Method for processing an audio signal, signal processing unit, binaural renderer, audio encoder and audio decoder
View Patent ↗A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is processed with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing of the late reverberation has generating a scaled reverberated signal, the scaling being dependent on the audio signal. The processed early part of the audio signal and the scaled reverberated signal are combined.
1. A method for processing an audio signal in accordance with a room impulse response, the method comprising:
applying the audio signal as an input signal to an early part processor and to a late reverberation processor;
processing, by the early part processor, the audio signal with an early part of the room impulse response to obtain a processed audio signal;
receiving, by the late reverberation processor, predefined reverberator parameters and processing the audio signal using the predefined reverberator parameters in accordance with a late reverberation of the room impulse response to obtain a reverberated signal and scaling the reverberated signal to obtain a scaled reverberated signal; and
combining the processed audio signal and the scaled reverberated signal,
wherein scaling the reverberated signal by the late reverberation processor comprises
setting a gain factor according to a predefined correlation measure of the audio signal, the predefined correlation measure having a fixed value determined empirically on the basis of an analysis of a plurality of audio signals, and applying the gain factor to the reverberated signal, or
obtaining a gain factor using a correlation analysis of the audio signal, and applying the gain factor to the reverberated signal.
2. The method of claim 1 , wherein the scaling is dependent on a condition of one or more input channels of the audio signal, wherein the condition of the one or more input channels of the audio signal comprises one or more of the number of input channels, the number of active input channels, and an activity in the one or more input channels.
3. The method of claim 2 , wherein the gain factor is determined based on the condition of the one or more input channels of the audio signal.
4. The method of claim 3 , wherein the gain factor is determined as follows:
g=c u +ρ·( c c −c u )
where
ρ=predefined or calculated correlation measure for the audio signal,
c u , c c =factors indicative of the condition of one or more input channels of the audio signal, with c u referring to totally uncorrelated channels, and c c relating to totally correlated channels,
wherein c u and c c are determined as follows:
c
u
=
10
10
·
l
og
10
(
K
in
)
20
=
K
in
c
c
=
10
20
·
l
og
10
(
K
in
)
20
=
K
in
where
K in =number of active input channels of the audio signal.
5. The method of claim 1 , wherein the correlation analysis of the audio signal comprises determining for an audio frame of the audio signal a combined correlation measure, and wherein the combined correlation measure is calculated by combining correlation coefficients for a plurality of channel combinations of one audio frame, each audio frame comprising one or more time slots.
6. The method of claim 5 , wherein combining the correlation coefficients comprises averaging a plurality of correlation coefficients of the audio frame.
7. The method of claim 5 , wherein determining the combined correlation measure comprises:
(i) calculating an overall mean value for every channel of the one audio frame,
(ii) calculating a zero-mean audio frame by subtracting the mean values from the corresponding channels,
(iii) calculating for a plurality of channel combination the correlation coefficient, and
(iv) calculating the combined correlation measure as the mean of a plurality of correlation coefficients.
8. The method of claim 5 , wherein the correlation coefficient for a channel combination is calculated as follows:
ρ
[
m
,
n
]
=
1
(
N
-
1
)
·
∑
i
∑
j
x
m
[
i
,
j
]
·
x
n
[
i
,
j
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*
∑
j
σ
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x
m
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j
]
)
·
σ
(
x
n
[
j
]
)
where
ρ[m, n]=correlation coefficient,
σ(x m [j])=standard deviation across one time slot j of channel m,
σ(x n [j])=standard deviation across one time slot j of channel n,
x m ,x n =zero-mean variables,
i∀[1, N]=frequency bands,
j∀[1, M]=time slots,
m, n∀[1, K]=channels,
*=complex conjugate.
9. The method of claim 1 , comprising delaying the scaled reverberated signal to match a start of the scaled reverberated signal to the transition point from early reflections to late reverberation in the room impulse response.
10. The method of claim 1 , wherein the audio signal is a multichannel audio input signal, and wherein processing, by the late reverberation processor, the audio signal with the late reverberation comprises applying the multichannel audio input signal to a downmixer for downmixing the multichannel audio input signal to a signal comprising a lower number of channels and applying the downmixed audio signal to a reverberator.
11. A non-transitory digital storage medium having stored thereon a computer program with program code for carrying out the method of claim 1 when being executed by a computer.
12. A signal processing unit, comprising:
an input for receiving an audio signal;
an early part processor receiving as input signal the received audio signal, wherein the early part processor is to process the received audio signal in accordance with an early part of a room impulse response to obtain a processed audio signal;
a late reverberation processor receiving as input signal the received audio signal, wherein the late reverberation processor is to receive predefined reverberator parameters to process the received audio signal using the predefined reverberator parameters in accordance with a late reverberation of the room impulse response to obtain a reverberated signal and to scale the reverberated signal to obtain a scaled reverberated signal; and
an output for combining the processed audio signal and the scaled reverberated signal into an output audio signal,
wherein the late reverberation processor is to scale the reverberated signal by
setting a gain factor according to a predefined correlation measure of the audio signal, the predefined correlation measure having a fixed value determined empirically on the basis of an analysis of a plurality of audio signals, and applying the gain factor to the reverberated signal, or
obtaining a gain factor using a correlation analysis of the audio signal, and applying the gain factor to the reverberated signal.
13. The signal processing unit of claim 12 , wherein the late reverberation processor comprises:
a reverberator receiving the audio signal and generating a reverberated signal; and
a gain stage coupled to an input or to an output of the reverberator and controlled by the gain factor.
14. The signal processing unit of claim 12 , comprising a correlation analyzer generating the gain factor dependent on the audio signal.
15. The signal processing unit of claim 12 , further comprising at least one of:
a low pass filter coupled to a gain stage, and
a delay element coupled between the gain stage and an adder, the adder further coupled to the early part processor and the output.
16. A binaural renderer, comprising the signal processing unit of claim 12 .
17. An audio encoder for coding audio signals, comprising:
the signal processing unit of claim 12 or the binaural renderer of claim 16 .
18. An audio decoder for decoding encoded audio signals, comprising:
the signal processing unit of claim 12 or the binaural renderer of claim 16 .