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: 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: 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: 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.

Abstract: The present disclosure provides systems and methods that provides users with information pertaining to the audio properties at one or more points of interest. A database associated the audio properties with the points of interest is built using audio input received from devices at the points of interest. The device may determine that audio properties associated with the received audio input. The audio properties may determine a type of background noise and/or a volume of the background noise. If the type of background noise is music, the audio properties may further include a music genre, a title of a song, whether the music is recorded or there is a live band, etc. The audio properties associated with the point of interest may be updated in a database real time.

Abstract: The transfer function estimation device includes: a correlation matrix computing unit 43 computing a correlation matrix of N frequency domain signals y(f,l); a signal space basis vector computing unit 44 obtaining M vectors v1(f), . . . , vM(f) from eigenvectors of the correlation matrix from highest in the order of corresponding eigenvalues; and a plural RTF estimation unit 45 determining ti(f), . . . , tM(f) that satisfy the relationship of Expression (1), determining a matrix D(f) that is not a zero matrix and that makes ui(f), . . . , uM(f) defined by Expression (2) sparse in a time direction, determining ci,1(f), . . . , cM,N(f) that satisfy the relationship of Expression (3), and outputting c1(f)/c1,j(f), . . . , cM(f)/cM,j(f) as a relative transfer function, where j is an integer of 1 or more and not more than N.

Abstract: Disclosed is a signal processing apparatus including a surrounding sound signal acquisition unit, a NC (Noise Canceling) signal generation part, a cooped-up feeling elimination signal generation part, and an addition part. The surrounding sound signal acquisition unit collects a surrounding sound to generate a surrounding sound signal. The NC signal generation part generates a noise canceling signal from the surrounding sound signal. The cooped-up feeling elimination signal generation part generates a cooped-up feeling elimination signal from the surrounding sound signal. The addition part adds together the generated noise canceling signal and the cooped-up feeling elimination signal at a prescribed ratio.

Abstract: A response activity pattern may be ascertained from a set of computing devices obtained during a monitoring mode of operation. The monitoring mode of operation can be initiated when a user command is determined to be a new or unknown command. The response activity pattern may be used to generate a response profile indicating an activity to carry out using one or more user devices to perform a task associated with the user command. When an indication of a previously unknown user command to perform a task is received, the generated response profile can be used to perform the designed task by carrying out the activity using the one or more user device. In this way, the activity that should be carried out to perform the task can be learned based on the monitored user activity related to the one or more user devices.

Abstract: In-ear sound pressure level, SPL, is determined that is caused by output audio being converted into sound by a headset worn by a user. The in-ear SPL is converted into a sound sample having units that are suitable for evaluating sound noise exposure. These operations are repeated to produce a sequence of sound samples during playback. This sequence of sound samples is then written to a secure database. Access to the database is authorized by the user. Other aspects are also described and claimed.