摘要:
A method for processing an audio signal (504) in accordance with a room impulse response is described. The audio signal (504) is processed (502) with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing (514) of the late reverberation comprises generating a scaled reverberated signal, the scaling (526) being dependent on the audio signal (504). The processed early part (506) of the audio signal (504) and the scaled reverberated signal are combined.
摘要:
Audio data processor, comprising: a receiver interface for receiving encoded audio data and metadata related to the encoded audio data; a metadata parser for parsing the metadata to determine an audio data manipulation possibility; an interaction interface for receiving an interaction input and for generating, from the interaction input, interaction control data related to the audio data manipulation possibility; and a data stream generator for obtaining the interaction control data and the encoded audio data and the metadata and for generating an output data stream, the output data stream comprising the encoded audio data, at least a portion of the metadata, and the interaction control data.
摘要:
Audio processor (10) comprising an input interface, a detector interface (32), a mixer (22) and an output interface. The input interface receiving at least two input audio channels (12 1 , 12 2 ), each input audio channel (12 1 , 12) being associated with a predetermined reproduction position of at least two loudspeakers (26 1 , 26 2 ) on at least one loudspeaker axis (16). The detector interface (32) receiving a position signal (18) indicating an information on a position of the at least two loudspeakers (26 1 , 26 2 ) with respect to an ear axis (20) of a listener (28), wherein the ear axis (20) and the at least one loudspeaker axis (16) have an angle (36) to each other, being greater than 0° and lower than 180°. The mixer (22) mixing the at least two input audio channels (12 1 , 12 2 ) to obtain the at least two output channels (14 1 , 14 2 ) depending on the position signal (18), such that a portion of the second input audio channel (12 2 ) in the first output channel (14 1 ) for a first angle (36) between the ear axis (20) and the loudspeaker axis (16) is greater than a portion of the second input audio channel (12 2 ) in the first output channel (14 1 ) for a second angle (36) between the ear axis (20) and the loudspeaker axis (16), wherein the first angle (36) is greater than the second angle (36). Further a portion of the first input audio channel (12 1 ) in the second output channel (14 2 ) for the first angle (36) is greater than the portion of the first input audio channel (12 1 ) in the second output channel (14 2 ) for the second angle (36), wherein the first angle (36) is greater than the second angle (36). The output interface outputting the at least two output channels (14 1 , 14 2 ) to the at least two loudspeakers.
摘要:
An equalization filter coefficient determinator for determining a current set of equalization filter target coefficients for use by an equalizer is configured to continuously or quasi-continuously fade between a plurality of different equalizer settings in dependence on one or more setting parameters, to obtain the current set of equalization filter target coefficients describing a current equalizer setting. A number of setting parameters is smaller than a number of equalization filter target coefficients of current set of equalization filter target coefficients. An equalization filter coefficient determinator is configured to linearly combine a plurality of equalization filter target coefficient set components in dependence on one or more setting parameters, to obtain the current set of equalization filter target coefficients. An equalization filter coefficient determinator is configured to obtain the current set of equalization filter target coefficients in dependence on a two-dimensional position information or a three-dimensional position information obtained using a two-dimensional or three-dimensional user input device. An apparatus comprises a user interface, an equalization filter coefficient determinator and an equalizer. An equalization filter coefficient processor may provide sets of basis equalization filter target coefficients. A system may use an equalization filter coefficient processor and an equalization filter coefficient determinator.
摘要:
An apparatus for extracting a direct and/or ambience signal from a downmix signal and spatial parametric information, the downmix signal and the spatial parametric information representing a multi-channel audio signal having more channels than the downmix signal, wherein the spatial parametric information comprises inter-channel relations of the multi-channel audio signal, is described. The apparatus comprises a direct/ambience estimator and a direct/ambience extractor. The direct/ambience estimator is configured for estimating a level information of a direct portion and/or an ambient portion of the multi-channel audio signal based on the spatial parametric information. The direct/ambience extractor is configured for extracting a direct signal portion and/or an ambient signal portion from the downmix signal based on the estimated level information of the direct portion or the ambient portion.
摘要:
A downmixer for providing a downmix signal on the basis of a plurality of input signals is configured to determine a magnitude value of a spectral domain value of the downmix signal on the basis of a loudness information of the input signals. The downmixer is configured to determine a phase value of the spectral domain value of the downmix signal and the downmixer is configured to apply the phase value in order to obtain a complex valued number representation of the spectral domain value of the downmix signal on the basis of the magnitude value of the spectral domain value of the downmix signal. An audio encoder uses such a downmixer. A method for downmixing and a computer program are also described.
摘要:
A downmixer for providing a downmix signal on the basis of a plurality of input signals is configured to determine a magnitude value of a spectral domain value of the downmix signal on the basis of a loudness information of the input signals. The downmixer is configured to determine a phase value of the spectral domain value of the downmix signal and the downmixer is configured to apply the phase value in order to obtain a complex valued number representation of the spectral domain value of the downmix signal on the basis of the magnitude value of the spectral domain value of the downmix signal. An audio encoder uses such a downmixer. A method for downmixing and a computer program are also described.
摘要:
An apparatus for generating loudspeaker signals is provided. The apparatus comprises an object metadata processor (110) and an object renderer (120). The object renderer (120) is configured to receive an audio object. The object metadata processor (110) is configured to receive metadata, comprising an indication on whether the audio object is screen-related, and further comprising a first position of the audio object. The object metadata processor (110) is configured to calculate a second position of the audio object depending on the first position of the audio object and depending on a size of a screen, if the audio object is indicated in the metadata as being screen-related. The object renderer (120) is configured to generate the loudspeaker signals depending on the audio object and depending on position information. The object metadata processor (110) is configured to feed the first position of the audio object as the position information into the object renderer (120), if the audio object is indicated in the metadata as being not screen-related. The object metadata processor (110) is configured to feed the second position of the audio object as the position information into the object renderer (120), if the audio object is indicated in the metadata as being screen-related.