Provided is a technique to generate a call environment that prevents call contents from being heard by a person other than a person speaking on the phone in a case where call voice is output from a speaker. Speakers installed in an automobile are denoted by SP.sub.1, ..., SP.sub.N, a first filter coefficient used to generate an input signal for a speaker SP.sub.n is denoted by F.sub.n (ω), and a second filter coefficient that is different from the first filter coefficient and is used to generate an input signal for the speaker SP.sub.n is denoted by .sup.~F.sub.n (ω). A call environment generation method includes: an acoustic signal generation step of generating, when detecting a start signal of a call, a call-time acoustic signal that is obtained by adjusting volume of an acoustic signal to be reproduced during the call, by using a predetermined volume value; a first local signal generation step of generating a sound signal S.sub.n as an input signal for the speaker SP.sub.n from a voice signal of the call by using the first filter coefficient F.sub.n (ω); and a second local signal generation step of generating an acoustic signal A.sub.n as an input signal for the speaker SP.sub.n from the call-time acoustic signal by using the second filter coefficient .sup.~F.sub.n (ω).

In some embodiments, a method for performing at least one of enhancement, decoding, or rendering of a multichannel audio signal in response to compression feedback or feedback from a smart amplifier. For example, the compression feedback may be indicative of amount of compression applied to each of multiple frequency bands, of the audio signal or an enhanced audio signal generated in response thereto. The enhancement (e.g., bass enhancement) may include dynamic routing of audio content of the input audio signal between channels of an enhanced audio signal generated in response thereto. The enhancement and compression may be performed on a per speaker class basis. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook or laptop computer, tablet, soundbar, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

The present disclosure discloses an assistive listening device. The assistive listening device includes a signal input module configured to receive an initial sound and convert the initial sound into an electric signal, a signal processing module configured to process the electric signal and generate a control signal, and at least one output energy converter configured to convert the control signal into a bone conduction sound wave that can be perceived by a user and an air conduction sound wave that can be heard by the user's ears. Within a target frequency range, the air conduction sound wave is transmitted to the user's ears, so that a sound intensity of the air conduction sound heard by the user's ears is greater than a sound intensity of the initial sound received by the signal input module.

The present disclosure provides an acoustic output apparatus including one or more status sensors, at least one low-frequency acoustic driver, at least one high-frequency acoustic driver, at least two first sound guiding holes, and at least two second sound guiding holes. The status sensors may detect status information of a user. The low-frequency acoustic driver may generate at least one first sound, a frequency of which is within a first frequency range. The high-frequency acoustic driver may generate at least one second sound, a frequency of which is within a second frequency range including at least one frequency exceeding the first frequency range. The first and second sound guiding holes may output the first and second spatial sound, respectively. The first and second sound may be generated based on the status information, and may simulate a target sound coming from at least one virtual direction with respect to the user.

Systems and methods are described for generating a transcript of a legal proceeding or other multi-speaker conversation or performance in real time or near-real time using multi-channel audio capture. Different speakers or participants in a conversation may each be assigned a separate microphone that is placed in proximity to the given speaker, where each audio channel includes audio captured by a different microphone. Filters may be applied to isolate each channel to include speech utterances of a different speaker, and these filtered channels of audio data may then be processed in parallel to generate speech-to-text results that are interleaved to form a generated transcript.

Disclosed herein are systems and methods for automatic correction of real ear measurements (REMs). A sound signal is produced through a receiver of a hearing device, and a sound pressure signal is sensed using a microphone placed inside the ear canal. The sound pressure signal is transformed to obtain a frequency response signal, a local minimum of the frequency response signal is detected above a programmable frequency level, and a spectral flatness of the frequency response signal is calculated in a selected frequency band surrounding the local minimum. If the spectral flatness is greater than a selected threshold value, acoustic correction is applied to the frequency response in the selected frequency band using an estimated transfer function to obtain a corrected sound pressure frequency response. The corrected sound pressure frequency response is used to modify, or make a recommendation to modify, a physical or operational characteristic of the hearing device.

A method of audio processing includes generating harmonics in a hybrid complex quadrature mirror filter domain. Generating the harmonics may include multiplication, using a feedback delay loop, and dynamic compression. The harmonics may be generated based on one or more hybrid sub-bands of the complex transform domain signal.

An audio device for improved talker discrimination is provided. To improve suppression of close talker interference, the audio device comprises at least a first and a second audio input to receive a first and second voice input signal; a first filter bank, configured to provide a plurality of first sub-band signals; a second filter bank, configured to provide a plurality of second sub-band signals; a correlator, configured to determine at least one signal correlation between at least a group of the first sub-band signals and at least a group of the second sub-band signals; and an attenuator, arranged to receive at least the group of the first sub-band signals and configured to conduct signal attenuation on the group of the first sub-band signals to provide gain-controlled sub-band signals, wherein the signal attenuation is based on the determined at least one signal correlation.

A hearing device, e.g. a hearing aid or a headset, configured to be worn by a comprises an input unit for providing at least one electric input signal in a time-frequency representation; and a signal processor comprising a target signal estimator for providing an estimate of the target signal; a noise estimator for providing an estimate of the noise; and a gain estimator for providing respective gain values in dependence of said target signal estimate and said noise estimate. The gain estimator comprises a trained neural network, wherein the outputs of the neural network comprise real or complex valued gains, or separate real valued gains and real valued phases. The signal processor is configured—at a given time instance t—to calculate changes Δx(i,t)=x(i,t)−{circumflex over (x)}(i,t−1), and Δh(j,t−1)=h(j,t−1)−ĥ(j,t−2) to an input vector x(t) and to the hidden state vector h(t−1), respectively, from one time instance, t−1, to the next, t, and where {circumflex over (x)}(i,t−1) and ĥ(j,t−2) are estimated values of x(i,t−1) and h(j,t−2), respectively, where indices i, j refers to the i.sup.th input neuron and the j.sup.th neuron of the hidden state, respectively, where 1≤i≤N.sub.ch,x and 1≤j≤N.sub.ch,oh, wherein N.sub.ch,x and N.sub.ch,oh is the number of processing channels of the input vector x and the hidden state vector h, respectively, and wherein the signal processor is further configured to provide that the number of updated channels among said N.sub.ch,x and said N.sub.ch,oh processing channels of the modified gated recurrent unit for said input vector x(t) and said hidden state vector h(t−1), respectively, at said given time instance t is limited to a number of peak values N.sub.p,x, and N.sub.p,oh, respectively, where N.sub.p,x is smaller than N.sub.ch,x, and N.sub.p,oh, is smaller than N.sub.ch,oh.