Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Systems and methods for adjusting audio levels in a plurality of audio signals are disclosed. One or more input signals containing one or more audio signals are depacketized. The audio signals are analyzed to determine if audio levels of the signals correspond to a desired audio level for at least some of the signals. The audio levels of signals that do not correspond are adjusted. Packets that do not require adjustment and the resulting adjusted packets are combined to form one or more output signals corresponding to the one or more input signals.

Abstract: A method includes: acquiring a delay amount of a second transfer function relative to a first transfer function, the first and second transfer function indicating sound transfer characteristics in a head of a user for a first and second sound source direction; and calculating a value of a third transfer function regarding a third sound source direction at the elapsed time, by interpolating a first value and a second value based on a first angular difference and a second angular difference, the first value being a value of the first transfer function at the elapsed time, the second value being a value of the second transfer function at a time after the elapsed time, the first angular difference being between the third sound source direction and the first sound source direction, the second angular difference being between the third sound source direction and the second sound source direction.

Abstract: A method may include adaptively generating an anti-noise signal from filtering a reference microphone signal with an adaptive filter in conformity with an error microphone signal and the reference microphone signal. The method may also include adjusting the response of the adaptive filter by combining injected noise with the reference microphone signal and receiving the injected noise by a copy of the adaptive filter so that the response of the copy is controlled by the adaptive filter adapting to cancel a combination of the ambient audio sounds and the injected noise and controlling the response of the adaptive filter with the coefficients adapted in the copy, whereby the injected noise is not present in the anti-noise signal and wherein each of a sample rate of the copy and a rate of adapting of the adaptive filter is significantly less than a sample rate of the adaptive filter.

Abstract: According to an embodiment, a transducer system includes a transducing element and a symmetry detection circuit. The transducing element includes a signal plate, a first sensing plate, and a second sensing plate. The symmetry detection circuit is coupled to a differential output of the transducer element and is configured to output an error signal based on asymmetry in the differential output.

Abstract: An audio amplifier circuit for providing an output signal to an audio transducer may include a power amplifier and a control circuit. The power amplifier may include an audio input for receiving an audio input signal, an audio output for generating the output signal based on the audio input signal, and a power supply input for receiving a power supply voltage, wherein the power supply voltage is variable among at least a first supply voltage and a second supply voltage greater than the first supply voltage. The control circuit may be configured to predict, based on one or more characteristics of a signal indicative of the output signal, an occurrence of a condition for changing the power supply voltage, and responsive to predicting the occurrence of the condition, change, at an approximate zero crossing of the signal indicative of the output signal, the power supply voltage.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: An acoustic echo canceller includes an adaptive filter and a double-talk detector. The adaptive filter includes a linear filter and a coefficient calculator. The linear filter has a transfer function that is controlled by a set of variable filter coefficients and that is configured to cancel an estimate of echo in a microphone signal to provide an output signal. The coefficient calculator is configured to update the set of variable filter coefficients based on a variable adaptation rate. The double-talk detector is configured to calculate changes in the energy of the variable filter coefficients (between updates of the coefficients). The acoustic echo canceller is configured to adjust the variable adaptation rate based on whether the energy of the variable filter coefficients is determined to be either oscillating or steadily changing (increasing or decreasing).

Abstract: An audio apparatus and a method of converting an audio signal are provided. The method includes: receiving a first audio signal including a plurality of channels; comparing audio signals of the plurality of channels to estimate a source position of the first audio signal; localizing a source of the first audio signal toward a three-dimensional (3D) position having an elevation component based on the estimated source position; converting the first audio signal into a second audio signal including the plurality of channels and at least one channel having, based on the localized source, a different elevation from the plurality of channels; and outputting the second audio signal.

Abstract: A portable speaker unit that includes a battery-powered, portable wireless speaker and can be seamlessly incorporated and operate as a speaker element of the vehicle speaker system. The portable speaker unit includes 2.1 channel audio capability with individually sealed speaker chambers for each of the 3 speakers within a housing. The portable speaker unit can be electrically and mechanically interfaced with the vehicle speaker system through a vehicle mount. The portable speaker unit can also operate as a portable wireless speaker when removed from the vehicle interface.

Abstract: A MEMS loudspeaker arrangement for generating sound waves in the audible wavelength spectrum includes a housing that defines a sound-conducting channel and a sound outlet arranged at the end of the sound-conducting channel. At least two MEMS loudspeakers are arranged in the interior of the housing so that they generate sound waves through the sound-conducting channel to the sound outlet. One of the MEMS loudspeakers is disposed downstream of the other in the direction of the sound outlet. A control unit is connected to control the MEMS loudspeakers so as to increase the maximum loudness of the MEMS loudspeaker arrangement. The first of the two MEMS loudspeakers is controlled to function as a sound generator for generating an initial wave, and the second MEMS loudspeaker is controlled to function as a sound amplifier for amplifying the initial wave.

Abstract: An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to contro

Abstract: A loudspeaker arrangement for a plurality of MEMS loudspeakers for generating sound waves in the audible wavelength spectrum includes a housing, which has a sound conduction cavity and at least one sound outlet opening, and at least two MEMS loudspeakers, arranged in the interior of the housing opposite and spaced apart from each other by the sound conduction cavity. Each MEMS loudspeaker has a cavity in the region of their opposite faces. The loudspeaker arrangement includes a shielding wall for acoustically decoupling the two MEMS loudspeakers from each other. The shielding wall is arranged in the interior of the housing between the two MEMS loudspeakers such that the sound conduction cavity is subdivided into a first and a second a cavity region respectively associated with one of the two MEMS loudspeakers.

Abstract: An earpiece includes an earpiece housing, a processor disposed within the earpiece, a speaker operatively connected to the processor, a microphone operatively connected the processor, and a global navigation satellite system (GNSS) receiver disposed within the earpiece. A system may includes a first earpiece having a connector with earpiece charging contacts, a charging case for the first earpiece, the charging case having contacts for connecting with the earpiece charging contacts, and a glob& navigation satellite system (GNSS) receiver disposed within the charging case.

Abstract: Some embodiments of the invention include a throwable microphone device. The throwable microphone device may comprise a housing. The throwable microphone device may include a microphone, a communication unit, a motion sensor, an orientation sensor, and a processor disposed within the housing. In some embodiments, the microphone may receive sound waves and generate a corresponding electrical audio signal. The communication unit may wirelessly transmit at least a portion of the electrical audio signals. The motion sensor may detect changes in acceleration of the throwable microphone device. The orientation sensor may detect changes in orientation of the throwable microphone device. The processor may be electrically coupled with the microphone, the communication unit, the motion sensor, and the orientation sensor. The processor may mute the throwable microphone device in response to data from the motion sensor and may also unmute the throwable microphone device in response to data from the orientation sensor.

Abstract: In an active noise reducing headphone, a signal processor applies filters and control gains of both the feed-forward and feedback active noise cancellation signal paths. The signal processor is configured to apply first feed-forward filters to the feed-forward signal path and apply first feedback filters to the feedback signal path during a first operating mode providing effective cancellation of ambient sound, and to apply second feed-forward filters to the feed-forward signal path during a second operating mode providing active hear-through of ambient sounds with ambient naturalness.