摘要:
A device for generating a binaural signal based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having a virtual sound source position associated to each channel, is described. It comprises a correlation reducer for differently processing, and thereby reducing a correlation between, at least one of a left and a right channel of the plurality of channels, a front and a rear channel of the plurality of channels, and a center and a non-center channel of the plurality of channels, in order to obtain an inter-similarity reduced set of channels; a plurality of directional filters, a first mixer for mixing outputs of the directional filters modeling the acoustic transmission to the first ear canal of the listener, and a second mixer for mixing outputs of the directional filters modeling the acoustic transmission to the second ear canal of the listener. According to another aspect, a center level reduction for forming the downmix for a room processor is performed. According to even another aspect, an inter-similarity decreasing set of head-related transfer functions is formed.
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:
A device for generating a binaural signal based on a multi-channel signal representing a plurality of channels and intended for reproduction by a speaker configuration having a virtual sound source position associated to each channel, is described. It comprises a correlation reducer for differently processing, and thereby reducing a correlation between, at least one of a left and a right channel of the plurality of channels, a front and a rear channel of the plurality of channels, and a center and a non-center channel of the plurality of channels, in order to obtain an inter-similarity reduced set of channels; a plurality of directional filters, a first mixer for mixing outputs of the directional filters modeling the acoustic transmission to the first ear canal of the listener, and a second mixer for mixing outputs of the directional filters modeling the acoustic transmission to the second ear canal of the listener. According to another aspect, a center level reduction for forming the downmix for a room processor is performed. According to even another aspect, an inter-similarity decreasing set of head-related transfer functions is formed.
摘要:
An apparatus for determining an estimated pitch lag is provided. The apparatus comprises an input interface (110) for receiving a plurality of original pitch lag values, and a pitch lag estimator (120) for estimating the estimated pitch lag. The pitch lag estimator (120) is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to said original pitch lag value.
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 audio signal decoder (100) for providing a decoded audio signal representation on the basis of an encoded audio signal representation comprises a decoder preprocessing stage (110) for obtaining a plurality of frequency band signals from the encoded audio signal representation, a clipping estimator (120), a level shifter (130), a frequency-to-time-domain converter (140), and a level shift compensator (150). The clipping estimator (120) analyzes the encoded audio signal representation and/or side information relative to a gain of the frequency band signals in order to determine a current level shift factor. The level shifter (130) shifts levels of the frequency band signals according to the level shift factor. The frequency-to-time-domain converter (140) converts the level shifted frequency band signals into a time-domain representation. The level shift compensator (150) acts on the time-domain representation for at least partly compensating a corresponding level shift and for obtaining a substantially compensated time-domain representation.
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:
An audio processor for processing an audio signal (100), comprises: an audio signal modifier (2) for modifying the audio signal (100) in response to a user input (200); a loudness controller (6) for determining a loudness compensation gain (C) based on a reference loudness (L ref ) or a reference gain (g i ) and a modified loudness (L mod ) or a modified gain (h i ), where the modified loudness (L mod ) or the modified gain (h i ) depends on the user input; and a loudness manipulator (5) for manipulating a loudness of a signal (101) using the loudness compensation gain (C).
摘要:
An apparatus for determining an estimated pitch lag is provided. The apparatus comprises an input interface (110) for receiving a plurality of original pitch lag values, and a pitch lag estimator (120) for estimating the estimated pitch lag. The pitch lag estimator (120) is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to said original pitch lag value.