摘要:
Plusieurs modes de realisation d'une technique sont presentes pour tirer profit de la correlation entre des signaux de maniere a fournir des informations utilisables concernant la frequence. Dans un mode specifique de realisation, un filtre (51) est adapte a une composante de signal d'un signal d'entree utilisant une variable generee par un adaptateur (58). La variable possede une relation determinee avec la frequence de la composante du signal. La variable est utilisee pour ponderer une combinaison de valeurs de signaux derivees d'un historique de signaux pour former une prediction d'un signal futur. Un additionneur (56) combine la nouvelle valeur du signal avec la prediction pour former un signal d'erreur qui est utilise par le filtre (51) et l'adaptateur (58). Lorsque le filtre converge, un indicateur de frequence (59) utilisant la variable produit un signal de sortie indiquant la frequence de la composante du signal.
摘要:
Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.
摘要:
An adaptive apparatus for adjusting a system parameter in an adaptive manner, wherein the parameter in the form of a vector is successively adjusted by using a vector error signal, a first adaptive gain signal in the form of a matrix corresponding to the vector error signal, a matrix signal and a second adaptive gain in the form of a matrix corresponding to the matrix signal.
摘要:
A sound source separation apparatus includes: a separation-matrix processor that transforms a plurality of observation signals corresponding to sounds being propagated from a plurality of sound sources into a frequency-domain signal group the separation-matrix processor updating a separation matrix based on the frequency-domain signal group and transforming the updated separation matrix into time-series filter coefficients to output; a filter-coefficient transformer that partially removes non-causal components from the filter coefficients to transform the filter coefficients, and a separator that supplies the filter coefficients to a filter group, the separator generating a plurality of separation signals separated from the plurality of observation signals corresponding to the separation matrix.
摘要:
An adaptive apparatus for adjusting a system parameter in an adaptive manner, wherein the parameter in the form of a vector is successively adjusted by using a vector error signal, a first adaptive gain signal in the form of a matrix corresponding to the vector error signal, a matrix signal and a second adaptive gain in the form of a matrix corresponding to the matrix signal.
摘要:
Embodiments of the invention disclose a signal processing device and a signal processing method and a device and a method for signal processing. The signal processing device includes a sampling module, a first segmentation module, a second segmentation module, and a detection module. The sampling module samples an input signal to generate a sample signal. The first segmentation module calculates a first segment value according to the sample signal during a first time interval. The second segmentation module calculates a second segment value according to the sample signal during a second time interval different in length from the first time interval. The detection module generates a detection signal according to the determination of whether the first segment value lies out of a first range, and whether the second segment value lies out of a second range.
摘要:
Embodiments of the invention disclose a signal processing device and a signal processing method and a device and a method for signal processing. The signal processing device includes a sampling module, a first segmentation module, a second segmentation module, and a detection module. The sampling module samples an input signal to generate a sample signal. The first segmentation module calculates a first segment value according to the sample signal during a first time interval. The second segmentation module calculates a second segment value according to the sample signal during a second time interval different in length from the first time interval. The detection module generates a detection signal according to the determination of whether the first segment value lies out of a first range, and whether the second segment value lies out of a second range.