摘要:
A method and a device for processing a stereo signal obtained from an encoder, which codes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (L 0 , R 0 ). A first signal and a third signal are added in order to obtain a first output signal (L 0w ), wherein the first signal QL 0wL ) comprises the first stereo signal (L 0 ) modified by a first complex function (g 1 ), and the third signal (L 0wR ) comprises the second stereo signal (R 0 ) modified by a third complex function (g 3 ). A second signal and a fourth signal are added to obtain a second output signal (R 0w ). The fourth signal (R 0wR ) comprises the second stereo signal (R 0 ) modified by a fourth complex function (g 4 ), and the second signal (R 0wL ) comprises the first stereo signal (L 0 ) modified by a second complex function (g 2 ). The complex functions (g 1 ,g 2 ,g 3 ,g 4 ) are functions of the spatial parameters (P) and are chosen such that an energy value of the difference (L 0wL -P 0wL ) between the first signal and the second signal is larger than or equal to the energy value of the sum (L 0wL +R 0wL ) of the first and the second signal and the energy value of the difference (R 0wR -L 0wR ) between the fourth signal and the third signal is larger than or equal to the energy value of the sum (R 0wR +L 0wR ) of the fourth signal and the third signal.
摘要:
Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of downsampled subband signals. Further, decoding is provided wherein an encoded audio signal comprising an encoded single channel audio signal and a set of spatial parameters is decoded by decoding the encoded single channel audio channel to obtain a plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, and deriving two audio channels from the spatial parameters, the sub-subband signals and those downsampled subband signals that are not further subband filtered.
摘要:
An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.
摘要:
Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of downsampled subband signals. Further, decoding is provided wherein an encoded audio signal comprising an encoded single channel audio signal and a set of spatial parameters is decoded by decoding the encoded single channel audio channel to obtain a plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, and deriving two audio channels from the spatial parameters, the sub-subband signals and those downsampled subband signals that are not further subband filtered.
摘要:
An audio encoder (109) has a hierarchical encoding structure and generates a data stream comprising one or more audio channels as well as parametric audio encoding data. The encoder (109) comprises an encoding structure processor (305) which inserts decoder tree structure data into the data stream. The decoder tree structure data comprises at least one data value indicative of a channel split characteristic for an audio channel at a hierarchical layer of the hierarchical decoder structure and may specifically specify the decoder tree structures to be applied by a decoder. A decoder (115) comprises a receiver (401) which receives the data stream and a decoder structure processor (405) for generating the hierarchical decoder structure in response to the decoder tree structure data. A decode processor (403) then generates output audio channels from the data stream using the hierarchical decoder structure.
摘要:
A multi-channel audio encoder (10) for encoding a multi-channel audio signal (101), e.g. a 5.1 channel audio signal, into a spatial down-mix (102), e.g. a stereo signal, and associated parameters (104, 105). The encoder (10) comprises first and second units (110, 120). The first unit (110) encodes the multi-channel audio signal (101) into the spatial down-mix (102) and parameters (104). These parameters (104) enable a multi-channel decoder (20) to reconstruct the multi-channel audio signal (203) from the spatial down-mix (102). The second unit (120) generates, from the spatial down-mix (102), parameters (105) that enable the decoder to reconstruct the spatial down-mix (202) from an alternative down-mix (103), e.g. a so-called artistic down-mix that has been manually mixed in a sound studio. In this way, the decoder (20) can efficiently deal with a situation in which an alternative down-mix (103) is received instead of the regular spatial, down-mix (102). In the decoder (20), first the spatial down-mix (202) is reconstructed from the alternative down-mix (103) and the parameters (105). Next, the spatial down-mix (202) is decoded into the multi-channel audio signal (203).
摘要:
A decoder receives (501) a bitstream comprising an encoded mono signal and stereo data. A time scale processor (503) generates a time scaled mono signal. A time-tofrequency processor generates frequency sample blocks of the time scaled signal, the block length being fixed and independent of the time scaling. A parametric stereo decoder (509) generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (511). A synchronization processor (515) synchronizes the stereo data with the time scaled signal by determining a time association between a parameter value and a frequency sample block. The parameter value and time association is used to determine synchronized stereo parameter values for that and other frequency sample blocks. The invention is particularly suitable for low complexity generation of time scaled stereo signals from MPEG-4 encoded signals.
摘要:
A device (1) for converting a first number (M) of input audio channels into a second, larger number (N) of output audio channels comprises: decorrelation units (3) for decomposing the input audio channels into a set of decorrelated auxiliary channels, at least one upmix unit (4) for combining the decorrelated auxiliary channels into the output audio channels, and at least one pre-processing unit (2) for pre-processing the input audio channels and feeding the pre-processed input audio channels to the decorrelation units (3). The pre-processing unit (2) and the upmix unit (4) are preferably controlled by audio parameters.
摘要:
An audio encoder for encoding a multi-channel audio signal includes an encoder combination module (ECM) for generating a dominant signal part (m) and a residual signal part (s) being a combined representation of first and second audio signals (x1, x2), the dominant and residual signal parts (m, s) being obtained by applying a mathematical procedure to the first and second audio signals (x1, x2), wherein the mathematical procedure involves a first spatial parameter (SP1) including a description of spatial properties of the first and second audio signals (x1, x2), a parameter generator (PG) for generating a first parameter (PS1) set including a second spatial parameter (SP2), and a second parameter (PS2) set including a third spatial parameter (SP3), and an output generator for generating an encoded output signal having a first output part (OP1) including the dominant signal part (m) and the first parameter set (PS1), and a second output part (OP2) including the residual signal part (s) and the second parameter set (PS2).
摘要:
Synthesizing an output audio signal is provided on the basis of an input audio signal, the input audio signal comprising a plurality of input sub-band signals, wherein at least one input sub-band signal is transformed (T) from the sub-band domain to the frequency domain to obtain at least one respective transformed signal, wherein the at least one input sub-band signal is delayed and transformed (D, T) to obtain at least one respective transformed delayed signal, wherein at least two processed signals are derived (P) from the at least one transformed signal and the at least one transformed delayed signal, wherein the processed signals are inverse transformed (T-1) from the frequency domain to the sub-band domain to obtain respective processed sub-band signals, and wherein the output audio signal is synthesized from the processed sub-band signals.