Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: A method of providing a haptic output includes detecting a condition; determining if a head-mounted haptic accessory comprising an array of two or more haptic actuators is being worn by a user; determining an actuation pattern for the array of haptic actuators; and in response to detecting the condition and determining that the head-mounted haptic accessory is being worn by the user, initiating the actuation pattern to produce a directional haptic output that is configured to direct the user's attention along a direction.

Abstract: A method and system for a headset with programmable modes, where the headset comprises a beamforming microphone: sensing an ambient sound level near the headset, and configuring the headset in one of a plurality of modes by configuring a beam pattern of the beamforming microphone based on at least the sensed ambient noise level. A user of the headset may configure the headset in the one of the plurality of modes or it may be automatically configured. The beamforming microphone may comprise an array of sound sensing elements. The headset may be configured in a quiet mode when the sensed ambient sound level is below that of a desired sound source and may be configured with a wide beam pattern. The headset may be configured in a loud mode when the sensed ambient sound level is above that of a desired sound source.

Abstract: A method, computer program product, and computing system for defining model representative of a plurality of acoustic variations to a speech signal, thus defining a plurality of time-varying spectral modifications. The plurality of time-varying spectral modifications may be applied to a plurality of feature coefficients of a target domain of a reference signal, thus generating a plurality of time-varying spectrally-augmented feature coefficients of the reference signal.

Abstract: Systems and methods for speech recognition are provided. Some methods can include detecting a relative location of a user within one of a plurality of voice pickup areas of a multi-directional microphone array and the multi-directional microphone array activating a setting of the multi-directional microphone array based on the relative location of the user within the one of the plurality of voice pickup areas.

Abstract: A stereo signal processing method includes performing delay estimation on a stereo signal of a current frame to determine an inter-channel time difference of the current frame, identifying a sign of the inter-channel time difference of the current frame is different from a sign of an inter-channel time difference of a previous frame of the current frame, performing delay alignment processing on the first-channel signal of the current frame based on the inter-channel time difference of the current frame, and performing delay alignment processing on the second-channel signal of the current frame based on the inter-channel time difference of the previous frame.

Abstract: A sound pickup method includes splitting a sound pickup signal of a microphone of which sound pickup in a predetermined range is blocked by an acoustic obstacle, into a plurality of sound pickup signals and subjecting at least one sound pickup signal among the plurality of sound pickup signals to band limitation processing, and comparing characteristic amounts of the plurality of sound pickup signals including the sound pickup signal after being subjected to the band limitation processing and controlling a gain of the microphone in accordance with a result of comparing the characteristic amounts.

Abstract: Methods and systems are provided for automatic noise control. Automatic noise control includes controlling a shadow noise control transfer function based on a shadow error signal and a filtered or unfiltered reference signal, generating the shadow error signal based on a filtered or unfiltered shadow anti-noise signal and an error signal, and substituting the noise control transfer function by the shadow noise control transfer function if the shadow error signal is smaller than the error signal.

Abstract: A method and device for processing information are provided. The method for processing information includes: determining a current output voltage of a power supply in a terminal device; determining a target parameter of an audio processing circuit in the terminal device according to the current output voltage; and configuring the audio processing circuit for processing an audio signal according to the target parameter. With the embodiments of the present disclosure, an impact of a change of the output voltage of the power supply on output volume of a loudspeaker can be reduced, thereby improving user experience.

Abstract: Implementations are directed to causing a computing device to transmit an audio data stream, via a communication channel, to a vehicle computing device of a vehicle. Various implementations include one or more vehicle speakers driven by the vehicle computing device to render audible output. Additionally or alternatively, various implementations include determining whether the audible output is captured by at least one microphone of a vehicle interface device within the vehicle. In many implementations, the vehicle interface device is communicatively coupled to the computing device.

Abstract: This disclosure provides a decoding method, and a decoding apparatus for a stereo signal. The decoding method includes: decoding a bitstream to obtain a first channel signal, a second channel signal, and a first ITD of a current frame of a stereo signal; performing a mixing processing on the first channel signal and the second channel signal, to obtain a third channel reconstructed signal and a fourth channel reconstructed signal; performing interpolation processing based on the first ITD and a second ITD of a previous frame previous to the current frame, to obtain a third ITD; and adjusting a delay of the third channel reconstructed signal and the fourth channel reconstructed signal based on the third ITD.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: Low-latency audio networking is disclosed. In one embodiment, an example playback device includes a processor and memory having stored thereon instructions executable by the processor. The example instructions are to cause the first playback device to perform functions comprising: receiving audio information; selecting a first frequency channel of a first spectrum based on a threshold latency associated with the audio information; transmitting to the second playback device via a second frequency channel of a second spectrum, control information that identifies the first frequency channel of the first spectrum; and transmitting to the second playback device via the first frequency channel of the first spectrum, the audio information to be played by the second playback device.

Abstract: The present disclosure relates to time-domain stereo parameter encoding methods and apparatus. One example time-domain stereo parameter encoding method includes determining a channel combination scheme for a current frame, determining a time-domain stereo parameter of the current frame based on the channel combination scheme for the current frame, and encoding the determined time-domain stereo parameter of the current frame, where the time-domain stereo parameter includes at least one of a channel combination ratio factor or an inter-channel time difference.

Abstract: An encoding parameter adjustment method is performed at a computer device. The method includes: obtaining a first audio signal, and determining a psychoacoustic masking threshold within a service frequency band in the first audio signal; obtaining a second audio signal, and determining a background environmental noise estimation value of the frequency within the service frequency band in the second audio signal; determining a masking tag corresponding to the service frequency band according to the psychoacoustic masking threshold of the first audio signal and the background environmental noise estimation value of the second audio signal; determining a masking rate of the service frequency band according to the masking tag corresponding to the frequency within the service frequency band; determining a first reference bit rate according to the masking rate of the service frequency band; and configuring an encoding bit rate of an audio encoder based on the first reference bit rate.

Abstract: The present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

Abstract: A method for processing a natural language communication includes a processor receiving a natural language communication and identifying, in the natural language communication, an ordered sequence of a plurality of word spellings of a natural human language. The processor performs constraint-based parsing of the natural language communication by reference to a consistent meaning framework (CMF) graph including a plurality of nodes each corresponding to a respective word meaning. The parsing includes scanning the phrase in multiple different parse scan directions and determining a respective meaning associated with each of multiple word spellings in a phrase of the natural language communication by reference to multiple constraints provided by multiple nodes in the CMF graph respectively corresponding to the word meanings.

Abstract: A consistent meaning framework (CMF) graph including a plurality of nodes linked by a plurality of edges is maintained in data storage of a data processing system. Multiple nodes among the plurality of nodes are meaning nodes corresponding to different word meanings for a common word spelling of a natural language. Each of the multiple word meanings has a respective one of a plurality of associated constraints. A natural language communication is processed by reference to the CMF graph. The processing includes parsing the natural language communication and selecting, for each of multiple word spellings in the natural language communication, a selected word meaning from among the word meanings provided by the CMF graph. The selected word meaning for each of the multiple word spellings in the natural language communication is recorded in data storage.

Abstract: Systems, devices, apparatuses, components, methods, and techniques for providing media content to a media playback system are provided. The techniques provide for determining whether a playback system is connected, or otherwise associated with, a playback device suitable for 3D audio playback. Upon determining that a playback system is or is not connected or otherwise associated with a playback device suitable for 3D audio playback, audio content in a corresponding format may then be transferred and/or played by a requesting system. In some examples, one or more filters may be applied to requested audio content to compensate for determined user head movement and/or to create simulated 3D audio from generic two-channel recording.

Abstract: Methods for training a neural sequence-to-sequence (seq2seq) model. A processor receives the model and training data comprising a plurality of training source sequences and corresponding training target sequences, and generates corresponding predicted target sequences. Model parameters are updated based on a comparison of predicted target sequences to training target sequences to reduce or minimize both a local loss in the predicted target sequences and an expected loss of one or more global or semantic features or constraints between the predicted target sequences and the training target sequences given the training source sequences. Expected loss is based on global or semantic features or constraints of general target sequences given general source sequences.