Abstract:
A local wave field synthesis apparatus, which includes a determination module for determining desired sound pressures and desired particle velocity vectors at a plurality of control points, a computation module for computing sound pressures and particle velocity vectors at the plurality of control points based on a set of filter parameters, an optimization module for computing an optimum set of filter parameters by jointly optimizing computed sound pressures towards the desired sound pressures and computed particle velocity vectors towards the desired particle velocity vectors, and a generator module for generating the drive signals based on the optimum set of filter parameters, wherein the plurality of control points are located on one or more contours around the one or more audio zones.
Abstract:
A microphone arrangement and a method using the microphone arrangement for recording surround sound in a mobile device, where the microphone arrangement comprises a first and a second microphone and arranged at a first distance to each other and configured to obtain a stereo signal, and comprises a third microphone configured to obtain a steering signal together with at least one of the first and second microphone or with a fourth microphone. The microphone arrangement also comprises a processor configured to separate the stereo signal into a front stereo signal and a back stereo signal based on the steering signal.
Abstract:
The disclosure is based on the finding that acoustic near-field transfer functions indicating acoustic near-field propagation channels between loudspeakers and ears of a listener can be employed to pre-process audio signals. Therefore, acoustic near-field distortions of the audio signals can be mitigated. The pre-processed audio signals can be presented to the listener using a wearable frame, wherein the wearable frame comprises the loudspeakers for audio presentation. The disclosure can allow for a high quality rendering of audio signals as well as a high listening comfort for the listener. The disclosure can provide the following advantages. By means of a loudspeaker selection as a function of a spatial audio source direction, cues related to the listener's ears can be generated, making the approach more robust with regard to front/back confusion. The approach can further be extended to an arbitrary number of loudspeaker pairs.
Abstract:
A method for processing an audio signal includes: decomposing an audio signal comprising spatial information into a set of audio signal components; and processing a first subset of the set of audio signal components according to a first processing scheme and processing a second subset of the set of audio signal components according to a second processing scheme different from the first processing scheme, wherein the first subset comprises audio signal components corresponding to at least one frontal signal source and the second subset comprises audio signal components corresponding to at least one ambient signal source; and wherein the second processing scheme is based on crosstalk cancellation.
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:
An apparatus and a method for enhancing a spatial perception of an audio signal are provided creating increased interaural-level differences. To obtain this effect, two dipoles are used: one for producing a left audio signal and one for producing a right audio signal.
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 a frequency transform of the audio channel signal; determining a frequency transform of a reference audio signal; determining inter channel differences for at least each frequency sub-band of a subset of frequency sub-bands, each inter channel difference indicating a phase difference or time difference between a band-limited signal portion of the audio channel signal and a band-limited signal portion of the reference audio signal in the respective frequency sub-band the inter-channel difference is associated to; determining a first average based on positive values of the inter-channel differences and determining a second average based on negative values of the inter-channel differences; and determining the encoding parameter based on the first average and on the second average.
Abstract:
A method for processing an audio signal includes: decomposing an audio signal comprising spatial information into a set of audio signal components; and processing a first subset of the set of audio signal components according to a first processing scheme and processing a second subset of the set of audio signal components according to a second processing scheme different from the first processing scheme, wherein the first subset comprises audio signal components corresponding to at least one frontal signal source and the second subset comprises audio signal components corresponding to at least one ambient signal source; and wherein the second processing scheme is based on crosstalk cancellation.
Abstract:
A digital compressor for compressing an input audio signal is presented. The digital compressor comprises a compression gain control for providing a compression gain parameter, and a compression parameter determiner for determining a compression ratio from the compression gain parameter. The compression parameter determiner may be configured to weight the compression gain parameter by a predetermined weight factor to obtain the compression ratio. The digital compressor further comprises an auxiliary signal generator for manipulating the input audio signal in dependence of the compression ratio to obtain a first auxiliary signal, and a combiner unit for combining the first auxiliary signal with the compression gain parameter to obtain a second auxiliary signal, and for combining the input audio signal with the second auxiliary signal to obtain the compressed audio signal.
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.