Abstract: The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Abstract: Gesture-responsive modification of a generated sound field is described.

Abstract: This control device for controlling a loudspeaker (14) in a loudspeaker enclosure, comprises: an input for an audio signal to be reproduced; a supply output for supplying an excitation signal for the loudspeaker; the calculation means (26, 36, 38, 70, 71, 80, 90) for calculating, at each time instant (t), at least one predicted current (iref(t)) for the excitation signal for the loudspeaker (14) as a function of the audio signal. It comprises an attenuator (71) that is capable of limiting the predicted current to a limited current value that is lower than a ceiling value by application, to the predicted current, of an attenuation gain which is a function of the predicted current.

Abstract: An FFT processor applies the fast Fourier transform to each frame of an input signal and generates frequency domain data. A first spectrum calculator calculates the amplitude spectrum of the input signal for the individual frequency components based on the frequency domain data. A dispersion index calculator calculates a dispersion index indicating the level of dispersion of the amplitude spectrum of the input signal in the latest frames. A noise index calculator calculates a noise index indicating the level of noise contained in the input signal. A second spectrum calculator calculates the amplitude spectrum of the noise for the individual frequency components. A gain calculator calculates a spectral gain for suppressing the noise for the individual frequency components. An IFFT processor applies the inverse fast Fourier transform to the frequency domain data after multiplication by the spectral gain at a multiplier, and generates a signal to be output.

Abstract: Correcting typographical errors in electronic text may include converting a text message containing at least one phonemic spelling of a word into speech by running a text-to-speech application programming interface (API) with the text message as input. The converted speech may be input to a speech-to-text API and the speech-to-text API executed to convert the speech to text. A text file comprising the text may be generated and/or output. The text file automatically contains a corrected version of the phonemic spelling of the word in text message.

Abstract: A vehicle sound effect generation apparatus generates a sound effect of an engine based on a vibration sound database including a fundamental wave sound having a fundamental frequency component and a plurality of adjustment wave sounds having a frequency component other than the fundamental frequency component. The vehicle sound effect generation apparatus includes a running state detecting unit that detects a running state of a vehicle, a sound effect generation unit that synthesizes the fundamental wave sound with the adjustment wave sounds based on the running state of the vehicle, and a visual guidance direction setting unit that sets a visual guidance direction at which a vehicle occupant shoots a look. The sound effect generation unit causes output characteristics of a sound effect in the visual guidance direction to increase more than output characteristics of a sound effect in a direction other than the visual guidance direction.

Abstract: A training method for multiple personalized acoustic models, and a voice synthesis method and device, for voice synthesis. The method comprises: training a reference acoustic model, based on first acoustic feature data of training voice data and first text annotation data corresponding to the training voice data (S11); acquiring voice data of a target user (S12); training a first target user acoustic model according to the reference acoustic model and the voice data (S13); generating second acoustic feature data of the first text annotation data, according to the first target user acoustic model and the first text annotation data (S14); and training a second target user acoustic model, based on the first text annotation data and the second acoustic feature data (S15).

Abstract: The present invention discloses a band-pass acoustic filter and an acoustic sensing apparatus. The band-pass acoustic filter including at least two MEMS microphone chips and an ASIC chip, wherein the output signals of the MEMS microphone chips are processed in the ASIC chip after being coupled.

Abstract: A first example playback device includes one or more processors and a non-transitory computer-readable medium storing instructions that, when executed by the one or more processors, cause the first playback device to perform functions. The functions include generating a mixed audio signal comprising one or more components of a first audio stream and one or more components of a second audio stream; and playing the mixed audio signal, where the one or more components of the second audio stream of the mixed audio signal played by the first playback device are played in synchrony with a second playback device configured to play the second audio stream. An example non-transitory computer-readable medium and an example method, both related to the first example playback device, are also disclosed herein.

Abstract: A localized sound projection system can coordinate the sounds of speakers to simulate the placement of an auditory cue in a 3D space. The system can include a plurality of speakers configured to output auditory signals and a sound localization controller configured to control the plurality of speakers to coordinate the auditory signals to simulate an origination location of a patient alarm. The sound localization controller can determine adjusted auditory signals and control a plurality of speakers to output the plurality of adjusted auditory signals. A method for dynamically controlling speaker settings in a medical environment can include determining volume settings corresponding to a speaker, monitoring a level of ambient noise corresponding to a room of a patient, controlling the volume settings of the speaker to reduce or increase a sound level output of a speaker. A patient monitoring system can be configured to physically manipulate medical devices in response to audible commands.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: Disclosed is a mounting assembly for use in a mobile terminal. The mobile terminal includes a main frame on which a motherboard is mounted. The mounting assembly includes a rack for mounting thereon at least one of a speaker or a vibrator, and the rack is detachably connectable to the main frame. By means of the mounting assembly, the speaker or the vibrator is not directly mounted on the motherboard of the mobile terminal.

Abstract: The disclosure includes a headset comprising one or more earphones including one or more sensing components. The headset also includes one or more voice microphones to record a voice signal for voice transmission. The headset also includes a signal processor coupled to the earphones and the voice microphones. The signal processor is configured to employ the sensing components to determine a wearing position of the headset. The signal processor then selects a signal model for noise cancellation. The signal model is selected from a plurality of signal models based on the determined wearing position. The signal processor also applies the selected signal model to mitigate noise from the voice signal prior to voice transmission.

Abstract: Speaker protection may be based on multiple speaker models with oversight logic that controls the speaker protection based on the multiple speaker models. At least one of the speaker models may be based on a speaker excursion determined from feedback information from the speaker, such as a current or voltage measured at the speaker. Excursion based on the speaker feedback may be used to determine an error in an excursion prediction made from the audio signal. The excursion prediction may then be compensated for that error. In some embodiments, a direct displacement estimate of excursion generated from speaker monitor signals is used to correct a fixed excursion model applied to an input audio signal.

Abstract: Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: A modified end user network access point includes a switching module and a controller to track usage of the end user network access point by wirelessly connected end user devices. The tracked usage information is transmitted from the end user network access point to a remotely coupled data center. The data center collects usage information from a plurality of different end user network access points. The collected usage information is in turn transmitted from the data center to a network carrier. The network carrier uses the received usage information to bill users for accessing the end user network access points and using any services provided by the end user network access points. An owner of each end user network access point is compensated by the network carrier each instance that an end user device utilizes the end user network access point to access the network carrier's communications network.

Abstract: A speaker system is disclosed for providing customised acoustical wavefronts with vertical and horizontal pattern control and amplitude and phase control. The system including a speaker housing (1) having therein at least a first array (2) of high frequency driver segments (3) and at least a secondary array (4) of low frequency driver segments (5) disposed behind said first array (2), said first array having sufficient space between said driver segments (3) to allow acoustic transparency whereby a wavefront from said secondary array (4) can substantially pass through said first array (2).

Abstract: An example apparatus for detecting loudspeaker faults includes a switch detector to detect that a loudspeaker fault detection switch is enabled. The apparatus also includes an audio receiver to receive an echo energy threshold and audio from a microphone in a test area. The apparatus further includes an audio analyzer to detect that the echo energy threshold is not exceeded by an acoustic echo in the captured audio. The apparatus includes a fault generator to generate a loudspeaker fault in response to detecting the acoustic echo does not exceed the echo energy threshold.