摘要:
An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform providing a visual representation of at least one audio parameter associated with at least one audio signal, detecting via an interface an interaction with the visual representation of the audio parameter, and processing the at least one audio signal associated with the audio parameter dependent on the interaction.
摘要:
An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform providing a visual representation of at least one audio parameter associated with at least one audio signal, detecting via an interface an interaction with the visual representation of the audio parameter, and processing the at least one audio signal associated with the audio parameter dependent on the interaction.
摘要:
The invention relates to a method and an apparatus for processing an audio signal, wherein the method comprises the steps of: filtering an audio signal into at least two frequency band signals, generating for each frequency band signal a plurality of sub-band signals, wherein for at least one frequency band signal the plurality of sub-band signals are generated using a time to frequency domain transform and for the at least one other frequency band the plurality of sub-band signals for the other frequency band are generated using a sub-band filterbank, and the apparatus comprises at least one processor and at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processor, cause the apparatus to perform the method.
摘要:
An apparatus for extending the bandwidth of an audio signal, the apparatus being configured to: generate an excitation signal from an audio signal, wherein in the audio signal comprises a plurality of frequency components; extract a feature vector from the audio signal, wherein the feature vector comprises at least one frequency domain component feature and at least one time domain component feature; determine at least one spectral shape parameter from the feature vector, wherein the at least one spectral shape parameter corresponds to a sub band signal comprising frequency components which belong to a further plurality of frequency components; and generate the sub band signal by filtering the excitation signal through a filter bank and weighting the filtered excitation signal with the at least one spectral shape parameter.
摘要:
Both a cascade and a multichannel joint Bayesian estimator are provided for suppressing acoustic echo. An expansion basis (Power/Fourier series) is selected to convert a sample-based input signal xt into a DFT-domain multichannel signal [Xτ,1, . . . Xτ,p]. The posterior of unknown states (e.g., mean Ŵτ and covariance Pτ of the echo path Wτ and the mean âτ and covariance Qτ of the nonlinear coefficients aτ; or channel-wise mean Ŵτ,i and multichannel covariance Pτ of a compound quantity formed by merging together the echo path Wτ and the ith nonlinear coefficient aτ,i) and model parameters θτ are estimated; and Kalman gain factor(s) Kτ are computed for optimal adaptation of the posterior of unknown states. An echo signal Ŷτ is estimated using the multichannel input signal [Xτ,1, . . . Xτ,p] and the adapted posterior; and an error signal Eτ is generated. Residual echo is suppressed by post-filtering the error signal Eτ with a weighting function ψτ which depends on the adapted posterior, and the filtered error signal ŝ′t is then transmitted to a far-end.
摘要:
The invention relates to a method and an apparatus for processing an audio signal, wherein the method comprises the steps of: filtering an audio signal into at least two frequency band signals, generating for each frequency band signal a plurality of sub-band signals, wherein for at least one frequency band signal the plurality of sub-band signals are generated using a time to frequency domain transform and for the at least one other frequency band the plurality of sub-band signals for the other frequency band are generated using a sub-band filterbank, and the apparatus comprises at least one processor and at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processor, cause the apparatus to perform the method.
摘要:
An apparatus for extending the bandwidth of an audio signal, the apparatus being configured to: generate an excitation signal from an audio signal, wherein in the audio signal comprises a plurality of frequency components; extract a feature vector from the audio signal, wherein the feature vector comprises at least one frequency domain component feature and at least one time domain component feature; determine at least one spectral shape parameter from the feature vector, wherein the at least one spectral shape parameter corresponds to a sub band signal comprising frequency components which belong to a further plurality of frequency components; and generate the sub band signal by filtering the excitation signal through a filter bank and weighting the filtered excitation signal with the at least one spectral shape parameter.
摘要:
Both a cascade and a multichannel joint Bayesian estimator are provided for suppressing acoustic echo. An expansion basis (Power/Fourier series) is selected to convert a sample-based input signal xt into a DFT-domain multichannel signal [Xτ,1, . . . Xτ,p]. The posterior of unknown states (e.g., mean Ŵτ and covariance Pτ of the echo path Wτ and the mean âτ and covariance Qτ of the nonlinear coefficients aτ; or channel-wise mean Ŵτ,i and multichannel covariance Pτ of a compound quantity formed by merging together the echo path Wτ and the ith nonlinear coefficient aτ,i) and model parameters θτ are estimated; and Kalman gain factor(s) Kτ are computed for optimal adaptation of the posterior of unknown states. An echo signal Ŷτ is estimated using the multichannel input signal [Xτ,1, . . . Xτ,p] and the adapted posterior; and an error signal Eτ is generated. Residual echo is suppressed by post-filtering the error signal Eτ with a weighting function ψτ which depends on the adapted posterior, and the filtered error signal ŝ′t is then transmitted to a far-end.