Abstract: Embodiments described herein involve generating and modifying a sub-index of a media index. An example method may involve receiving requests to play back media items in a media index, tracking the requests to play back the media items in the media index, determining a set of media items from among the media items in the media index based on the tracked requests to play back media items in the media index, and indexing the determined set of media items into a sub-index. The set of media items have been played at least a pre-determined threshold number of times within a pre-determined threshold period of time. The media items in the sub-index are a subset of the media items in the main index.

Abstract: Methods, systems, and computer programs are presented for managing audio files of a user to reduce latencies in play start times on local devices. The audio files are stored on cloud storage managed by a server. One method includes processing a plurality of audio files associated with a user, where the processing is configured to create audio snippet files from each of the plurality of audio files. The audio snippet files representing a beginning part of each of the plurality of audio files. The method also includes transmitting the audio snippet files to a client device and detecting a request from the client to begin playing a first audio file from the plurality of audio files of the user. The first audio file being stored on the cloud storage managed by the server.

Abstract: An electronic device may comprise audio processing circuitry and a first connector having a first contact and a second contact. In a first mode of operation, the audio processing circuitry is configured to output one or more audio signals carrying music and/or voice via the first contact and the second contact. In a second mode of operation, the audio processing circuitry is configured to output a signal for delivering supply current via the first contact and the second contact. While the electronic device is in the first mode of operation, a gain and/or volume limit of the audio processing circuitry may be set to a first level, and while the electronic device is in the second mode of operation, the gain and/or volume limit of the audio processing circuitry may be set to a second level that is higher than the first level.

Abstract: The present invention relates to the continuous monitoring of an audio signal and identification of audio items within an audio signal. The technology disclosed utilizes predictive caching of fingerprints to improve efficiency. Fingerprints are cached for tracking an audio signal with known alignment and for watching an audio signal without known alignment, based on already identified fingerprints extracted from the audio signal. Software running on a smart phone or other battery-powered device cooperates with software running on an audio identification server.

Abstract: A device for altering processing of a signal during processing of the signal has a display unit, an input unit and a processing unit. The display unit is configured for graphically displaying at least a predetermined interval to-be-processed-next of the signal. The input unit is configured for receiving at least one of: a selection input for selecting duration of a time period within the interval and a command input for commanding altering the processing using the time period. The processing unit is configured for processing the signal and for altering the processing in response to the command input. The input unit is touch-sensitive and the processing unit is configured to control the display unit for displaying graphically the time period at least while the input unit is touched.

Abstract: Techniques related to input and output signal synchronization and latency jitter compensation for audio systems are discussed. Such techniques may include determining a number of virtually buffered samples based on a detected latency between an audio capture thread and an audio playback thread and synchronizing an audio input signal and an audio output signal based on the number of virtually buffered samples.

Abstract: A parameter quantifying deviation from Gaussianity distribution of a patient's sounds, in the form of a non-Gaussianity distribution index, may be used to assist in diagnosis of sleep dysfunction such as OSAHS. A method for diagnosing a sleeping disorder of a subject includes processing a digitized audio signal from the subject with at least one electronic processor. The processing includes estimating a parameter quantifying deviation from Gaussianity distribution of the audio signal. in the form of a non-Gaussianity Index (NGI). The NGI value is then applied to a diagnostic model. The presence of a sleeping disorder is then indicated based on the output of the diagnostic model.

Abstract: A playback device includes a first buffer and a second buffer, each having storage regions, and a processing unit. The processing unit performs: a storage process that causes input audio data to be stored in the storage regions of the first buffer in order; a first playback process that causes the stored audio data to be played back in the order in which the audio data was stored; a designation process that designates, in response to a user input, at least one of the plurality of storage regions of the first buffer in which the audio data is stored; a copy process that causes the audio data stored in the designated storage region of the first buffer to be copied to the second buffer; and a second playback process that causes the audio data copied to the second buffer to be repeatedly played back.

Abstract: The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for implementing such anti-aliasing and/or anti-imaging filters with reduced computational complexity. In addition, the present document relates to further aspects of an audio encoding and decoding system, such as the phase relation between the channels of a multi-channel audio signal and/or the structure of the bitstream of an encoded audio signal.

Abstract: Disclosed are systems, methods and devices for improving audio processing and playing multiple sounds using fewer resources. Various implementations may create audio files, in some cases from a single sample of a type of sound, where the files represent specific quantities of that type of sound, such as 10 footsteps or 1000 claps. Given a desired target number of sounds to play, various implementations determine an appropriate combination of the employable audio files to represent the desired number of sounds, and play or render the combination as an approximation of the desired target number of sounds. By using single audio files that represent and sound like multiple instances of a specified type of sound, the various implementations use significantly fewer computing and audio resources than playing an equivalent number of audio files each of which contains a single instance of the specified type of sound.

Abstract: The present invention relates to a mobile audio player, comprising memory means adapted to store music data and non-music data, input means adapted to receive a user input with user preferences, radio program generating means adapted to generate, without a connection to an external entity, an individualized radio program from stored music data and non-music data depending on user preferences input via the input means, and audio output means adapted to output audio signals representing a generated radio program.

Abstract: Examples described herein involve determining an audio processing algorithm for a playback device to apply when playing audio content in a playback zone. Determining the audio processing algorithm may involve causing the playback device to play a first audio signal in the playback zone, receiving data indicating a second audio signal detected by a microphone of the playback device. Based on the second audio signal and a characteristic of the playback device, an audio processing algorithm may be determined. The audio processing algorithm may be calculated, or identified in a database based on an acoustic characteristic of the playback zone. The acoustic characteristic of the playback zone may be determined based on the second audio signal and the characteristic of the playback device. The audio processing algorithm may then be applied by the playback device.

Abstract: In general, the subject matter described in this disclosure can be embodied in methods, systems, and program products for identifying that a first audio stream includes first, second, and third sources of audio. A computing system identifies that a second audio stream includes the first, second, and third sources of audio. The computing system determines that the first and second sources of audio are part of a first conversation. The computing system generates a third audio stream that combines the first source of audio from the first audio stream, the first source of audio from the second audio stream, the second source of audio from the first audio stream, and the second source of audio from the second audio stream, and diminishes the third source of audio from the first audio stream, and the third source of audio from the second audio stream.

Abstract: Disclosed aspects relate to managing a set of devices using a set of acoustic emission data which indicates device-health of the set of devices. The set of devices is coupled with a set of acoustic emission sensors. Based on the set of acoustic emission data, a triggering event related to a first device of the set of devices is detected. Using the set of acoustic emission data, an event response which includes a first modification with respect to operation of the first device is determined. Establishment of the event response which includes the first modification with respect to operation of the first device is initiated.

Abstract: It is inter alia disclosed to select, by a first apparatus, one or more recording devices to obtain a set of selected recording devices, or to receive, at the first apparatus, information on a set of one or more selected recording devices. The selected recording devices in the set of selected recording devices are selected from a set of recording devices at least based on relevance levels determined for the recording devices of the set of recording devices. A combination of audio signals recorded by the selected recording devices of the set of selected recording devices in an audio scene to obtain one or more combined audio signals for rendering is performed or caused by the first apparatus.

Abstract: A system for exploiting visual information for enhancing audio signals via source separation and beamforming is disclosed. The system may obtain visual content associated with an environment of a user, and may extract, from the visual content, metadata associated with the environment. The system may determine a location of the user based on the extracted metadata. Additionally, the system may load, based on the location, an audio profile corresponding to the location of the user. The system may also load a user profile of the user that includes audio data associated with the user. Furthermore, the system may cancel, based on the audio profile and user profile, noise from the environment of the user. Moreover, the system may include adjusting, based on the audio profile and user profile, an audio signal generated by the user so as to enhance the audio signal during a communications session of the user.

Abstract: Examples described herein involve maintaining a database of audio processing algorithms. Maintaining the database may involve generating or updating audio processing algorithm entries. In one example, generating an audio processing algorithm may involve a computing device causing a playback device to play a first audio signal in a playback zone, receiving (i) data indicating one or more characteristics of a playback zone, and (ii) data indicating a second audio signal detected by a microphone of the playback device in the playback zone. Based on the second audio signal and a characteristic of the playback device, an audio processing algorithm may be determined. The an association between the determined audio processing algorithm at least one of the one or more characteristics of the playback zone may be stored in the database.

Abstract: An example method involves monitoring, by a computing device, a communications feed for an indication of media and detecting, in the communications feed, the indication of the media. The method may further involve identifying at least one media item corresponding to the indication of the media and causing a playback queue of a media playback system to include one or more of the identified at least one media item.

Abstract: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

Abstract: A method and system for streaming a soundtrack from a server to a remote user device for a reader of electronic media content. The soundtrack is defined by multiple audio regions. Each audio region defined by an audio track for playback in the audio region, a start position in the electronic media content corresponding to where the playback of the audio region is to begin, and a stop position in the electronic media content corresponding to where the playback of the audio region is to cease. The streaming of the soundtrack is based on control data generated by the remote user device.