摘要:
An apparatus for processing an audio signal comprises an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.
摘要:
Estimating noise in an audio signal (102) includes determining (S100) an energy value (174) for the audio signal (102); converting (S102) the energy value (174) into the log2-domain; and estimating (S104) a noise level (182) for the audio signal (102) based on the converted energy value (178) directly in the log2-domain. The energy value (174) is converted (S102) into the log2-domain as follows: E n_log = ⌊ log 2 1 + E n_lin ⋅ 2 N ⌋ 2 N └ x ┘ floor (x), indicating the largest integer less than or equal to x, E n_log energy value of band n in the log2-domain, E n _ lin energy value of band n in the linear domain, N quantization resolution. The noise estimates are transmitted as parameters in the form of Silence Insertion Descriptor, SID, frames to update the amplitude of random sequences generated in each frequency band at a decoder side during inactive phases.
摘要:
Partitioned block frequency domain adaptive filter device comprising: a frequency domain adaptive filter configured for filtering a frequency domain representation of a time domain input signal depending on a set of filter coefficients consisting of a plurality of blocks of filter coefficients in order to produce a filtered signal; a plurality of parallel arranged filter update blocks, each of the filter update blocks being configured for updating one of the blocks of filter coefficients based on an update signal gathered by a circular correlation of a block of the frequency domain representation signal and a frequency domain control signal comprising a representation of the filtered signal; wherein each of the filter update blocks comprises an adaptation module configured for executing an adaptation sequence comprising the steps of calculating an approximation of a constrained gradient update for the filter coefficients of the respective block of filter coefficients by applying an approximated constraining matrix having a lesser complexity than a constraining matrix to an unconstrained gradient update for the filter coefficients of the respective block of filter coefficients, wherein the unconstrained gradient update is derived from the update signal, and calculating a cumulative error introduced on the unconstrained gradient update by applying the approximated constraining matrix to the unconstrained gradient update; wherein each of the filter update blocks comprises a correction module configured for executing a correction sequence comprising the steps of calculating a corrected constrained gradient update for the filter coefficients of the respective block of filter coefficients by applying the frequency domain constraining matrix to a sum of the approximation of the constrained gradient update and the cumulative error.
摘要:
An apparatus for multichannel interference cancellation in a received audio signal comprising two or more received audio channels to obtain a modified audio signal comprising two or more modified audio channels is provided. The apparatus comprises a first filter unit (112) being configured to generate a first estimation of a first interference signal depending on a reference signal. Moreover, the apparatus comprises a first interference canceller (114) being configured to generate a first modified audio channel of the two or more modified audio channels from a first received audio channel of the two or more received audio channels depending on the first estimation of the first interference signal. Furthermore, the apparatus comprises a second filter unit (122) being configured to generate a second estimation of a second interference signal depending on the first estimation of the first interference signal. Moreover, the apparatus comprises a second interference canceller (124) being configured to generate a second modified audio channel of the two or more modified audio channels from a second received audio channel of the two or more received audio channels depending on the second estimation of the second interference signal.
摘要:
The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.
摘要:
An apparatus for decoding an encoded audio signal, wherein one or more tracks are associated with the encoded audio signal, each one of the tracks having a plurality of track positions and a plurality of pulses is provided. The apparatus comprises a pulse information decoder (110) and a signal decoder (120). The pulse information decoder (110) is adapted to decode a plurality of pulse positions, wherein each one of the pulse positions indicates one of the track positions of one of the tracks to indicate a position of one of the pulses of the track, and wherein the pulse information decoder is configured to decode the plurality of pulse positions by using a track positions number indicating a total number of the track positions of at least one of the tracks, a total pulses number indicating a total number of the pulses of at least one of the tracks, and one state number. The signal decoder (120) is adapted to decode the encoded audio signal by generating a synthesized audio signal using the plurality of pulse positions and a plurality of predictive filter coefficients being associated with the encoded audio signal.
摘要:
The invention provides a decoder being configured for processing an encoded audio bitstream, wherein the decoder includes: a bitstream decoder configured to derive a decoded audio signal from the bitstream, wherein the decoded audio signal includes at least one decoded frame; a noise estimation device configured to produce a noise estimation signal containing an estimation of the level and/or the spectral shape of a noise in the decoded audio signal; a comfort noise generating device configured to derive a comfort noise signal from the noise estimation signal; and a combiner configured to combine the decoded frame of the decoded audio signal and the comfort noise signal in order to obtain an audio output signal.
摘要:
A method is described that estimates noise in an audio signal (102). An energy value (174) for the audio signal (102) is estimated (S100) and converted (S102) into the logarithmic domain. A noise level for the audio signal (102) is estimated (S104) based on the converted energy value (178).