Not applicable.
Not applicable.
The present invention relates to audio coding.
Parametric stereo or multi-channel audio coding as described e.g. in C. Faller and F. Baumgarte, “Efficient representation of spatial audio using perceptual parametrization,” in Proc. IEEE Workshop on Appl. of Sig. Proc. to Audio and Acoust., October 2001, pp. 199-202, uses spatial cues to synthesize down-mix—usually mono or stereo—audio signals to signals with more channels. Usually, the down-mix audio signals result from a superposition of a plurality of audio channel signals of a multi-channel audio signal, e.g. of a stereo audio signal. These less channels are waveform coded and side information, i.e. the spatial cues, relating to the original signal channel relations is added to the coded audio channels. The decoder uses this side information to re-generate the original number of audio channels based on the decoded waveform coded audio channels.
A basic parametric stereo coder may use inter-channel level differences (ILD) as a cue needed for generating the stereo signal from the mono down-mix audio signal. More sophisticated coders may also use the inter-channel coherence (ICC), which may represent a degree of similarity between the audio channel signals, i.e. audio channels. Furthermore, when coding binaural stereo signals e.g. for 3D audio or headphone based surround rendering, also an inter-channel phase difference (IPD) may play a role to reproduce phase/delay differences between the channels.
The synthesis of ICC cues may be relevant for most audio and music contents to re-generate ambience, stereo reverb, source width, and other perceptions related to spatial impression as described in J. Blauert, Spatial Hearing: The Psychophysics of Human Sound Localization, The MIT Press, Cambridge, Mass., USA, 1997. Coherence synthesis may be implemented by using de-correlators in frequency domain as described in E. Schuijers, W. Oomen, B. den Brinker, and J. Breebaart, “Advances in parametric coding for high-quality audio,” in Preprint 114th Conv. Aud. Eng. Soc., March 2003. However, the known synthesis approaches for synthesizing multi-channel audio signals may suffer from an increased complexity.
A goal to be achieved by the present invention is to provide an efficient concept for synthesizing a multi-channel audio signal from a down-mix audio signal.
The invention is based on the finding, that a multi-channel audio signal may efficiently be synthesized from a down-mix audio signal upon the basis of at least three signal copies of the down-mix audio signal. The down-mix audio signal may comprise e.g. a sum of a left audio channel signal and a right audio channel signal of a multi-channel audio signal, e.g. of a stereo audio signal. Thus, a first copy may represent a first audio channel, a second copy may represent a diffuse sound and a third copy may represent a second audio channel. In order to synthesize, e.g. generate, the multi-channel audio signal, the second copy may be used to generate two de-correlated signals which may respectively be combined with the respective audio channel in order to synthesize the multi-channel audio signal. In order to obtain the two de-correlated signals, the second copy may be pre-stored or delayed in particular in frequency domain. However, the de-correlated signals may be obtained directly in time domain. In both cases, a low complexity arrangement may be achieved.
According to a first implementation form, the invention relates to an audio signal synthesizer for synthesizing a multi-channel audio signal from a down-mix audio signal, the audio signal synthesizer comprising a transformer for transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal, the transformed audio signal representing a spectrum of the down-mix audio signal, a signal generator for generating a first auxiliary signal, for generating a second auxiliary signal, and for generating a third auxiliary signal upon the basis of the transformed audio signal, a de-correlator for generating a first de-correlated signal, and for generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated, and a combiner for combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and for combining the second auxiliary signal with the second de-correlated signal to obtain the second audio signal, the first audio signal and the second audio signal forming the multi-channel audio signal. The transformer may be a Fourier transformer or a filter bank for providing e.g. a short-time spectral representation of the down-mix audio signal. In this regard, the de-correlated signals may be regarded as being de-correlated if a first cross-correlation value of a cross-correlation between these signals is less than another cross-correlation value of the cross-correlation.
According to an implementation form of the first aspect, the transformer comprises a Fourier transformer or a filter to transform the down-mix audio signal into frequency domain. The Fourier transformer may be e.g. a fast Fourier transformer.
According to an implementation form of the first aspect, the transformed audio signal occupies a frequency band, wherein the first auxiliary signal, the second auxiliary signal and the third auxiliary signal share the same frequency sub-band of the frequency band. Correspondingly, the other sub-bands of the frequency band may correspondingly be processed.
According to an implementation form of the first aspect, the signal generator comprises a signal copier for providing signal copies of the transformed audio signal, a first multiplier for multiplying a first signal copy by a first weighting factor for obtaining a first weighted signal, a second multiplier for multiplying a second signal copy by a second weighting factor for obtaining a second weighted signal, and a third multiplier for multiplying a third signal copy by a third weighting factor for obtaining a third weighted signal, and wherein the signal generator is configured to generate the auxiliary signals upon the basis of the weighted signals. The weighting factors may be used to adjust or scale the power of the respective signal copy to the respective first audio channel, second audio channel and the diffuse sound.
According to an implementation form of the first aspect, the audio signal synthesizer comprises a transformer for transforming the first weighted signal into time domain to obtain the first auxiliary signal, for transforming the second weighted signal into time domain to obtain the second auxiliary signal, and for transforming the third weighted signal into time domain to obtain the third auxiliary signal. The transformer may be e.g. an inverse Fourier transformer.
According to an implementation form of the first aspect, the first weighting factor depends on a power of a right audio channel of the multi-channel audio signal, and wherein the second weighting factor depends on a power of a left audio channel of the multi-channel audio signal. Thus, the power of both audio channels may respectively be adjusted.
According to an implementation form of the first aspect, the de-correlator comprises a first storage for storing a first copy of the third auxiliary signal in frequency domain to obtain the first de-correlated signal, and a second storage for storing a second copy of the third auxiliary signal in frequency domain to obtain the second de-correlated signal. The first storage and the second storage may be configured for storing the copy signals for different time periods in order to obtain de-correlated signals.
According to an implementation form of the first aspect, the de-correlator comprises a first delay element for delaying a first copy of the third auxiliary signal to obtain the first de-correlated signal, and a second delay element for delaying a second copy of the third auxiliary signal to obtain the second de-correlated signal. The delay elements may be arranged in time domain or in frequency domain.
According to an implementation form of the first aspect, the de-correlator comprises a first all-pass filter for filtering a first copy of the third auxiliary signal to obtain the first de-correlated signal, and a second all-pass filter for filtering a second copy of the third auxiliary signal to obtain the second de-correlated signal. Each all-pass filter may be formed by an all-pass network, by way of example.
According to an implementation form of the first aspect, the de-correlator comprises a first reverberator for reverberating a first copy of the third auxiliary signal to obtain the first de-correlated signal, and a second reverberator for reverberating a second copy of the third auxiliary signal to obtain the second de-correlated signal.
According to an implementation form of the first aspect, the combiner is configured to add up the first auxiliary signal and the first de-correlated signal to obtain the first audio signal, and to add up the second auxiliary signal and the second de-correlated signal to obtain the second audio signal. Thus, the combiner may comprise adders for adding up the respective signals.
According to an implementation form of the first aspect, the audio signal synthesizer further comprises a transformer for transforming the first audio signal and the second audio signal into time domain. The transformer may be e.g. an inverse Fourier transformer.
According to an implementation form of the first aspect, the first audio signal represents a left channel of the multi-channel audio signal, wherein the second audio signal represents a right channel of the multi-channel audio signal, and wherein the de-correlated signals represent a diffuse audio signal. The diffuse audio signal may represent a diffuse sound.
According to an implementation form of the first aspect, the audio signal synthesizer further comprises an energy determiner for determining an energy of the first de-correlated signal and an energy of the second de-correlated signal, a first energy normalizer for normalizing the energy of the first de-correlated signal, and a second energy normalizer for normalizing the energy of the second de-correlated signal.
According to a second aspect, the invention relates to a method for synthesizing, e.g. for generating, a multi-channel audio signal, e.g. a stereo audio signal, from a down-mix audio signal, the method comprising transforming the down-mix audio signal into frequency domain to obtain a transformed audio signal, the transformed audio signal representing a spectrum of the down-mix audio signal, generating a first auxiliary signal, a second auxiliary signal and a third auxiliary signal upon the basis of the transformed audio signal, generating a first de-correlated signal from the third auxiliary signal, and generating a second de-correlated signal from the third auxiliary signal, the first de-correlated signal and the second de-correlated signal being at least partly de-correlated, and combining the first auxiliary signal with the first de-correlated signal to obtain a first audio signal, and combining the second auxiliary signal with the second de-correlated signal to obtain the second channel signal, the first audio signal and the second audio signal forming the multi-channel audio signal.
According to some embodiments, a method for generating a multi-channel audio signal from a down-mix signal may comprise the steps of: receiving a down-mix signal, converting the input down-mix audio signal to a plurality of subbands, applying factors in the subband domain to generate subband signals representing correlated and un-correlated signal of a target multi-channel signal, converting the generated subband signals to the time-domain, de-correlating the generated time-domain signals representing un-correlated signal, and combining the time-domain signals representing correlated signal with the de-correlated signals.
According to a fourth aspect, the invention relates to a computer program for performing the method for synthesizing a multi-channel audio signal when executed on a computer.
Further embodiments of the invention will be described with respect to the following figures, in which:
The transformer 101 may be e.g. a Fourier transformer or any filter bank (FB) which is configured to provide a short time spectrum of the down-mix signal. The down-mix signal may be generated upon the basis of combining a left channel and a right channel of e.g. a recorded stereo signal, by way of example.
The signal generator 103 may comprise a signal copier 109 providing e.g. three copies of the transformed audio signal. For each copy, the audio signal synthesizer may comprise a multiplier. Thus, the signal generator 103 may comprise a first multiplier 111 for multiplying a first copy by a first weighting factor w1, a second multiplier 113 for multiplying a second copy by a second weighting factor w3, and a third multiplier 115 for multiplying a third copy by a weighting factor w2.
According to some embodiments, the multiplied copies form weighted signals Y1(k, i), D(k, i) and Y2(k, i) which may respectively be provided to the inverse transformers 117, 119 and 121. The inverse transformers 117 to 121 may e.g. be formed by inverse filter banks (IFB) or by inverse Fourier transformers. At the outputs of the inverse transformers 117 to 121, the first, second and third auxiliary signals may be provided. In particular, the third auxiliary signal at the output of the inverse transformer 119 is provided to the de-correlator 105 comprising a first de-correlating element D1 and a second de-correlating element D2. The de-correlating elements D1 and D2 may be formed e.g. by delay elements or by reverberation elements or by all-pass filters. By way of example, the de-correlating elements may delay copies of the third auxiliary signal with respect to each other so that a de-correlation may be achieved. The respective de-correlated signals are provided to the combiner 107 which may comprise a first adder 123 for adding a first de-correlated signal to the first auxiliary signal to obtain the first audio signal, and a second adder 125 for adding the second de-correlated signal to the second auxiliary signal to obtain the second audio signal.
As depicted in
With reference to
The generated time-frequency representation of the three signals, Y1(k,i), Y2(k,i), and D(k,i), are converted back to the time domain by using an IFB or an inverse transformer. By way of example, two independent de-correlators D1 and D2 are applied to d(n) in order to generate two at least partly independent signals, which are added to y1(n) and y2(n) to generate e.g. the final stereo output left and right signals, i.e. first and second audio signals, z1(n) and z2(n).
With reference to generating or computing the weighting factors, if an amplitude of the downmix signal is |M|=g√{square root over (|L|2+|R|2)}, L and denoting the amplitudes of the left, L, and right, R, channel, then, at the decoder, the relative power of the left and right channels are known according to the following formulas based on the ICLD:
It shall be noted that in the following, for brevity of notation, the indices k and i are often neglected.
Given the ICC (coherence) the amount of diffuse sound in the left and right channels, PD(k,i), can be computed according to the formula:
Before using further, PD may be lower bounded by zero and upper bounded by the minimum of P1 and P2.
The weighting factors are computed such that the resulting three signals Y1, Y2, and D may have powers equal to P1, P2, and PD, i.e.:
where the power of the down-mix audio signal is P=1 since P1, P2, and PD may be normalized, and the factor of g relates to the normalization that is used for the down-mix input signal. In the conventional case, when the down-mix signal may be the sum multiplied by 0.5, and g may be chosen to be 0.5.
If the amplitude of the downmix signal is
then some adaptations may be made. The channel level differences (CLDs) may be applied to the downmix at the decoder side using the following formulas for c1 and c2:
The definitions for c1 and c2 may allow recovering the correct amplitude for the left and the right channel.
P1 and P2 may be defined according to the previous definition as:
leading to
Then PD may be defined based on the above P1 and P2 as aforementioned.
If a case is considered where ICC=1, and if the amplitude of the downmix signal is assumed to be
then the definition of P1, P2 and PD may be used and applied on the downmix signal, yielding:
To cancel the effect of the mismatch between downmix computation and the assumption on P1 and P2 factors, some adaptations of the above formulas may be performed. Assuming:
For the downmix signal defined as
the w1, w2 and w3 may be adapted to keep the energy of the left and right channel according to:
w1=2√{square root over ((P1−Pd)*factor)}
w2=2√{square root over ((P2−Pd)*factor)}
w3=2√{square root over ((Pd)*factor)}
In the case ICC=1, the definitions of w1, w2 and w3 allow to obtain exactly the same result as with the weighting factor c1 and c2.
Another alternative adaptation method is described in the following:
In a stereo coder based on CLD, there are two gains for left and right channel, respectively. The gains may be multiplied to the decoded mono signal to generate the reconstructed left and right channel.
The gains may thus be calculated according to the following equations:
These gain factors may be used to compute:
P1=c12
P2=c22
P=P1+P2
These P1, P2 and P may further be used to calculate the w1, w2 and w3 as aforementioned. The factors w1, w2 and w3 may be scaled by
and then applied to the left, right and diffuse signal, respectively.
Alternatively, as opposed to computing the signals Y1, Y2, and D to have a power of P1, P2, and PD, respectively, a Wiener filter may be applied to approximate the true signals Y1, Y2, and D in a least mean squares sense. In this case, the Wiener filter coefficients are:
Regarding the de-correlators, the diffuse signal in the time domain before de-correlation, d(n), has the short-time power spectra desired for the diffuse sound, due to the way how the scale factors w1, w2, and w3 were computed. Thus, the goal is to generate two signals d1(n) and d2(n) from d(n) using de-correlators without changing the signal power and short-time power spectra more than necessary.
For this purpose, two orthogonal filters D1 and D2 with unity L2 norm may be used. Alternatively one may use orthogonal all-pass filters or reverberators in general. For example, two orthogonal finite impulse response (FIR) filters, suitable for de-correlation are:
D1(n)=w(n)n1(n)
D2(n)=w(n)n2(n)
where n1(n) is a random variable, such as a white Gaussian noise for indices 0≦n≦M and otherwise zero. n2(n) is similarly defined as random variable independent of n1(n). The window w(n) can for example be chosen to be a Hann window with an amplitude such that the L2 norm of the filters D1(n) and D2(n) is one.
A second copy of the third auxiliary signal is provided to the first delay element D1 which output is provided to a first energy normalizer 305 normalizing an output of the first delay element D1 e.g. with respect to its energy E(D1). An output of the first energy normalizer 305 is multiplied with the output of the multiplier 303 by a multiplier 307, which output is provided to the adder 123.
A third copy of the third auxiliary signal is provided to the second delay element D2 which output is provided to a second energy normalizer 309 normalizing an output of the second delay element D2 e.g. with respect to its energy E(D2). An output of the second energy normalizer 309 is multiplied with the output of the multiplier 303 by a multiplier 311, which output is provided to the adder 125.
In
Still in reference to
A low complexity way of doing de-correlation is simply using different delays for D1 and D2. This approach may exploit the fact that the signal representing de-correlated sound d(n) contains little transients. By way of example, the delays of 10 milliseconds (ms) and 20 ms for D1 and D2 may be used.
This application is a continuation of International Application No. PCT/CN2010/075308, filed on Jul. 20, 2010, which is hereby incorporated by reference in its entirety.
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Number | Date | Country | |
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20130129096 A1 | May 2013 | US |
Number | Date | Country | |
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Parent | PCT/CN2010/075308 | Jul 2010 | US |
Child | 13744690 | US |