Abstract: The disclosure is directed to a process that can predict an audio glitch, and then attempt to preempt the audio glitch. The process can monitor the systems, processes, and execution threads on a larger system or device, such as a mobile device or an in-vehicle device. Using a learning algorithm, such as deep neural network (DNN), the information collected can generate a prediction of whether an audio glitch is likely to occur. An audio glitch can be an audio underrun condition. 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 underrun condition has abated, the system adjustments can be reversed fully or in steps to return the system to its state prior to the system adjustments.

Abstract: In an example implementation, a method may involve sending a message that arranges a media playback system into a pre-defined state, wherein a first playback device is in a first zone having a first zone name, a second playback device is in a second zone having a second zone name, and a queue has a particular set of media items. The method may also involve detecting a trigger indicating a request to reset the media playback system. The method may further involve causing the media playback system to revert to the pre-defined state, wherein reverting may include assigning the first zone name to the first zone and joining the first playback device to the first zone, assigning the second zone name to the second zone and joining the second playback device to the second zone, and configuring the queue with the particular set of media items.

Abstract: According to an embodiment, a noise cancellation method comprises updating a first far-end channel filter according to a first variance yielded based on an input signal and a first far-end signal included in the input signal, providing a first output signal obtained by removing a first far-end noise signal from the input signal through the first far-end channel filter, updating a second far-end channel filter according to a second variance yielded based on the first output signal and a second far-end signal included in the input signal, and providing a second output signal obtained by removing a second far-end noise signal from the first output signal through the second far-end channel filter.

Abstract: A dialog device for executing a conversation with a user using a conversational sentence generated from content information is provided. The dialog device includes an information acquisition unit that acquires the content information from an information source, and changes the content information used in generation of the conversational sentence, and continues the conversation with the user. The dialog device sets multiple attribute information pieces with respect to specific content information used in generation of the conversational sentence. The dialog device controls the information acquisition unit to acquire other content information, having at least a part of attribute information pieces equal to the specific content information, as a candidate used in generation of the conversational sentence next to the specific content information.

Abstract: Apparatus, systems, and methods are disclosed for cancellation and content recognition of audio received over a High-Definition Multimedia Interface (HDMI)/Audio Return Channel (ARC) port of a display device. The media device receives audio over the HDMI/ARC port from the display device, processes the audio by identifying, decoding, re-encoding, and fingerprinting the audio with identification information, and subsequently streams the audio within the media streaming environment, including to external speakers, such as voice-enabled speakers. Based on the identification information and recognized content of the processed audio, the media device offers additional content for streaming to a user. The media device continuously monitors for spoken voice received over a voice-enabled speaker while the media device simultaneously receives, processes, and streams audio from the display device.

Abstract: Some examples include receiving audio content through a microphone of an electronic device and determining whether embedded data is included in the received audio content. The electronic device may decode the received audio content to extract the embedded data. In addition, the electronic device may perform at least one of: sending a communication to a computing device over a network based on the extracted embedded data, or presenting information on a display of the electronic device based on the extracted embedded data.

Abstract: A method of operation of a navigation system includes: detecting a message overlap between a current audio message and a message delivery timeframe of a contemporaneous audio message; determining an audio message interchange between the current audio message and a contemporaneous audio message based on a message waveform of the current audio message, a remaining message duration of the current audio message, or a combination thereof; and delivering the current audio message and the contemporaneous audio message based on the audio message interchange.

Abstract: Aspects of the subject disclosure may include, for example, a method for spherical video streaming over a network by a processing system including a processor, the method including prefetching audio data for rendering the spherical video; determining whether there is a single source or a plurality of sources of audio in the audio data; responsive to a determination that there is only one dominant source of the audio in the audio data, prefetching video data in a field of view centered on spatial coordinates of the dominant source of audio; responsive to a determination that there is the plurality of sources of the audio in the audio data, sorting the plurality of sources of the audio according to an estimated sound source energy for each source in the plurality of sources, thereby creating a sorted list of the plurality of sources; and prefetching video data in plural fields of view centered on spatial coordinates of each audio source in the sorted list of the plurality of sources, wherein the prefetching is

Abstract: Methods and devices for correcting warping in spatial audio may include identifying a geometric transform that defines a geometric warping between a first spatial geometric model that represents how sound is produced in a first volumetric space and a second spatial geometric model that represents how sound is produced in a second volumetric space different from the first volumetric space. The methods and devices may include determining an inverse of the geometric transform that compensates for the geometric transform. The methods and devices may include applying the inverse of the geometric transform to a first location in the first spatial geometric model by mapping the first location to a second location in the second spatial geometric model to correct for the geometric warping.

Abstract: This disclosure concerns the playback of audio, e.g. in the form of music. More particularly, the disclosure concerns the playback of streamed audio. In one example embodiment, there is a method of operating an electronic device for playback of an audio stream. While an audio stream is being played, for example at a user interface of the electronic device, a first request to adjust an energy level (e.g., a tempo) of the played audio stream is received. In response to receiving the first request, the energy level (e.g., the tempo) of the played audio stream is adjusted.

Abstract: A system and method for disabling adaptation in an adaptive feedforward control system at low speeds. The method includes generating a noise signal representative of undesired noise detected by a noise sensor of a vehicle and generating a noise-cancellation signal via a controller within the vehicle. Residual noise resulting from the combination of the acoustic energy of the noise-cancellation signal and the undesired noise is detected by a reference sensor, which generates an error signal based on the residual noise. The error signal and a speed signal from a speed sensor on the vehicle are transmitted to an adaptive processing module for the generation of a filter update signal. The adaptive processing module selectively permits or prevents the filter update signal to adapt filter coefficients of a filter when the speed signal is within a set of conditions.

Abstract: Systems, devices, apparatuses, components, methods, and techniques for saving media content to a context for later playback are provided. An example media-playback device for identifying and playing media content for a user traveling in a vehicle includes a context detecting device, a context-driven playback engine, and a media playback engine. Contexts are established by parameters that can be detected by a media-playback device. Contexts are situations that are defined by one or more locations, times, events, activities, people, and devices. Media content is saved to the contexts for later playback. The contexts are detected by the context detecting device, the associated media content is identified by the context-driven playback engine, and the media content is automatically played through the media playback engine, without additional input required by the user.

Abstract: Example systems and techniques disclosed herein facilitate interoperability between different media playback systems referred to herein as a virtual line-in (VLI) media playback system and a native playback system. When a VLI session is created by a VLI sender, a first native playback device can join a VLI group as a VLI receiver. As a VLI receiver, the first native playback device receives audio content and playback commands from the VLI sender to facilitate synchronous playback with other VLI receivers. At the same time, this native playback device can concurrently operate as a native domain group coordinator of a native domain synchrony group. As the native domain group coordinator, the native playback device translates VLI domain audio, control, and timing signals into the native domain and distributes such signals to native domain group members. In this way, the native domain group members can synchronize their playback with the VLI group.

Abstract: A natural language processing system that includes an artificial intelligence (AI) engine, a tagging engine, and a resource allocation engine. The AI engine is configured to receive a set of audio files and to identify concepts within the set of audio files. The AI engine is further configured to determine a usage frequency for each of the identified concepts and to generate an AI-defined tag for concepts with a usage frequency that is greater than a usage frequency threshold. The tagging engine is configured to receive an audio file and to modify metadata for the audio file to include AI-defined tags. The resource allocation engine is configured to identify a storage location from among the plurality of storage devices based on tags associated with the audio file and send the audio file to the identified storage location.

Abstract: This application relates to methods and apparatus for transfer of multiple digital data streams, especially of digital audio data over a single communications link such as a single wire. The application describes audio interface circuitry comprising a pulse-length-modulation (PLM) modulator (204). The PLM is responsive to a plurality of data streams (PDM-R, PDM-L), to generate a series of data pulses (PLM) with a single data pulse having a rising and falling edge in each of a plurality of transfer periods defined by a first clock signal (TCLK). The timing of the rising and falling edge of each data pulse is dependent upon on a combination of the then current data samples from the plurality of data streams. The duration and position of the data pulse in the transfer window in effect defines a data symbol encoding the data. Circuitry for receiving and extracting the data is also disclosed.

Abstract: In an apparatus configured to perform signal processing on audio data of a first sampling rate, methods disclosed herein comprise receiving audio data of a second sampling rate, the second sampling rate being higher than the first sampling rate. The methods comprise applying filtering to the audio data of the second sampling rate to thereby produce first filtered audio data and second filtered audio data, the first filtered audio data comprising mainly component frequencies which are audible to the human ear, the second filtered audio data comprising mainly components frequencies which are substantially inaudible to the human ear. The methods further comprise applying first signal processing to the first filtered audio data; and applying second signal processing to the second filtered audio data, the second signal processing having a lower computational complexity than the first signal processing. Corresponding apparatus and computer readable media are also disclosed herein.

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: A music service application that can be run on a wireless mobile device enables audio data to be progressively downloaded from a remote server and also enables locally stored data to be played efficiently. Audio content that is relevant to a user is identified and downloaded to the user's mobile device, in some cases with minimal or no effort by the user. Continuous play features ensure that the user can experience an uninterrupted music experience, both in online and offline modes. Social features such as playlists and preferences of other users are leveraged, to provide users with popular music that is relevant to their interests.

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: A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals.