Abstract:
An audio signal processing device for generating a plurality of output signals for a plurality of loudspeakers from an input audio signal comprises a driving function determining unit adapted to determined driving functions of a plurality of loudspeakers for generating a virtual left binaural signal source and a virtual right binaural signal source based upon a position and a directivity of the virtual left binaural signal source, a position and a directivity of the virtual right binaural signal source and positions of the plurality of loudspeakers. Moreover, it comprises a filtering unit adapted to filter a left binaural signal and a right binaural signal using the driving functions of the plurality of loudspeakers resulting in the plurality of output signals. The left binaural signal and the right binaural signal constitute the input audio signal or are derived there from.
Abstract:
The invention relates to a portable electronic device, comprising: at least two directional microphones for stereo sound pickup, each one of the two directional microphones defining a direct sound direction and an opposite sound direction towards which the directional microphones are directed; and a housing comprising for each of the directional microphones a first hole and a second hole, the first hole being located at a different side of the portable electronic device than the second hole.
Abstract:
An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.
Abstract:
The invention relates to a method for determining an encoding parameter for an audio channel signal of a multi-channel audio signal, the method comprising: determining for the audio channel signal a set of functions from the audio channel signal and a reference audio signal; determining a first set of encoding parameters based on a smoothing of the set of functions with respect to a frame sequence of the multi-channel audio signal, the smoothing being based on a first smoothing coefficient; determining a second set of encoding parameters based on a smoothing of the set of functions with respect to the frame sequence of the multi-channel audio signal, the smoothing being based on a second smoothing coefficient; and determining the encoding parameter based on a quality criterion with respect to the first set of encoding parameters and/or the second set of encoding parameters.
Abstract:
According to the invention, a device (101, 101′) for postprocessing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver (103; 103′) for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal and a classification indication indicating a transient type of the at least one channel signal, wherein the classification indication is associated to the at least one channel signal, and a postprocessor (105; 105′) for postprocessing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication.
Abstract:
Methods and devices for a low complex inter-channel difference estimation are provided. A method for the estimation of inter-channel differences (ICDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of ICD values for the ICDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each ICD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of ICD values, a weighted ICD value by multiplying each of the plurality of ICD values with a corresponding frequency-dependent weighting factor, and calculating an ICD range value for the predetermined frequency range by adding the plurality of weighted ICD values.
Abstract:
The invention relates to a portable electronic device, comprising: at least two directional microphones for stereo sound pickup, each one of the two directional microphones defining a direct sound direction and an opposite sound direction towards which the directional microphones are directed; and a housing comprising for each of the directional microphones a first hole and a second hole, the first hole being located at a different side of the portable electronic device than the second hole.
Abstract:
An apparatus and a method for generating an acoustic signal with an enhanced spatial effect, said apparatus comprising a signal filter bank adapted to filter a difference audio signal with a filter characteristic to limit a bandwidth of said difference audio signal, wherein said bandwidth limited difference audio signal is applied to at least one pair of loudspeakers for dipole sound emission.
Abstract:
The invention relates to a parametric audio encoder, comprising a parameter generator, the parameter generator being configured to determine a first set of encoding parameters and reference audio signal values, wherein the reference audio signal is another audio channel signal or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals, to determine a first encoding parameter average based on the first set of encoding parameters of the audio channel signal, to determine a second encoding parameter average based on the first encoding parameter average of the audio channel signal and at least one other first encoding parameter average of the audio channel signal, and to determine the encoding parameter based on the first encoding parameter average of the audio channel signal and the second encoding parameter average of the audio channel signal.
Abstract:
The invention relates to an audio signal processing apparatus (100) for processing an input earpiece audio signal (x) upon the basis of a microphone audio signal (y), the audio signal processing apparatus (100) comprising a voice activity detector (101) being configured to determine a voice activity indicator signal (xvad) upon the basis of the input earpiece audio signal (x), a noise magnitude determiner (103) being configured to determine a microphone noise magnitude indicator signal (wy) upon the basis of the microphone audio signal (y), a gain factor determiner (105) being configured to determine a gain factor signal (ΔG) upon the basis of the voice activity indicator signal (xvad) and the microphone noise magnitude indicator signal (wy), and a weighter (107) being configured to weight the input earpiece audio signal (x) by the gain factor signal (ΔG) to obtain an output earpiece audio signal.