used in audio processing devices, such as hearing aids, headsets, ear phones, active ear protection systems, handsfree telephone systems, mobile telephones, etc.

Abstract: A workspace assembly includes at least a first sound sensor located in a first facility space, at least one communication device located within the first space, and a processor in communication with the at least a first sound sensor and the communication device. The processor is adapted to compare the volume of sound emanating from within the first space to a threshold level and to generate a signal via the communication device when the volume of sound emanating from within the first space exceeds the threshold level. The processor also periodically automatically adjusts the threshold level.

Abstract: Implementations generally relate to automated equalizer adjustments based on audio characteristics. In some implementations, a method includes detecting music that is being currently played on an audio device. The method further includes adjusting one or more equalizer settings of an equalizer device based at least in part on a music genre associated with the music. The method further includes outputting the music based at least in part on the adjusting of the one or more equalizer settings of the equalizer device.

Abstract: The invention discloses an earphone device comprising a first case, a first speaker unit, a first recording unit, and a second recording unit. The first speaker unit, disposed inside the first case, emits a first testing sound signal according to a test command. The first recording unit, disposed inside the first case, records a first environment sound signal according to a record command or a noise cancelling command. The second recording unit, disposed inside the first case, records a first feedback sound signal, related to the first testing sound signal, according to the test command.

Abstract: A digitally controlled microphone with audio compressor includes a capsule; a compressor circuitry having a compressor engine, a digitally controlled root mean square (RMS) detector and a digitally controlled analog processor; a bypass switch to the compressor circuitry; an analog to digital converter (ADC); a digital processor that sends digital audio to a host and receives a second digital audio signal; a digital to analog converter (DAC) and a summing circuit that combine the audio with the audio from the host and send it to a digitally-controlled headphone amplifier and a headphone jack; a microphone body; a serial host interface connector; and hardware controls on the microphone body.

Abstract: Embodiments are provided for satellite volume control. An example method includes receiving an input at a playback device to adjust a volume for a plurality of playback devices that are grouped for synchronous playback of audio content, wherein the plurality of playback devices includes the playback device. The method also includes sending a first message over a network from the playback device to a device associated with the plurality of playback devices, the first message including information based on the input, wherein the information is used to adjust the volume of the plurality of playback devices. The method also includes receiving a second message at the playback device over the network, the second message including information for the volume of the playback device, wherein the volume is based on the adjusted volume of the plurality of playback devices.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output.

Abstract: Various implementations include approaches for dynamically demonstrating device capabilities in a wearable audio device based upon detected environmental conditions. In certain cases, these approaches include initiating a demonstration using the wearable audio device to provide the user with an example of the device capabilities, based upon detected environmental conditions proximate the wearable audio device and/or a biometric state of the device user.

Abstract: At least one exemplary embodiment is directed to a method of earphone feature control by using a graphic user interface on a device wirelessly linked to the earphone.