Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: A device for estimating Direction of Arrival (DOA) of sound from Q?1 sound sources is provided. The device is configured to obtain a phase difference matrix, which includes measured phase difference values, each of the measured phase difference values being a measured value of a phase difference between two microphone units for a frequency bin in a range of frequencies of the sound. The device is further configured to generate a replicated phase difference matrix by replicating the measured phase difference values to other potential sinusoidal periods, calculate a DOA value for each phase difference value in the replicated phase difference matrix, and determine, as Q DOA results, the Q most prominent peak values in a histogram generated based on the calculated DOA values.

Abstract: The proposed method for localizing a target sound source from a plurality of sound sources, wherein a multi-channel recording signal of the plurality of sound sources comprises a plurality of microphone channel signals, comprises converting each microphone channel signal into a respective channel spectrogram in a time-frequency domain, blindly separating the channel spectrograms to obtain a plurality of separated source signals, identifying, among the plurality of separated source signals, the separated source signal that best matches a target source model, estimating, based on the identified separated source signal, a binary mask reflecting where the target sound source is active in the channel spectrograms in terms of time and frequency, applying the binary mask on the channel spectrograms to obtain masked channel spectrograms, and localizing the target sound source from the plurality of sound sources based on the masked channel spectrograms.

Abstract: An apparatus configured to obtain a plurality of phase differences is provided. Each phase difference represents a difference in phase between a spectral component in a first sound signal and a corresponding spectral component in a second sound signal, to estimate an aliasing frequency. Phase differences in frequency bands above the aliasing frequency are expected to be wrapped, and to unwrap the plurality of phase differences in dependence on the estimated aliasing frequency. A phase difference obtained from spectral components that are comprised in one frequency band is unwrapped independently of any phase difference obtained from spectral components that are comprised in another frequency band.

Abstract: A device having an input that is configured to receive a sound signal captured from an acoustic environment where the device is located, is provided. The device also includes a signal processor that is configured to obtain a reverberation signature from the sound signal and classify the acoustic environment as being of a particular type in dependence on that reverberation signature. It also includes a controller configured to control an operation of the device in dependence on the classification of the acoustic environment. By using the reverberation signature to classify the environment, the device it is not dependent on a particular type of signal having to be present in the environment at the time when the sound signal is captured. The reverberation properties of the environment will be embodied by any sound signal captured from that environment.

Abstract: A device for estimating Direction of Arrival (DOA) of sound from Q?1 sound sources is provided. The device is configured to obtain a phase difference matrix, which includes measured phase difference values, each of the measured phase difference values being a measured value of a phase difference between two microphone units for a frequency bin in a range of frequencies of the sound. The device is further configured to generate a replicated phase difference matrix by replicating the measured phase difference values to other potential sinusoidal periods, calculate a DOA value for each phase difference value in the replicated phase difference matrix, and determine, as Q DOA results, the Q most prominent peak values in a histogram generated based on the calculated DOA values.

Abstract: Example audio signal processing apparatuses and methods for processing an input audio signal into an output audio signal are disclosed. An example audio signal processing apparatus comprises a decomposer configured to decompose the input audio signal into a direct audio signal and a diffuse audio signal, a modifier configured to modify the direct audio signal in order to obtain a modified direct audio signal, wherein the modifier comprises a bandwidth extender configured to extend an upper cutoff frequency of a frequency range of the direct audio signal, and a combiner configured to combine the modified direct audio signal with the diffuse audio signal to obtain the output audio signal.

Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: An apparatus is configured to obtain a plurality of phase differences, wherein each phase difference represents a difference in phase between a spectral component in a first sound signal and a corresponding spectral component in a second sound signal, to estimate an aliasing frequency, wherein phase differences in frequency bands above the aliasing frequency are expected to be wrapped, and to unwrap the plurality of phase differences in dependence on the estimated aliasing frequency, wherein a phase difference obtained from spectral components that are comprised in one frequency band is unwrapped independently of any phase difference obtained from spectral components that are comprised in another frequency band.

Abstract: A device has an input that is configured to receive a sound signal captured from an acoustic environment in which the device is located. The device also includes a signal processor that is configured to obtain a reverberation signature from the sound signal and classify the acoustic environment as being of a particular type in dependence on that reverberation signature. It also includes a controller configured to control an operation of the device in dependence on the classification of the acoustic environment. By using the reverberation signature to classify the environment, the device it is not dependent on a particular type of signal having to be present in the environment at the time when the sound signal is captured. The reverberation properties of the environment will be embodied by any sound signal captured from that environment.

Abstract: An input audio signal (101) is separated into input audio signal components (X(k,b)). A set of two or more band branches (105e) provides output audio signal components (Y(k,b)). The set of band branches (105e) comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal (101) and the summed audio signal (y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal (103) is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: A signal processing apparatus for enhancing a voice component within a multi-channel audio signal comprising a left channel audio signal, a center channel audio signal, and a right channel audio signal, the signal processing apparatus comprising a filter and a combiner; wherein the filter is configured to determine an overall magnitude of the multi-channel audio signal over frequency based on the multi-channel audio signal, to obtain a gain function based on a ratio between a magnitude of the center channel audio signal and the overall magnitude of the multi-channel audio signal, and to weight the left channel audio signal, the center channel audio signal, and the right channel audio signal by the gain function; and wherein the combiner is configured to combine individually the left channel audio signal, the center channel audio signal, and the right channel audio signal with the weighted right channel audio signal.

Abstract: The proposed method for localizing a target sound source from a plurality of sound sources, wherein a multi-channel recording signal of the plurality of sound sources comprises a plurality of microphone channel signals, comprises converting each microphone channel signal into a respective channel spectrogram in a time-frequency domain, blindly separating the channel spectrograms to obtain a plurality of separated source signals, identifying, among the plurality of separated source signals, the separated source signal that best matches a target source model, estimating, based on the identified separated source signal, a binary mask reflecting where the target sound source is active in the channel spectrograms in terms of time and frequency, applying the binary mask on the channel spectrograms to obtain masked channel spectrograms, and localizing the target sound source from the plurality of sound sources based on the masked channel spectrograms.

Abstract: The disclosure relates to an audio signal processing apparatus for modifying a stereo image of a stereo signal. The apparatus includes a panning index modifier configured to apply a mapping function to at least all panning indexes of stereo signal time-frequency segments that are within a frequency bandwidth, a first panning gain determiner configured to determine modified panning gains for time-frequency signal segments of the first and second audio signal based on the modified panning indexes, and a re-panner configured to re-pan the stereo signal according to ratios between the modified panning gains and panning gains of the first and second audio signal that correspond to the modified panning gains in time and frequency.

Abstract: Example audio signal processing apparatuses and methods for processing an input audio signal into an output audio signal are disclosed. An example audio signal processing apparatus comprises a decomposer configured to decompose the input audio signal into a direct audio signal and a diffuse audio signal, a modifier configured to modify the direct audio signal in order to obtain a modified direct audio signal, wherein the modifier comprises a bandwidth extender configured to extend an upper cutoff frequency of a frequency range of the direct audio signal, and a combiner configured to combine the modified direct audio signal with the diffuse audio signal to obtain the output audio signal.

Abstract: The disclosure relates to an audio signal processing apparatus for modifying a stereo image of a stereo signal. The apparatus includes a panning index modifier configured to apply a mapping function to at least all panning indexes of stereo signal time-frequency segments that are within a frequency bandwidth, a first panning gain determiner configured to determine modified panning gains for time-frequency signal segments of the first and second audio signal based on the modified panning indexes, and a re-panner configured to re-pan the stereo signal according to ratios between the modified panning gains and panning gains of the first and second audio signal that correspond to the modified panning gains in time and frequency.

Abstract: A signal processing apparatus for enhancing a voice component within a multi-channel audio signal comprising a left channel audio signal, a center channel audio signal, and a right channel audio signal, the signal processing apparatus comprising a filter and a combiner; wherein the filter is configured to determine an overall magnitude of the multi-channel audio signal over frequency based on the multi-channel audio signal, to obtain a gain function based on a ratio between a magnitude of the center channel audio signal and the overall magnitude of the multi-channel audio signal, and to weight the left channel audio signal, the center channel audio signal, and the right channel audio signal by the gain function; and wherein the combiner is configured to combine individually the left channel audio signal, the center channel audio signal, and the right channel audio signal with the weighted right channel audio signal.

Abstract: A notebook computer has a basic body comprising a bottom shell and a top shell. A display unit is arranged in a pivotable lid housing and a keyboard is detachably mounted on the top shell. A cutout is provided in the top shell and an outwardly directed electrical contact configuration is secured on an assembly that is attached to the basic body such that the contact configuration is secured in the cutout. The contact configuration is for electrically contacting the keyboard, which has a corresponding mating contact configuration.