Abstract: Latency on different devices (e.g., devices of differing brand, model, vintage, etc.) can vary significantly and tens of milliseconds can affect human perception of lagging and leading components of a performance. As a result, use of a uniform latency estimate across a wide variety of devices is unlikely to provide good results, and hand-estimating round-trip latency across a wide variety of devices is costly and would constantly need to be updated for new devices. Instead, a system has been developed for crowdsourcing latency estimates.

Abstract: In one embodiment, a method includes, by a first electronic device: receiving, from an input sensor of the first electronic device, a predetermined input, where the predetermined input corresponds to a request to wirelessly connect the first electronic device to one or more second electronic devices within a communication range of a transceiver of the first electronic device, wirelessly connecting, using the transceiver of the first electronic device responsive to receiving the predetermined input, to the one or more second electronic devices using a communications profile associated with each of the one or more second electronic devices, where the communications profile includes one or more parameters for verifying to connect the first electronic device and the one or more second electronic devices, and redirecting a live audio communication between the first electronic device and a control device to the one or more second electronic devices.

Abstract: The disclosure is directed to a process that can predict and prevent an audio artifact from occurring. The process can monitor the systems, processes, and execution threads on a larger system/device, such as a mobile or in-vehicle device. Using a learning algorithm, such as deep neural network (DNN), the information collected can generate a prediction of whether an audio artifact is likely to occur. The process can use a second learning algorithm, which also can be a DNN, to generate recommended system adjustments that can attempt to prevent the audio glitch from occurring. The recommendations can be for various systems and components on the device, such as changing the processing system frequency, the memory frequency, and the audio buffer size. After the audio artifact has been prevented, the system adjustments can be reversed fully or in steps to return the system to its state prior to the system adjustments.

Abstract: An example technique includes a computing system storing media item identifiers of curated media items associated with one or more service providers. A media curating service aggregates the media item identifiers of curated media items. The example technique further involves receiving, from a media playback system, a first message comprising a service provider access identifier. The service provider access identifier is based on a user account of the media playback system registered to at least one service provider. Based on receiving the first message, the computing system determines media item identifiers of curated media items that are associated with the at least one service provider with which the user account of the media playback system is registered and causes the media playback system to play back the curated media items based on the determined media item identifiers of the curated media items.

Abstract: A device comprising: at least two interfaces for interconnecting between at least one audio source and at least one audio sink device; and unidirectional circuitry configured to allow transmission of an audio signal only from said at least one audio source to said at least one audio sink device.

Abstract: Signal to noise ratio, SNR, is determined in an acoustic ambient environment of an against-the-ear audio device worn by a user, wherein the acoustic ambient environment contains speech by a talker. When the SNR is above a threshold, dynamic range control is applied, as positive gain versus input level, to an audio signal from one or more microphones of the audio device. When the SNR is below the threshold, the dynamic range control applies as zero gain or negative gain to the audio signal. Other aspects are also described and claimed.

Abstract: Provided herein are system, apparatus, device, method and/or computer program product embodiments, and/or combinations and sub-combinations thereof, for transmitting messages to a display device based on detected audio output from the display device. In a given embodiment, a digital media player can transmit a message to the display device to control the audio output of the speaker using a transmission protocol, and transmit a set of instructions to the remote control device to confirm the display device processed the message by determining whether an attribute associated with the audio output from the speaker is within a predetermined threshold. In response to receiving a message that an attribute associated with the audio output is within a predetermined threshold, the digital media player can determine that the display device is enabled to receive communication via the transmission protocol.

Abstract: Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

Abstract: Systems and methods are provided for monitoring auditory stimulation provided to a listener through a personal listening device. In one example, a method includes monitoring one or more acoustic profile parameters of an audio signal received from an audio source and rendered on a headphone assembly based on one or more received transducer parameters of one or more transducers of the headphone assembly; monitoring one or more listener-headphone engagement parameters based on headphone position data from a sensor coupled to the headphone assembly; and adjusting the audio signal and/or acoustic stimulation based on the monitoring of the one or more acoustic profile parameters and the monitoring of one or more listener-headphone engagement parameters, wherein the one or more listener-headphone engagement parameters include a current engagement status of the headphone assembly with respect to the user and a duration of active engagement of the headphone assembly with respect to the user.

Abstract: A method for muting and unmuting a microphone is provided. The method includes providing a processor, receiving an input microphone signal, measuring the input microphone signal for a loudness level at a sampling rate, calculating a mute threshold level, checking if the loudness level is higher than or equal to the mute threshold level, and resetting a mute delay timer upon determining that the loudness level is higher than or equal to the mute threshold level and obtaining the input microphone signal, or checking if the mute delay timer is running upon determining that the loudness level is not higher than or equal to the mute threshold level and attenuating the input microphone signal if the mute delay timer is not running or obtaining the input microphone signal if the mute delay timer is still running, and writing the input microphone signal or attenuated input microphone signal to an output buffer.

Abstract: An example playback device is a first playback device in a media system. The first playback device is configured to resolve audio conflicts with one or more other playback devices in the media system by: (i) capturing, via a microphone of the first playback device, audio content played back by a second playback device, (ii) identifying the second playback device as a source of the captured audio content; and (iii) responsive to identifying the second playback device as the source of the captured audio content, altering a playback characteristic of the second playback device or the first playback device to reduce an audio interference between the first and second playback devices.

Abstract: An example computing device is configured to transmit playback preference tracking information to a media service registered with a media playback system and, after transmitting the playback preference information, (a) determine a first time boundary corresponding to a beginning of a zone scene that causes a first playback device in a first zone of the media playback system to be configured for synchronous playback of media with a second playback device in a second zone of the media playback system, (b) determine a second time boundary corresponding to an end of the zone scene, and (c) transmit, to the media service registered with the media playback system, a request to remove playback preference tracking information received between the first time boundary and the second time boundary.

Abstract: A method for controlling target sound playback. A display screen of a control device is configured to display a target sound user setting for controlling target sound playback, wherein the target sound user setting controls a stored target sound level parameter that is stored within memory of the control device. A target sound sequence is generated in accordance with the user setting. A speaker is driven with the target sound sequence. Other aspects are also described and claimed.

Abstract: Techniques are disclosed relating to implementing user-created controls to modify music content. A music generator system may be configured to automatically generate output music content by selecting and combining audio tracks based on various parameters. Users may create their own control elements that the music generator system may train (e.g., using AI techniques) to generate output music content according to a user's intended functionality of a user-created control element.

Abstract: A wearable electronic device (WED) worn on a head of a user displays a virtual image that has sound. One or more processors process the sound for the virtual image into binaural sound for the user. The binaural sound has a sound localization point (SLP) with a coordinate location that occurs behind the virtual image while the image is located a near-field distance from the head of the user.

Abstract: A method of managing at least one audio playback equipment group including at least one piece of audio playback equipment is provided with a light-emitting identification interface. The method includes the steps of: using the light-emitting identification interface of the piece of audio playback equipment to display a light pattern characteristic at least of the group to which said piece of equipment belongs and/or characteristic of the role of the piece of equipment within said group in the event of at least one of the pieces of audio playback equipment being selected, and performing an action relating to the associated piece of audio playback equipment.

Abstract: A remote test infrastructure can enable a developer to use a local browser to run and test an application on a remote device. The remote device is coupled to a host machine and connected via a communication network to the browser. The infrastructure can stream a video feed of the display of the remote device to the browser, enabling the developer to examine the result of testing and development of the application running on the remote device. For applications that have features requiring audio input, the infrastructure can pair an external component, such as the host machine, as a virtual audio input device, mimicking an external microphone. The virtual audio input device can capture an audio file or an audio stream and provide the audio as input to the application running on the remote device thereby simulating the application receiving an input from an external microphone.

Abstract: A method performed by an audio source device. The method obtains an input audio signal and determines a sound output level of a headset based on the input audio signal, a user volume setting, and a sound output sensitivity of the headset. The method determines whether the sound output level is above a threshold. In response to determining that the sound output level is above the threshold, a scalar gain is applied upon the input audio signal to produce an output audio signal for output by the headset.

Abstract: An adaptive music playback system is disclosed. The system includes a composition system that receives information corresponding to user activity levels. The composition system determines target musical criteria corresponding to the user activity levels and modifies the composition of a song in response to changes in user activity.

Abstract: A plurality of microphone signals can be captured with a plurality of microphones of the device. One or more echo dominant audio signals can be determined based on a pick-up beam directed towards one or more speakers of a playback device. Sound that is emitted from the one or more speakers and sensed by the plurality of microphones can be removed from plurality of microphone signals, by using the one or more echo dominant audio signals as a reference, resulting in clean audio.