摘要:
An apparatus for extracting an ambient signal from an input audio signal comprises a gain-value determinator configured to determine a sequence of time-varying ambient signal gain values for a given frequency band of the time-frequency distribution of the input audio signal in dependence on the input audio signal. The apparatus comprises a weighter configured to weight one of the sub-band signals representing the given frequency band of the time-frequency-domain representation with the time-varying gain values, to obtain a weighted sub-band signal. The gain-value determinator is configured to obtain one or more quantitative feature-values describing one or more features of the input audio signal and to provide the gain-value as a function of the one or more quantitative feature values such that the gain values are quantitatively dependent on the quantitative values. The gain value determinator is configured to determine the gain values such that ambience components are emphasized over non-ambience components in the weighted sub-band signal.
摘要:
An apparatus for extracting an ambient signal from an input audio signal comprises a gain-value determinator configured to determine a sequence of time-varying ambient signal gain values for a given frequency band of the time-frequency distribution of the input audio signal in dependence on the input audio signal. The apparatus comprises a weighter configured to weight one of the sub-band signals representing the given frequency band of the time-frequency-domain representation with the time-varying gain values, to obtain a weighted sub-band signal. The gain-value determinator is configured to obtain one or more quantitative feature-values describing one or more features of the input audio signal and to provide the gain-value as a function of the one or more quantitative feature values such that the gain values are quantitatively dependent on the quantitative values. The gain value determinator is configured to determine the gain values such that ambience components are emphasized over non-ambience components in the weighted sub-band signal.
摘要:
An apparatus for processing an audio signal to obtain control information for a speech enhancement filter has a feature extractor for extracting at least one feature per frequency band of a plurality of frequency bands of a short-time spectral representation of a plurality of short-time spectral representations, where the at least one feature represents a spectral shape of the short-time spectral representation in the frequency band. The apparatus additionally has a feature combiner for combining the at least one feature for each frequency band using combination parameters to obtain the control information for the speech enhancement filter for a time portion of the audio signal. The feature combiner can use a neural network regression method, which is based on combination parameters determined in a training phase for the neural network.
摘要:
An apparatus for processing an audio signal to obtain control information for a speech enhancement filter has a feature extractor for extracting at least one feature per frequency band of a plurality of frequency bands of a short-time spectral representation of a plurality of short-time spectral representations, where the at least one feature represents a spectral shape of the short-time spectral representation in the frequency band. The apparatus additionally has a feature combiner for combining the at least one feature for each frequency band using combination parameters to obtain the control information for the speech enhancement filter for a time portion of the audio signal. The feature combiner can use a neural network regression method, which is based on combination parameters determined in a training phase for the neural network.
摘要:
In order to generate a multi-channel signal having a number of output channels greater than a number of input channels, a mixer is used for upmixing the input signal to form at least a direct channel signal and at least an ambience channel signal. A speech detector is provided for detecting a section of the input signal, the direct channel signal or the ambience channel signal in which speech portions occur. Based on this detection, a signal modifier modifies the input signal or the ambience channel signal in order to attenuate speech portions in the ambience channel signal, whereas such speech portions in the direct channel signal are attenuated to a lesser extent or not at all. A loudspeaker signal outputter then maps the direct channel signals and the ambience channel signals to loudspeaker signals which are associated to a defined reproduction scheme, such as, for example, a 5.1 scheme.
摘要:
In order to generate a multi-channel signal having a number of output channels greater than a number of input channels, a mixer is used for upmixing the input signal to form at least a direct channel signal and at least an ambience channel signal. A speech detector is provided for detecting a section of the input signal, the direct channel signal or the ambience channel signal in which speech portions occur. Based on this detection, a signal modifier modifies the input signal or the ambience channel signal in order to attenuate speech portions in the ambience channel signal, whereas such speech portions in the direct channel signal are attenuated to a lesser extent or not at all. A loudspeaker signal outputter then maps the direct channel signals and the ambience channel signals to loudspeaker signals which are associated to a defined reproduction scheme, such as, for example, a 5.1 scheme.
摘要:
An apparatus for generating an ambient signal from an audio signal includes a compressor for lossy compression of a representation of the audio signal so as to obtain a compressed representation of the audio signal describing a compressed audio signal. The apparatus for generating the ambient signal further includes a calculator for calculating a difference between the compressed representation of the audio signal and the representation of the audio signal so as to obtain a discrimination representation. The apparatus further includes a provider for providing the ambient signal using the discrimination representation. An apparatus for deriving a multi-channel audio signal from an audio signal includes an apparatus for generating an ambient signal from an audio signal, an apparatus for providing the audio signal as a front-loudspeaker signal and an apparatus for providing the ambient signal as a back-loudspeaker signal.
摘要:
An apparatus for generating an ambient signal from an audio signal includes a compressor for lossy compression of a representation of the audio signal so as to obtain a compressed representation of the audio signal describing a compressed audio signal. The apparatus for generating the ambient signal further includes a calculator for calculating a difference between the compressed representation of the audio signal and the representation of the audio signal so as to obtain a discrimination representation. The apparatus further includes a provider for providing the ambient signal using the discrimination representation. An apparatus for deriving a multi-channel audio signal from an audio signal includes an apparatus for generating an ambient signal from an audio signal, an apparatus for providing the audio signal as a front-loudspeaker signal and an apparatus for providing the ambient signal as a back-loudspeaker signal.
摘要:
In order to analyze an information signal, a significant short-time spectrum is extracted from the information signal, the means for extracting being configured to extract such short-time spectra which come closer to a specific characteristic than other short-time spectra of the information signal. The short-time spectra extracted are then decomposed into component signals using ICA analysis, a component signal spectrum representing a profile spectrum of a tone source which generates a tone corresponding to the characteristic sought for. From a sequence of short-time spectra of the information signal and from the profile spectra determined, an amplitude envelope is eventually calculated for each profile spectrum, the amplitude envelope indicating how a profile spectrum of a tone source all in all changes over time. The profile spectra and all the amplitude envelopes associated therewith provide a description of the information signal which may be evaluated further, for example for transcription purposes in the case of a music signal.
摘要:
A significant short-time spectrum is extracted from an information signal, the means for extracting being configured to extract such short-time spectra which come closer to a specific characteristic than others. The short-time spectra extracted are then decomposed into component signals using ICA analysis, a component signal spectrum representing a profile spectrum of a tone source which generates a tone corresponding to the characteristic sought. From a sequence of short-time spectra of the information signal and from the profile spectra determined, an amplitude envelope is calculated for each profile spectrum to indicate how a tone source profile spectrum changes over time. The profile spectra and all the amplitude envelopes associated therewith provide a description of the information signal which may be evaluated further, for example for transcription purposes in the case of a music signal.