Abstract: A method for improving voice quality is provided herein. The method includes receiving acoustic signals from a microphone array; receiving sensor signals from an accelerometer sensor of the headset; generating, by a beamformer, a speech output signal and a noise output signal according to the acoustic signals; best-estimating the speech output signal according to the sensor signals to generate a best-estimated signal; and generating a mixed signal according to the speech output signal and the best-estimated signal.

Abstract: Disclosed are an audio encoding method, to which BRIR/RIR parameterization is applied, and a method and device for reproducing audio by using parameterized BRIR/RIR information.

Abstract: A sensor is configured to infer a rotational speed of a tire of a vehicle. A frequency generator is configured to synthesize frequencies of a tire cavity resonance according to the rotational speed of the tire to generate a sense signal. An active noise control filter is configured to generate an antinoise signal from the sense signal. A loudspeaker configured to convert the antinoise signal into antinoise and to radiate the antinoise to a listening position. The antinoise signal is configured so that the antinoise reduces sound of the tire cavity resonance at the listening position.

Abstract: A stereo signal processing method and apparatus, where the 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: A multi-microphone algorithm for detecting and differentiating interference sources from desired talker speech in advanced audio processing for smart home applications is described. The approach is based on characterizing a persistent interference source when sounds repeated occur from a fixed spatial location relative to the device, which is also fixed. Some examples of such interference sources include TV, music system, air-conditioner, washing machine, and dishwasher. Real human talkers, in contrast, are not expected to remain stationary and speak continuously from the same position for a long time. The persistency of an acoustic source is established based on identifying historically-recurring inter-microphone frequency-dependent phase profiles in multiple time periods of the audio data. The detection algorithm can be used with a beamforming processor to suppress the interference and for achieving voice quality and automatic speech recognition rate improvements in smart home applications.

Abstract: There is provided a horn device for a railcar capable of delivering horn sound to a long distance in a traveling direction and being less likely to spread sound in a car width direction. A horn device for a railcar according to one aspect of the present invention includes: a whistle portion that is supplied with compressed air to make horn sound in a traveling direction; a duct located in the traveling direction of the whistle portion and having an opening portion at an end portion in the traveling direction; and at least one partition plate that is disposed perpendicularly to a car width direction, extends in a car longitudinal direction, and divides the interior of the duct.

Abstract: Some disclosed systems and methods include a surround sound controller and one or more wireless headphones that switch between operating in various modes. In a first mode, the surround sound controller uses a first MCS to transmit first surround sound audio information to a first pair of headphones. In a second mode, the surround sound controller uses a second MCS to transmit (a) the first surround sound audio information to the first pair of headphones and (b) second surround sound audio information to a second pair of headphones. In operation, the first MCS corresponds to a lower data rate at a higher wireless link margin than the second MCS.

Abstract: There is provided a voice processing system capable of acquiring user voice more clearly, the voice processing system including: a mounting unit configured to be attached to a user. The mounting unit includes at least three voice acquisition units configured to acquire voice data for beamforming.

Abstract: The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

Abstract: The present document describes a method (600) for estimating source parameters of audio sources (101) from mix audio signals (102), with. The mix audio signals (102) comprise a plurality of frames. The mix audio signals (102) are representable as a mix audio matrix in a frequency domain and the audio sources (101) are representable as a source matrix in the frequency domain. The method (600) comprises updating (601) an un-mixing matrix (221) which is configured to provide an estimate of the source matrix from the mix audio matrix, based on a mixing matrix (225) which is configured to provide an estimate of the mix audio matrix from the source matrix. Furthermore, the method (600) comprises updating (602) the mixing matrix (225) based on the un-mixing matrix (221) and based on the mix audio signals (102). In addition, the method (600) comprises iterating (603) the updating steps (601, 602) until an overall convergence criteria is met.

Abstract: A system is provided for recording electronic data records including an interactive audio interface configured to receive a real time voice command, parse the real time voice command for keywords and values and trigger the values to be stored; an external database configured to store the values parsed from the real time voice command and associate a time stamp with the stored values; a secure file transfer tool configured to transfer the stored values and time stamp to a secure datacenter; and an electronic records module in the secure datacenter configured to store the values in corresponding data records. Related methods and systems are also provided herein.

Abstract: An information handling system may include an air mover configured to cause movement of gaseous fluid within the information handling system in order to thermally cool one or more components of the information handling system and a dynamic noise generation system configured to receive an indication of expected acoustical energy generated by the air mover and generate sound in accordance with an output noise signal based on noise generated by the air mover.

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: Techniques for configuring device-specific skills as top-level skills are described. When a system receives a user input the system performs NLU processing to determine an intent of the user input. In some instances, the system may identify a device-specific skill associated with the device interacted with by the user. At least partially in parallel to performing NLU processing to determine the intent of the user input, the system may also perform NLU processing to determine a likelihood that the user input corresponds to an intent actionable by the device-specific skill. Once the system has finished NLU processing, the system may implement one or more prioritization rules to determine whether the user input should be sent to the device-specific skill or another skill of the system.

Abstract: In a method for controlling an audio system of a vehicle, an intention to communicate and/or a voice of at least one of a specific, occupant of the vehicle and/or of an occupant on a specific seat of the vehicle, are/is sensed, and at least one audio signal of the vehicle is changed as a function of the sensed intention to communicate and/or the voice of the occupant.

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: A multi-channel decorrelator for providing a plurality of decorrelated signals on the basis of a plurality of decorrelator input signals is configured to premix a first set of N decorrelator input signals into a second set of K decorrelator input signals, wherein K<N. The multi-channel decorrelator is configured to provide a first set of K? decorrelator output signals on the basis of the second set of K decorrelator input signals. The multi-channel decorrelator is further configured to upmix the first set of K? decorrelator output signals into a second set of N? decorrelator output signals, wherein N?>K?. The multi-channel decorrelator can be used in a multi-channel audio decoder. A multi-channel audio encoder provides complexity control information for the multi-channel decorrelator.

Abstract: The present disclosure relates to an information processing apparatus and an information processing method for enabling provision of a more convenient speech recognition service. The information processing apparatus includes a speech recognition unit that performs speech recognition for speech information based on an utterance of a user, and a correction portion estimation unit that collates content of a sentence obtained as a speech recognition result with collation information necessary for determining accuracy of the content to estimate, for the sentence, a correction portion that requires correction. The sentence obtained as a speech recognition result is displayed together with the correction portion estimated by the correction portion estimation unit and presented to the user.

Abstract: System, method, and apparatus including receiving an identification of content played by a playback system and an identification of location of the playback system. A playlist of content is associated with the location. A request is received to retrieve the playlist of content and the requested playlist of content is sent to a playback system.