Abstract: A method for determining an output stereo signal comprising determining a first differential signal and determining a second differential signal; determining a first power spectrum based on the first differential signal and determining a second power spectrum based on the second differential signal; determining a first weighting function and a second weighting function as a function of the first power spectrum and the second power spectrum; and filtering a first signal, which represents a first combination of the first input audio channel signal and the second input audio channel signal, and filtering a second signal, which represents a second combination of the first input audio channel signal and the second input audio channel signal.

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: An audio compression system for compressing an input audio signal, and the audio compression system comprises a digital filter for filtering the input audio signal, where the digital filter comprises a frequency transfer function having a magnitude over frequency, where the magnitude is formed by an equal loudness curve of a human ear to obtain a filtered audio signal, and a compressor which is configured to compress the input audio signal upon the basis of the filtered audio signal to obtain a compressed audio signal.

Abstract: A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.

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: 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: A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.

Abstract: A method and an apparatus for estimating an interchannel delay of a sound signal are, related to the communication field and capable of realizing a stable sound field in a crosstalk. The method includes calculating an error between an actual interchannel phase difference and a predicted interchannel phase difference of a sound signal. The predicted interchannel phase difference is predicted according to a predetermined interchannel delay of the sound signal. The method also includes determining whether the sound signal is a sound signal in a crosstalk according to the error. The method further includes, if the sound signal is a sound signal in the crosstalk, setting an interchannel delay corresponding to the sound signal to a fixed value. The apparatus is configured to perform the method.

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 system and a method for evaluating an acoustic transfer function, wherein the acoustic transfer function is a transfer function from one acoustic source to a reproduction area sampled by a limited number of microphone modules.

Abstract: A virtual stereo synthesis method includes acquiring at least one sound input signal on a first side and at least one sound input signal on a second side, separately performing ratio processing on a preset head related transfer function (HRTF) left-ear component and a preset HRTF right-ear component of each sound input signal on the second side, to obtain a filtering function of each sound input signal on the second side, separately performing convolution filtering on each sound input signal on the second side and the filtering function of the sound input signal on the second side, to obtain the filtered signal on the second side, and synthesizing all of the sound input signals on the first side and all of the filtered signals on the second side into a virtual stereo signal where the method may alleviate a coloration effect, and reduce calculation complexity.

Abstract: A method for determining an output stereo signal comprising determining a first differential signal and determining a second differential signal; determining a first power spectrum based on the first differential signal and determining a second power spectrum based on the second differential signal; determining a first weighting function and a second weighting function as a function of the first power spectrum and the second power spectrum; and filtering a first signal, which represents a first combination of the first input audio channel signal and the second input audio channel signal, and filtering a second signal, which represents a second combination of the first input audio channel signal and the second input audio channel signal.

Abstract: The invention relates to a method for processing a multi-channel audio signal which carries a plurality of audio channel signals. The method comprises determining a time-scaling position using the plurality of audio channel signals and time-scaling each audio channel signal of the plurality of audio channel signals according to the time-scaling position to obtain a plurality of time scaled audio channel signals.

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: 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: A method for parametric spatial audio coding of a multi-channel audio signal comprising a plurality of audio channel signals is provided, the method comprising: calculating at least two different spatial coding parameters for an audio channel signal of the plurality of audio channel signals, selecting at least one spatial coding parameter of the at least two different spatial coding parameters associated with the audio channel signal on the basis of the values of the calculated spatial coding parameters; including a quantized representation of the selected spatial coding parameter into a parameter section of an audio bitstream; and setting a parameter type flag in the parameter section of the audio bitstream indicating the type of the selected spatial coding parameter being included into the audio bitstream.

Abstract: Embodiments of the present invention provide a stereo signal down-mixing method, encoding/decoding apparatus and system. The down-mixing method includes: converting a first channel time-domain signal and a second channel time-domain signal into a first channel frequency-domain signal and a second channel frequency-domain signal; obtaining a frequency-domain channel signal level difference and a frequency-domain channel signal phase difference between the two channel frequency-domain signals; for each frequency bin in each frequency band, using a function based on the frequency-domain channel signal level difference and frequency-domain channel signal phase difference to obtain a down-mixed signal phase that is located between phases of the two channel frequency-domain signals, and obtaining a down-mixed signal amplitude through calculation; and obtaining a frequency-domain down-mixed signal according to the phase and amplitude.

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.