WO2010070016A1 - Method and apparatus for applying reverb to a multi-channel audio signal using spatial cue parameters - Google Patents
Method and apparatus for applying reverb to a multi-channel audio signal using spatial cue parameters Download PDFInfo
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- WO2010070016A1 WO2010070016A1 PCT/EP2009/067350 EP2009067350W WO2010070016A1 WO 2010070016 A1 WO2010070016 A1 WO 2010070016A1 EP 2009067350 W EP2009067350 W EP 2009067350W WO 2010070016 A1 WO2010070016 A1 WO 2010070016A1
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S3/00—Systems employing more than two channels, e.g. quadraphonic
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- G—PHYSICS
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- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K15/00—Acoustics not otherwise provided for
- G10K15/08—Arrangements for producing a reverberation or echo sound
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
- H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
Definitions
- the invention relates to methods and systems for applying reverb to a multichannel downmixed audio signal indicative of a larger number of individual audio
- this is done by upmixingthe input signal and applying reverb to at least some of its individual channels in response to at least one spatial cue parameter (indicative of least one spatial cue for the input signal) so as to apply different reverb impulse responses for each of the individual channels to which reverb is applied.
- the individual channels are downmixed to generate
- the input signal is a QMF (quadrature mirror filter) domain MPEG Surround (MPS) encoded signal, and the upmixing and reverb application are performed in the QMF domain in response to MPS spatial cue parameters including at least some of Channel Level Difference (CLD), Channel Prediction Coefficient (CPC), and Inter-channel Cross Correlation (ICC)
- CLD Channel Level Difference
- CPC Channel Prediction Coefficient
- ICC Inter-channel Cross Correlation
- reverberator (or “reverberator system”) is used to denote a system configured to apply reverb to an
- audio signal (e.g., to all or some channels of a multi-channel audio signal).
- system is used in a broad sense to denote a device, system, or subsystem.
- a subsystem that implements a reverberator may be referred to as a reverberator system (or reverberator), and a system including such a reverberator subsystem (e.g., a decoder
- reverberator system (or reverberator).
- production of signals by speakers denotes causing the speakers to produce sound in response to the signals, including by performing any required amplification and/or other processing of the signals.
- linear combination of values vi, V2, ..., V n , (e.g., n elements of a subset of a set of X individual audio channel signals occuring at a time, t, where n is less than or equal to X) denotes a value equal to aivi + a2V2 + ...+ a n v n , where a ⁇ , a 2 , ..., a n are coefficients.
- each coefficient can be positive or negative or zero).
- the expression is used in a broad sense herein, for example to cover the case that one of the coefficients is equal to 1 and the others are equal to zero (e.g., the case that the linear combination aivi + a2V2 + ...+ a n v n is equal to vi(or v2, ..., or V n ).
- spatial cue parameter of a multichannel audio signal denotes any parameter indicative of at least one spatial cue for the audio signal, where each such "spatial cue” is indicative (e.g., descriptive) of the spatial image of the multichannel signal.
- spatial cues are level (or intensity) differences between (or ratios of) pairs of the channels of the audio signal, phase differences between such channel pairs, and measures of correlation between such channel pairs.
- spatial cue parameters are the Channel Level Difference (CLD) parameters and Channel Prediction Coefficient (CPC) parameters which are part of a conventional MPEG Surround (“MPS”) bitstream, and which are employed in MPEG surround coding.
- CLD Channel Level Difference
- CPC Channel Prediction Coefficient
- M M channels
- N N decoded audio channels
- a typical, conventional MPS decoder is operable to perform upmixingto generate N decoded audio channels (where N is greater than two) in response to a time-domain, 2- channel, downmixed audio input signal (and MPS spatial cue parameters including Channel Level Difference and Channel Prediction Coefficient parameters).
- a typical, conventional MPS decoder is operable in a binaural mode to generate a binaural signal in response to a time-domain, 2-channel, downmixed audio input signal and spatial cue parameters, and in at least one other mode to perform upmixing to generate 5 0 (where the notation "x.y” channels denotes “x” full frequency channels and "y” subwoofer channels), 5 1, 7.0, or 7.1 decoded audio channels in response to a time-domain, 2- channel, downmixed audio input signal and spatial cue parameters.
- the input signal undergoes time domain-to-frequency domain transformation into the QMF (quadrature mirror filter) domain, to generate two channels of QMF domain frequency components
- QMF quadrature mirror filter
- N is greater than two, and N is typically equal to 5 or 7
- L' and R' 2-channel downmixed audio signal
- MPS spatial cue parameters including Channel Level Difference parameters and Channel Prediction Coefficient parameters
- the downmixed input signal (L' and R') is indicative of "X" individual audio channels, where X is greater than 2.
- the downmixed input signal is typically indicative of five individual channels (e.g., left-front, right front, center, left-surround, and right-surround channels)
- Each of the "left" input signal L' and the "right” input signal R' is a sequence of QMF domain frequency components generated by transforming a 2 channel, time- domain MPS encoded signal (not indicated in Fig 1) in a time doma ⁇ n-to QMF domain transform stage (not shown in Fig 1)
- the downmixed input signals L' and R' are decoded into IM individual channel signals Sl, S2, . ., SN, in decoder 1 of Fig 1, in response to the MPS spatial cue parameters which are asserted (with the input signals) to the Fig. 1 system.
- the N sequences of output QMF domain frequency components, Sl, S2, .., SN are typically transformed back into the time domain by a QMF domain to-time domain transform stage (not shown in Fig 1), and can be asserted as output from the system without undergoing post-processing
- SN undergo postprocessing (in the QMF domain) in post-processor 5 to generate an N-channel audio output signal comprising channels OUTl, OUT2, , OUTN
- the N sequences of output QMF domain frequency components, OUTl, OUT2, . , OUTN are typically transformed back into the time domain by a QMF doma ⁇ n-to time domain transform stage (not shown in Fig 1), and asserted as output from the system
- a binaural mode operating in a binaural mode generates 2 channel binaural audio output Sl and S2, and optionally also 2-channel binaural audio output OUTl and 0UT2, in response to a 2-channel downmixed audio signal (L' and R') and MPS spatial cue parameters (including Channel Level Difference parameters and Channel Prediction Coefficient parameters).
- L' and R' 2-channel downmixed audio signal
- MPS spatial cue parameters including Channel Level Difference parameters and Channel Prediction Coefficient parameters
- post-processor 5 When reproduced by a pair of headphones, the 2-chan ⁇ el audio output Sl and S2 is perceived at the listener's eardrums as sound from "X" loudspeakers (where X >2 and X is typically equal to 5 or 7) at any of a wide variety of positions (determined by the coefficients of decoder 1), including positions in front of and behind the listener In the binaural mode, post-processor 5 can apply reverb to the 2-channel output (Sl, S2) of decoder 1 (in this case, post-processor 5 implements an artificial reverberator).
- X loudspeakers
- the 2-channel output of post-processor 5 (OUTl and OUT2) is a binaural audio output to which reverb has been applied, and which when reproduced by headphones is perceived at the listener's eardrums as sound from "X" loudspeakers (where X >2 and X is typically equal to 5) at any of a wide variety of positions, including positions in front of and behind the listener
- Reproduction of signals Sl and S2 (or OUTl and OUT2) generated during binaural mode operation of the Fig. 1 decoder can give the listener the experience of sound that comes from more than two (e g , five) "surround” sources At least some of these sources are virtual. More generally, it is conventional for virtual surround systems to use head- related transfer functions (HRTFs) to generate audio signals (sometimes referred to as virtual surround sound signals) that, when reproduced by a pair of physical speakers (e g , loudspeakers positioned in front of a listener, or headphones) are perceived at the listener's eardrums as sound from more than two sources (e.g., speakers) at any of a wide variety of positions (typically including positions behind the listener).
- HRTFs head- related transfer functions
- the MPS decoder of Fig. 1 operating in the binaural mode could be implemented to apply reverb using an artificial reverberator implemented by postprocessor 5.
- This reverberator could be configured to generate reverb in response to the two-channel output (Sl, S2) of decoder 1 and to apply the reverb to the signals Sl and S2 to generate reverbed two-channel audio OUTl and 0UT2.
- the reverb would be applied as a post process stereo-to-stereo reverb to the 2-channel signal Sl, S2 from decoder 1, such that the same reverb impulse response is applied to all discrete channels determined by one of the two downmixed audio channels of the binaural audio output of decoder 1 (e.g., to left-front and left-surround channels determined by downmixed channel Sl), and the same reverb impulse response is applied to all discrete channels determined by the other one of the two downmixed audio channels of the binaural audio (e g., to right front and right-surround channels determined by downmixed channel S2).
- a reverberator has what is known as a Feedback Delay Network-based (FDN-based) structure
- FDN-based Feedback Delay Network-based
- An advantage of this structure relative to other reverb structures is the ability to efficiently produce and apply multiple uncorrelated reverb signals to multiple input signals
- This feature is exploited in the commercially available Dolby Mobile headphone virtualizer which includes a reverberator having FDN based structure and is operable to apply reverb to each channel of a five channel audio signal (having left-front, right-front, center, left- surround, and right surround channels) and to filter each reverbed channel using a different filter pair of a set of five head related transfer function (“HRTF”) filter pairs
- HRTF head related transfer function
- the Dolby Mobile headphone virtualizer is also operable in response to a two- channel audio input signal, to generate a two channel "reverbed" audio output (a two- channel virtual surround sound output to which reverb has been applied)
- a two channel "reverbed” audio output a two- channel virtual surround sound output to which reverb has been applied
- the reverbed audio output is reproduced by a pair of headphones, it is perceived at the listener's eardrums as H RTF-filtered, reverbed sound from five loudspeakers at left front, right front, center, left rear (surround), and right rear (surround) positions.
- the virtualizer upmixes a downmixed two-channel audio input (without using any spatial cue parameter received with the audio input) to generate five upmixed audio channels, applies reverb to the upmixed channels, and downmixes the five reverbed channel signals to generate the two-channel reverbed output of the virtualizer.
- the reverb for each upmixed channel is filtered in a different pair of HRTF filters.
- the inventor has recognized that it would be desirable to individually determine (and generate) a different reverb signal for each of the discrete channels of an upmix determined from downmixed audio, from each of the discrete channels of the upmix, or to determine and generate a different reverb signal for (and from) each of a set of linear combinations of values of such discrete channels.
- the inventor has also recognized that with such individual determination of reverb signals for the individual upmix channels (or linear combinations of values of such channels), reverb having a different reverb impulse response can be applied to the upmix channels (or linear combinations)
- the invention is a method for applying reverb to an M- channel downmixed audio input signal indicative of X individual audio channels, where X is a number greater than M.
- the method includes the steps of:
- each of the reverb channel signals at a time, f is a linear combination of at least a subset of values of the X individual audio channels at the time, t;
- reverb applied to at least one of the reverb channel signals has a different reverb impulse response than does the reverb applied to at least one other one of the reverb channel signals.
- X Y, but in other embodiments X is not equal to Y.
- Y is greater than M, and the input signal is upmixed in step (a) in response to the spatial cue parameters to generate the Y reverb channel signals. In other embodiments, Y is equal to M orY is less than M.
- the input signal is a sequence of values L(t), R(t) indicative of five individual channel signals, Lfront, Rfront, C, Lsur, and R S ur.
- Each of the five individual channel signals is a sequence of values
- W is an MPEG Surround upmix matrix of form W 12
- the input signal is an M-channel, MPEG Surround (“MPS") downmixed signal
- steps (a) and (b) are performed in the QMF domain
- the spatial cue parameters are received with the input signal.
- the spatial cue parameters may be or include Channel Level Difference (CLD) parameters and/or Channel Prediction Coefficient (CPC) parameters of the type comprising part of a conventional MPS bitstream.
- CLD Channel Level Difference
- CPC Channel Prediction Coefficient
- the invention typically includes the step of transforming this time-domain signal into the QMF domain to generate QMF domain frequency components, and performing steps (a) and (b) in the QMF domain on these frequency components.
- the method also includes a step of generating an N-channel downmixed version of the Y reverbed channel signals (including each of the channel signals to which reverb has been applied and each of the channel signals, if any, to which reverb has not been applied), for example by encoding the reverbed channel signals as an N-channel, downmixed MPS signal.
- the input downmixed signal is a 2-channel downmixed MPEG Surround (“MPS”) signal indicative of five individual audio channels (left-front, right-front, center, left-surround, and right surround channels), and reverb determined by a different reverb impulse response is applied to each of at least some of these five channels, resulting in improved surround sound quality.
- MPS MPEG Surround
- the inventive method also includes a step of applying to the reverbed channel signals corresponding head-related transfer functions (HRTFs), by filtering the reverbed channel signals in an HRTF filter.
- HRTFs head-related transfer functions
- Other aspects of the invention are a reverberator configured (e.g., programmed) to perform any embodiment of the inventive method, a virtualizer including such a reverberator, a decoder (e g , an MPS decoder) including such a reverberator, and a computer readable medium (e g , a disc) which stores code for implementing any embodiment of the inventive method
- FIG 1 is a block diagram of a conventional MPEG Surround decoder system.
- FIG 2 is a block diagram of a multiple input, multiple output, FDN based reverberator (100) that can be implemented in accordance with an embodiment of the present invention
- FIG 3 is a block diagram of a reverberator system including reverberator 100 of
- Fig. 2 conventional MPS processor 102, time doma ⁇ n-to-QMF domain transform filter 99 for transforming a multi-channel input into the QMF domain for processing in reverberator 100 and processor 102, and QMF domain-to-time domain transform filter
- the invention is a method for applying reverb to an M- channel downmixed audio input signal indicative of X individual audio channels, where X is a number greater than M, and a system configured to perform the method.
- the method includes the steps of:
- each of the reverb channel signals at a time, t is a linear combination of at least a subset of values of the X individual audio channels at the time, t, and (b) individually applying reverb to each of at least two of the reverb channel signals (e.g., in the QMF domain), thereby generating Y reverbed channel signals.
- spatial cue parameters indicative e.g , descriptive
- QMF quadrature mirror filter
- the reverb applied to at least one of the reverb channel signals has a different reverb impulse response than does the reverb applied to at least one other one of the reverb channel signals.
- X Y, but in other embodiments X is not equal to Y.
- Y is greater than M, and the input signal is upmixed in step (a) in response to the spatial cue parameters to generate the Y reverb channel signals. In other embodiments, Y is equal to IvI or Y is less than M.
- FIG. 2 is a block diagram of multiple input, multiple output, FDN-based reverberator 100 which can be implemented in a manner to be explained below to perform this method.
- Each of the reverb channel signals at a time, t is a linear combination of a subset of values of the X individual upmix audio channels at the time, t.
- matrix B upmixes the input signal to generate the reverb channel signals.
- M is equal to 2.
- Element 40 is configured to add the output of gain element gl (i.e., apply feedback from the output of gain element gl) to reverb channel signal Ul.
- Element 41 is configured to add the output of gain element g2 to reverb channel signal U2.
- Element 42 is configured to add output of gain element g3 to reverb channel signal U3.
- Element 43 is configured to add the output of gain element g4 to reverb channel signal U4; scattering matrix 32 (matrix "A"), which is coupled to receive the outputs of addition elements 40, 41, 42, and 43.
- Matrix 32 is preferably a 4 * 4 unitary matrix configured to assert a filtered version of the output of each of addition elements 40, 41, 42, and 43 to a corresponding one of delay lines, z 'M> , where 0 ⁇ k - 1 ⁇ 3, and is preferably a fully populated matrix in order to provide maximum diffuseness.
- z-MA ⁇ are labeled respectively as delay lines 50, 51, 52, and 53 in Fig. 2; gain elements, gk, where 0 ⁇ k - 1 ⁇ 3, which apply gain the outputs of delay lines, z 'Mt , thus providing damping factors for controlling the decay time of the reverb applied in each upmix channel.
- Each gain element, gk is typically combined with a low- pass filter.
- the gain elements apply different, predetermined gain factors for the different QMF bands.
- Reverbed channel signals Rl, R2, R3, and R4, respectively, are asserted at the outputs of gain elements gl, g2, g3, and g4; and post-mix matrix 34 (matrix "C"), which is an N * 4 matrix coupled and configured to downmix and/or upmix (and optionally to perform other filtering on) the reverbed channel signals Rl, R2, R3, and R4 asserted at the outputs of gain elements gk, in response to at least a subset (e.g., all or some) of the spatial cue parameters asserted to matrix 30, thereby generating an N-channel, QMF domain, downmixed, reverbed audio output signal comprising channels Sl, S2, ..., and SN.
- matrix 34 is a constant matrix whose coefficients do not vary with time in response to any spatial cue parameter.
- the inventive system has Y reverb channels (where Y is less than or greater than four), pre-mix matrix 30 is configured to generate Y discrete reverb channel signals in response to the downmixed, M-channel, input signal and the spatial cue parameters, scattering matrix 32 is replaced by an YxY matrix, and the inventive system has Y delay lines, z 'Mt .
- a pre-mix matrix (a variation on matrix 30 of Fig. 2) generates two discrete reverb channel signals (e.g., in the quadrature mirror filter or "QMF" domain): one a mix of the front channels; the other a mix of the surround channels.
- Reverb having a short decay response is generated from (and applied to) one reverb channel signal and reverb having a long decay response is generated from (and applied to) the other reverb channel signal (e.g., to simulate a room with "live end/dead end” acoustics).
- post-processor 36 optionally is coupled to the outputs of matrix 34 and operable to perform post-processing on the downmixed, reverbed output Sl, S2, ..., SN of matrix 34, to generate an N-channel post-processed audio output signal comprising channels OUTl, 0UT2, ..., and OLJTN.
- N 2
- the Fig. 2 system outputs a binaural, downmixed, reverbed audio signal Sl, S2 and/or a binaural, post-processed, downmixed, reverbed audio output signal OUT, 0UT2.
- L left
- C center
- R right
- LS left-surround
- RS right-surround
- post-mix matrix 34 is omitted and the inventive reverberator outputs Y-channel reverbed audio (e.g., upmixed, reverbed audio) in response to an M-channel downmixed audio input.
- matrix 34 is an identity matrix.
- the Fig. 2 system has four reverb channels and four delay lines, z 'M " , variations on the system (and other embodiments of the inventive reverberator) implement more than or less than four reverb channels.
- the inventive reverberator includes one delay line per reverb channel.
- the input signal is an M-channel
- the input signal asserted to the inputs of matrix 30 comprises QMF domain signals INl(t,f), IN2(t,f) and INM(t,f), and the Fig. 2 system performs processing (e.g., in matrix 30) and reverb application thereon in the QMF domain.
- the spatial cue parameters asserted to matrix 30 are typically Channel Level Difference (CLD) parameters and/or Channel Prediction
- CPC Coefficient
- ICC Inter-channel Cross Correlation
- the inventive method would include a preliminary step of transforming this time-domain signal into the QMF domain to generate QMF domain frequency components, and would perform above-described steps (a) and (b) in the QMF domain on these frequency components.
- the FIG. 3 system includes filter 99 for transforming this time-domain signal into the QMF domain
- the FIG 3 system includes reverberator 100 (corresponding to and possibly identical to reverberator 100 of Fig 2), conventional MPS processor 102, time domain- to-QMF domain transform filter 99 coupled and configured to transform each of the time domain input channels ll(t), I2(t), .
- FIG. 3 system also includes QMF domain- to time domain transform filter 101, which is coupled and configured to transform the N channel combined output of reverberator 100 and processor 102 into the time domain Specifically, filter 99 transforms time-domain signals ll(t), I2(t), , and IM(t) respectively into QMF domain signals INl(t,f), !N2(t,f), ..., and INM(t,f), which are asserted to reverberator 100 and processor 102.
- QMF domain- to time domain transform filter 101 which is coupled and configured to transform the N channel combined output of reverberator 100 and processor 102 into the time domain
- filter 99 transforms time-domain signals ll(t), I2(t), , and IM(t) respectively into QMF domain signals INl(t,f), !N2(t,f), ..., and INM(t,f), which are asserted to reverberator 100 and processor 102.
- Each of the N channels output from processor 102 is combined (in an adder) with the corresponding reverbed channel output of reverberator 100 (Sl, S2, ..., or SN indicated in Fig 2, or one of OUTl, 0UT2, ..., or OUTN indicated in Fig. 2 if reverberator 100 of Fig 3 also includes a post processor 36 as shown in Fig 2).
- Filter 101 of Fig. 3 transforms the combined (reverbed) output of reverberator 100 and processor 102 (N sequences of QMF domain frequency components Sl'(t, f), S2 (t,f), .
- the input downmixed signal is a 2- channel downmixed MPS signal indicative of five individual audio channels (left-front, right-front, center, left-surround, and right surround channels), and reverb determined by a different reverb impulse response is applied to each of these five channels, resulting in improved surround sound quality.
- the coefficients of the constant matrices B and C would not change as a function of time in response to spatial cue parameters indicative of the downmixed input audio, and the so-modified Fig. 2 system would operate in a conventional stereo-to-stereo reverb mode.
- reverb having the same reverb impulse response would be applied to each individual channel in the downmix (i.e., left-front channel content in the downmix would receive reverb having the same impulse response as would right-front channel content in the downmix).
- the Fig. 2 system can produce and apply reverb to each reverb channel determined by the downmixed input to the system, with individual reverb impulse responses for each of the reverb channels.
- less reverb is applied in accordance with the invention to a center channel (for clearer speech/dialog) than to at least one other reverb channel so that the impulse response of the reverb applied each of these reverb channels is different.
- the impulse responses of the reverb applied to different reverb channels are not based on different channel routing to matrix 30 and are instead simply different scale factors applied by pre-mix matrix 30 or post-mix matrix 34 (and/or at least one other system element) to different reverb channels.
- matrix 30 is a 4 x 2 matrix having time-varying coefficients which depend on current values of coefficients, wij, where / ranges from 1 to 3 and j ranges from 1 to 2.
- the input signal is a sequence of QMF domain value pairs
- INl(t,f) L(t)
- IN2(t,f) R(t)
- Lf rO nt a sequence of values of five individual channel signals
- the time-varying coefficients of matrix 30 would depend also on the following four, time-varying channel gain values, in which CLD Xi ls is the current value of the left front/surround CLD parameter, and CLD n IS is the current value of the right front/surround CLD parameter;
- the time-varying coefficients of matrix 30 would be:
- the matrix multiplication performed by matrix 30 (having the coefficients shown in Equation 3) can be represented as:
- matrix multiplication is equivalent to an upmix to five individual channel signals (by the MPEG Surround upmix matrix W defined above) followed by a downmix of these five signals to the four reverb channel signals by matrix Bo.
- matrix 30 is implemented with the following coefficients:
- K hf , K Rf , K c , K 15 and K RS are fixed reverb gain values for the different channels, and gif, g ⁇ s , grf, gif, and wii to W32 are as in Equation 2 and Ia, respectively.
- the four fixed reverb gain values are substantially equal to each other, except that K 0 typically has a slightly lower value than the others (a few decibels lower than the values of the others) in order to apply less reverb to the center channel (e.g., for dryer sounding speech/dialog).
- Matrix 30 implemented with the coefficients of Equation 4, is equivalent to the product of the MPEG Surround upmix matrix W defined above and the following downmix matrix Bo:
- reverb channels Ul, U2, U3, and U4 would cause reverb channels Ul, U2, U3, and U4, respectively, to be the left-front upmix channel (feeding branch 1' of the Fig. 2 system), the right-front upmix channel (feeding branch 2' of the Fig. 2 system), the left-surround upmix channel (feeding branch 3' of the Fig. 2 system), and a combined right-surround and center upmix channel (the right-surround channel plus the center channel) feeding branch 4' of the Fig. 2 system.
- the reverb individually applied to the four branches of the Fig. 2 system would have individually determined impulse responses.
- matrix 30's coefficients are determined in another manner in response to available spatial cue parameters.
- matrix 30's coefficients are determined in response to available MPS spatial cue parameters to cause matrix 30 to implement a TTT upmixer operating in a mode other than in a prediction mode (e.g., an energy mode with or without center subtraction). This can be done in a manner that will be apparent to those of ordinary skill in the art given the present description, using the well known upmixing formulas for the relevant cases that are described in the MPEG standard (ISO/IEC 23003-1:2007).
- matrix 30 is a 4 x 1 matrix having time-varying coefficients:
- the coefficients are gain factors are derived from the CLD parameters CiD 11 h , CLD n IS , CLD 0 h and CLD 1 , , available as part of a conventional MPS bitstream.
- discrete reverb channels e.g., upmix channels
- other spatial cue parameters are employed to upmix a downmixed input signal (e.g., including by control channel weighting).
- ICC parameters available as part of a conventional MPS bitstream) that describe front-back diffuseness are used to determine coefficients of the pre-mix matrix and thereby to control reverb level.
- the inventive method also includes a step of applying to the reverbed channel signals corresponding head-related transfer functions (HRTFs), by filtering the 5 reverbed channel signals in an HRTF filter.
- HRTFs head-related transfer functions
- matrix 34 of the Fig. 2 system is preferably implemented as the HRTF filter which applies such HRTFs to, and also performs the above-described downmixing operation on, reverbed channels Rl, R2, R3, and R4.
- Such implementation of matrix 34 would typically perform the same filtering as a 5 ⁇ 4 matrix followed by a 2 * 5 matrix, where the 5 x 4 matrix generates five virtual
- the HTRF filter would typically perform for each individual QMF band a matrix multiplication by a matrix with complex valued entries.
- MPS encoded, downmixed input signal are filtered with corresponding HRTFs as follows.
- the HRTFs in the parametric QMF domain essentially consist of left and right gain parameter values and Inter-channel Phase Difference (IPD) parameter values that characterize the downmixed input signal.
- IPD Inter-channel Phase Difference
- the HRTFs are constant gain values (four gain values for each of the left and the right channel, respectively): gg R i Fj r L >
- fractional delay is applied in at least one reverb channel, and/or reverb is generated and applied differently to different frequency bands of frequency components of audio data in at least one reverb channel.
- Some such preferred implementations of the inventive reverberator are variations on the Fig. 2 system that are configured to apply fractional delay (in at least one reverb channel) as well as integer sample delay.
- a fractional delay element is connected in each reverb channel in series with a delay line that applies integer delay equal to an integer number of sample periods (e.g., each fractional delay element is positioned after or otherwise in series with one of delay lines 50, 51, 52, and 53 of Fig. 2).
- Some of the above-noted preferred implementations of the inventive reverberator are variations on the Fig. 2 system that are configured to apply reverb differently to different frequency bands of the audio data in at least one reverb channel, in order to reduce complexity of the reverberator implementation.
- the audio input data, INl-INM are QMF domain MPS data
- the reverb application is performed in the QMF domain
- the reverb is applied differently to the following four frequency bands of the audio data in each reverb channel:
- reverb is applied in this band as in the above- described embodiment of Fig. 2, with matrix 30 implemented with the coefficients of Equation 4);
- One of such eight lowest frequency bands can be used as a complex QMF buffer band, so that complex-valued arithmetic calculations are performed for only seven of the eight lowest QMF frequency bands (so that reverb is applied in this relatively low frequency range as in the above-described embodiment of Fig. 2, with matrix 30 implemented with the coefficients of Equation 4), and real-valued arithmetic calculations are performed for the other 56 QMF frequency bands, with the crossover between complex valued and real valued calculations occurring at the frequency (7 * 44.1kHz)/(64*2) which is approximately equal to 2.4 kHz.
- reverb is applied in the relatively high frequency range as in the above-described Fig.
- Reverb is applied in the relatively low frequency range (below 2.4 kHz) as in the Fig. 2 embodiment, e.g., with matrix 30 implemented with the coefficients of Equation 4); 8 kHz- 15 kHz: reverb is applied in this band by a simple delay technique. For example, reverb is applied in a way similar to the manner it is applied the above- described Fig.
- pre-mix matrix 30 e.g., to apply less reverb to the center channel than to each other channel
- the two delay branches can be simply fed to left and right outputs, respectively, or can be switched so that echoes from the left front (Lf) and left surround (Ls) channels end up in the right output channel and echoes from the right front (Rf) and right surround (Rs) channels end up in the left output channel
- the 2 ⁇ 2 pre-mix matrix can have the following coefficients:
- the inventive system applies reverb to an M-channel downmixed audio input signal indicative of X individual audio channels, where X is a number greater than M, including by generating Y discrete reverb channel signals in response to the downmixed signal but not in response to spatial cue parameters
- the system individually applies reverb to each of at least two of the reverb channel signals in response to spatial cue parameters indicative of spatial image of the downmixed input signal, thereby generating Y reverbed channel signals
- the coefficients of a pre mix matrix e.g., a variation on matrix 30 of Fig 2
- a scattering matrix e.g., a variation on matrix 32 of Fig.
- a gain stage e.g., a variation on the gain stage comprising elements gl-gk of Fig. 2
- a post-mix matrix e.g., a variation on matrix 34 of Fig. 2 operates on the reverb channel signals in a manner determined by spatial cue parameters indicative of spatial image of the downmixed input signal, to apply reverb to each of at least two of the reverb channel signals.
- the inventive reverberator is or includes a general purpose processor coupled to receive or to generate input data indicative of an M- channel downmixed audio input signal, and programmed with software (or firmware) and/or otherwise configured (e.g., in response to control data) to perform any of a variety of operations on the input data, including an embodiment of the inventive method.
- a general purpose processor would typically be coupled to an input device (e.g , a mouse and/or a keyboard), a memory, and a display device.
- an input device e.g , a mouse and/or a keyboard
- 3 system could be implemented in a general purpose processor, with inputs ll(t), I2(t), ..., IM(t), being input data indicative of M channels of downmixed audio data, and outputs Sl(t), S2(t), .., SN(t), being output data indicative of N channels of downmixed, reverbed audio.
- a conventional digital-to-analog converter could operate on this output data to generate analog versions of the output audio signals for reproduction by speakers (e.g., a pair of headphones)
- speakers e.g., a pair of headphones
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JP2011541416A JP5524237B2 (en) | 2008-12-19 | 2009-12-16 | Method and apparatus for applying echo to multi-channel audio signals using spatial cue parameters |
CN2009801511567A CN102257562B (en) | 2008-12-19 | 2009-12-16 | Method and apparatus for applying reverb to a multi-channel audio signal using spatial cue parameters |
EP09801205.7A EP2377123B1 (en) | 2008-12-19 | 2009-12-16 | Method and apparatus for applying reverb to a multi-channel audio signal using spatial cue parameters |
US13/132,321 US8965000B2 (en) | 2008-12-19 | 2009-12-16 | Method and apparatus for applying reverb to a multi-channel audio signal using spatial cue parameters |
RU2011129154/08A RU2509442C2 (en) | 2008-12-19 | 2009-12-16 | Method and apparatus for applying reveberation to multichannel audio signal using spatial label parameters |
BRPI0923174-9A BRPI0923174B1 (en) | 2008-12-19 | 2009-12-16 | METHOD AND REVERBERATOR TO APPLY REVERBERATION TO AN AUDIO INPUT SIGNAL WITH DOWNMIXING OF CHANNELS |
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