Abstract: Systems and methods are described herein for automatically changing the priority of a media asset using a continuous listening device. The system may receive an audio clip of a conversation a user, and then determine whether that conversation relates to any of the programs recorded or scheduled to be recorded on a storage device associated with the user. In response to determining that the media asset does relate to one of the programs recorded or scheduled to be recorded on the storage device, a user profile may be consulted to determine past instances of the user discussing the media asset, and, if a measure of the total number of instances the user discussed the media asset meets a threshold measure, the priority of the media asset may be updated.

Abstract: Aspects of the present disclosure provide methods, apparatuses, and systems for a personalized wake-up system. A sleep assistance device outputs a sound. In response to detecting a sound that exhibits one or more predetermined sound properties, actions are taken to adjusting the sound in an effort to wake the subject. Examples of predetermined sound properties include any sound detected over a threshold decibel level, sounds detected at a certain frequency spectrum. According to aspects, subject may further configure the system to alert the subject based on a set of personalized sounds the subjects considers to be important. According to aspects, the audio device or system is configured to determine the subject is awake based on collected biosignal parameters. The audio device or system takes further actions to disrupt the subject's sleep until the subject is determined to be awake.

Abstract: Systems and methods for intelligent playback of media content may include an intelligent media playback system that, in response to determining the speech tempo in audio content by measuring syllable density of speech in the audio content, automatically adjusts a playback speed of the audio content as the audio content is being played based on the determined speech tempo. In some embodiments, the system may automatically and dynamically adjust the playback speed to result in a desired target speech tempo. In addition, the system may determine whether to automatically adjust playback speed of the audio content, as the media is being played, based on the detected speech tempo of the speech in the audio content and the determined type of content of media. Such automatic adjustments in playback speed result in more efficient playback of the audio content.

Abstract: A content management system and/or client device can enable a user to initiate a quick play mode where a content category and content medium are selected for the user. A client device and/or a content management system can select a content medium for a user based on one or more factors, such as the content category. Certain content categories of content can be preferably delivered in certain content mediums. In some embodiments, a content management system and/or client device can select a content medium for a user based on contextual data gathered from the user. Contextual data can be data describing the user's current state and/or environment. For example, contextual data can include data such as the time of day, geographic location, etc.

Abstract: A skip behavior analyzer is part of a media delivery system that allows for unbiased A/B testing of a plurality of versions of a song. The media delivery system stores a plurality of versions of a song and randomly selects, for each requesting device, a version of the song to associate with that device. Each time the device requests the song, thereafter, the media delivery system will provide the same version of the song for consistency. The media delivery system then gathers song play and skip information, calculates the differences in distribution of the skip behavior, and provides the skip information to allow a music composer to better determine which version of a song is more popular and why that is so.

Abstract: An audio playback control method of a mobile terminal comprises: collecting proximity information of a user's hand within a preset time period by using a plurality of proximity sensors of a wireless earphone; acquiring information about whether each proximity sensor is shielded at each moment within the preset time period according to the proximity information; determining the order in which different proximity sensors are shielded according to the information about each proximity sensor being shielded; acquiring an operation trajectory of the user's hand according to the order in which different proximity sensors are shielded; and acquiring a corresponding audio playback operation instruction according to the operation trajectory and a preset trajectory instruction library, to realize the control over the audio playback of the mobile terminal.

Abstract: A method, computer program product, and computing system for determining a time delay between a first audio signal received on a first audio detection system and a second audio signal received on a second audio detection system. The first and second audio detection systems are located within a monitored space. The first audio detection system is located with respect to the second audio detection system within the monitored space.

Abstract: Systems, device and techniques are disclosed for performing a media action such as a democratic skip action based on input from one or more users that are part of a media group. One or more users within a media group may provide preference indicators (e.g., skip, keep, null) and, upon reaching a threshold number of preference indicators, a media action (e.g., skip song) may be taken. The preference indicators may be weighed based on one or more factors such as a user or media item.

Abstract: A method, computer program product, and computing system for initially aligning two or more audio signals to address coarse temporal misalignment between the two or more audio signals. The two or more audio signals are detected by two or more audio detection systems within a monitored space. The two or more audio signals are subsequently realigned to address ongoing temporal signal drift between the two or more audio signals.

Abstract: Embodiments described include systems and methods for incorporating a watermark in an audio output. An embedded browser, which is executable on one or more processors of a client device, may detect an audio data stream from a network application accessed via the embedded browser. A watermarking engine of the embedded browser intercepts the audio data stream responsive to detecting the audio data stream. The watermarking engine incorporates a digital signal corresponding to a watermark into the audio data stream, prior to being produced as an audio output by an audio speaker. The watermarking engine causes the watermark to be present in the audio output produced by the audio speaker, the watermark configured to be inaudible by a human and recordable by an audio recording device.

Abstract: Example techniques disclosed herein relate to location-based playlists. In an example implementation, a system receives, from a mobile device, first GPS data indicating that the mobile device is located at a first physical location, wherein an application on the mobile device is registered with a particular user account of the cloud service. The system determines that the first physical location represented in the first GPS data corresponds to a location of a first business associated with one or more first playback devices within the cloud service. The one or more first playback devices are playing back first audio tracks. As each first audio track is at least partially played back, the system adds the respective first audio track to a location-based playlist associated with the particular user account of the cloud service. The computing system causes the mobile device to display an indication of the location-based playlist.

Abstract: The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

Abstract: An audio mixer includes a reception portion and a controller. The reception portion receives a user's operation. The controller performs: display processing that displays a screen indicating a correspondence between a plurality of channels and a bus on a display; and setting processing that, when a specific operation is received through the reception portion and the channel is specified, associates the specified channel with a specific bus and outputs signals other than a signal of the associated channel from the specific bus.

Abstract: Disclosed herein are example techniques for multimedia experience based on biometric data. An example implementation may involve receiving first biometric data representing one or more first biological characteristics of an individual. After receiving the first biometric data, the example implementation may involve correlating the one or more first biological characteristics of the individual with a listening state of the individual. The example implementation may further involve receiving second biometric data representing one or more second biological characteristics of the individual. After receiving the second biometric data, the example implementation may involve determining that the one or more second biological characteristics corresponds to the one or more first biological characteristics.

Abstract: In one example of the disclosure, subject audio data, indicative of a subject audio captured via a transducer during an operation of a subject electromechanical device, is accessed. Baseline audio data, indicative of a baseline audio captured during an operation of a baseline electromechanical device, is accessed. An electronic file that includes differences between the subject audio data and the baseline audio data is created.

Abstract: Systems and methods for detecting a tampering and identifying a location of a digital recording are provided. A frequency sequence and a phase angle sequence may be extracted from the digital recording. A portion of the frequency sequence may be matched to one of a plurality of reference frequency sequences, and a portion of the phase angle sequence may be matched to one of a plurality of reference phase angle sequences. Tampering of the digital recording may be detected when the frequency and phase sequences differ from the matched reference sequences. Moreover, a noise sequence may be extracted from the extracted frequency sequence. A location of the digital recording may be identified by matching the noise sequence to one of a plurality of noise sequences of the plurality of reference frequency sequences.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: An audio system with an audio playback device that is configured to operatively connect to a digital audio source for playback of content from the digital audio source and a software program configured to run on a separate computer, wherein the software program provides a user interface configured to present to a first authorized user a selection of whether the audio playback device will be public and thus available to other authorized users or private and thus not available to other authorized users. In response to the first authorized user making a selection of whether the audio playback device will be public or private, an audio playback device access control is provided, wherein the access control is configured such that if the audio playback device is public, other authorized users are able to access and control the public audio playback device and if the audio playback device is private, other authorized users are not able to control the private audio playback device.

Abstract: Method and devices for processing audio signals based on sound profiles are provided. A sound profile can include data related to haptic movement of the audio data which is specific to a left ear or a right ear, demographic information, ethnicity information, age information, location information, social media information, intensity score of the audio data, previous usage information, or device information. A sound profile can be customized for individual user to include the inaudible frequency range at high frequency end and low frequency end. Audio data within the inaudible frequency range can be compensated by haptic movement corresponding to the inaudible frequency range. A sound profile can further include an audio frequency range and its lowest audible volume for a user. A sound profile can be provided to a user without any action from the user's part.

Abstract: The present invention is a method and a computer program for allowing individual artists to produce a latency corrected collaborative audio work by providing a first artist the opportunity to create a base track which may then be accessed by additional artists who can add audio tracks which harmonize with the base track. Artists may add synchronized tracks to the base track at any time from a remote location thereby allowing artists in different geographical locations to produce a collaborative work of art simultaneously or over an extended time period. The present invention includes a novel method for latency correction, which corrects for delays caused by various types of user equipment, and ensures each track is properly synchronized with the others. The present invention also includes a novel interface where a slot machine or slider user interface allows for ease of use from a browser.