Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement

Audio Rendering Based on Multiple Impulse ResponsesProposed algorithm

Experimental ResultsConclusion

Hybrid Reverberator Using Multiple ImpulseResponses for Audio Rendering Improvement

Andrea Primavera1, Stefania Cecchi1, Francesco Piazza1, Junfeng Li2, and

Yonghong Yan2

1 A3lab - DII - Universita Politecnica delle Marche -Ancona - ITALY

2 Institute of Acoustics, Chinese Academy ofSciences - Beijing - CHINA

IIHMSP, October 2013, Beijing, China.

Andrea Primavera Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement 1/22



1 Audio Rendering Based on Multiple Impulse ResponsesIntroductionState of the artProposed algorithm

2 Proposed algorithmAnalysis of reverberation tailSynthesis of the reverberation effectReal-time Reproduction of Moving Listener Position

3 Experimental ResultsExperimental SetupReverberation TimeClarity IndexSubjective Analysis

4 ConclusionConclusionQuestionsBibliography




IntroductionState of the artProposed algorithm

Linear convolution is a widely used operation typically employed for audiorendering purpose aiming to reproduce the reverberation effect generatedwhen a sound is produced within an enclosed space.

LINEARCONVOLUTION

STATICPROCEDURE

It allows to reproduce only the acoustic effect produced taking intoaccount a specific sound source with the relative receiver position.

One of the main problems

• Time varying convolution to simulate the moving receiverpositions performing IRs interpolation [1].

• A large impulse response (IR) database is needed.

Solution






LINEARCONVOLUTION

STATICPROCEDURE





Solution






LINEARCONVOLUTION

STATICPROCEDURE





Solution





PROBLEM: Large impulse responses database required high memoryusage.

In [2] a database reduction procedure for auralization purpose with movinglistener position has been proposed:

• Early reflections contain most of the information regarding thelocation of the sound source and receiver.

• Late reflections gives more information about room properties(e.g., size, geometry, materials) [3].

• The information in the late reverberation tail is largelyredundant across multiple impulse responses recorded in thesame space.

Consideration






In [2] a database reduction procedure for auralization purpose with movinglistener position has been proposed:

• Mixing time evaluation to discriminate late from earlyreflections.

• Approximation of the reverberation tail of the whole IRdatabase as stochastic process (i.e., white noise).

Metodology






Taking into account the procedure described in [2] a novel metho-dology has been proposed considering the advantages introduced byhybrid reveberation structure [4] [5].

• It is possible to approximate the convolution operation usingrecursive structure (i.e., IIR filters).

• This procedure allows to further reduce the database dimensionwith respect to [2].

• The employed structures permits to decrease the computationalload required to perform the real-time auralization.

Proposed Solution




Analysis of reverberation tailSynthesis of the reverberation effectReal-time Reproduction of Moving Listener Position

Three are the main phases of the approach presented for the reproductionof moving listener position exploiting a hybrid reverberator structure:

1 Analysis of reverberation tail:Generate a prototype representing the database average reveberationtail.

2 Synthesis of the reverberation effect:Approximate the reverberation tail prototype exploiting an hybridreverberation algorithm [4] [5].

3 Real-time reproduction of moving listener position:Reverberation effect reproduction using the hybrid reverberationstructure (mixed FIR/IIR filter network).





1 Analysis of reverberation tail:

• Mixing time analysis:The partitioning of early from late reflections has been performedexploiting gaussianity [4] [6] and phase evolution estimators [7].

• Prototype evaluation:The reverberation tail prototype is computed as a mean of the IRsdatabase after the maximum mixing time tm.

htail =1

N

Lm∑t=tm

hn(t) (1) hn: database IRs

• Scaling operation:In order to simulate the distance among the source and the differentlistereners position a scaling factor is evaluated.

Sm =RMS(hs )

RMS(hm)=

√1

Lm

Lm∑t=tm

h2s (t)√

1Lm

Lm∑t=tm

h2m(t)

(2)

hs : synthesized IRhm: original IR





2 Synthesis of the reverberation effect:

+EARLY

REFLECTIONS

DEVICE

LATE

REFLECTIONS

DEVICE

DELAY ×x[n] y[n]

gain

Hybrid reverberator block diagram for the single channel case.

Based on the convolution with areal IR for the reproduction of theearly echoes.

Early reflections device

Based on IIR filters network (e.g.,comb and/or all-pass) and a FDNmatrix [8] for the simulation of thereverberation tail.

Late reflections device

LBCF

+x[n]

LBCF

LBCF

LBCF

AP AP

A

++

++

++

+

+y[n]

NAP filters

NLBCF filters

Late reflections device block diagram

for the single channel case.






+EARLY

REFLECTIONS

DEVICE

LATE

REFLECTIONS

DEVICE

DELAY ×x[n] y[n]

gain

Hybrid reverberator block diagram for the single channel case.

An automatic procedure allows toset the parameters of hybrid re-verberator in order to emulate areal environment starting from itsimpulse.

FIR TO IIR APPROXIMATION

Autotuning procedure

LBCF

+x[n]

LBCF

LBCF

LBCF

AP AP

A

++

++

++

+

+y[n]

NAP filters

NLBCF filters

Late reflections device block diagram

for the single channel case.






Two are the main phases of the autotuning procedure:

Evaluation of the mixing time to set theearly reflection device:

• Gaussianity estimators:Similarities between IR behaviorand gaussian noise can be foundin late reflections.Kurtosis and MAD/SD ratiohave been used.

• Phase distortion evaluation:The unwrapped phase of the IRtends to become not linear withlate reflections evolution.

Early Reflections Partitioning An offline adaptation procedure, ba-sed on SPSA [9], has been used toiteratively find the IIR parameters.A single loss function computed in cep-stral domain [10] has been adopted inthe minimization procedure.

L = max

max

√√√√√ K∑

i=1

M∑j=1

[Tr (i, j) − Ta(i, j)]2

where:• Tr is a matrix representing the

MFCC derived from the real IR.

• Ta is the MFCC obtained by theartificial IR.

Late Reflections Analysis





3 Real-time reproduction of moving listener position:

The movement reproduction is generate changing the reverberator para-meters as a function of the listener position:

• Early reflection device: The filter coefficientsare obtained as alinear or bilinear interpolation of the impulse response, with relationto the number of the involved IR.

• Late reflection device: The coefficients are fixed reproducing thesame reverberation tail for all the different positions.




Experimental SetupReverberation TimeClarity IndexSubjective Analysis

The effectiveness of the presented technique has been proved taking intoaccount account the IR database of a real environment (i.e., St. MargaretsChurch in York [11]).

As reported in [11], a total of 18IRs has been derived using:

• A logarithmic sweep signalexcitation (20 Hz - 22 kHz).

• Sample rate of 96kHz.

• A Soundfield SPS422Bmicrophone.

• A Genelec S30D loudspeaker.





The reverberation time as a function of frequency has been analyzed inorder to provide an objective evaluation between real and synthesize IRs.

Energy Decay Relief (EDR) of one of

the eighteen real IR.

Energy Decay Relief (EDR) of

artificial IR.





The reverberation time as a function of frequency has been analyzed inorder to provide an objective evaluation between real and synthesize IRs.

Mean difference in reverberation time between the measured and synthesized IRs

exploiting (a) the proposed approach and (b) the technique described in [2].

Since the obtained errors are comparable, the effectiveness of the proposedtechnique in time frequency behaviors reproduction is confirmed.





Another parameter employed in objective analysis is clarity (C50 and C80):

C50

Mean real Mean synth Mean err Std errProposed approach 1.75 2.28 0.52 1.09

Approach presented in [2] 1.75 0.10 1.65 1.97

C80Proposed approach 4.23 5.49 1.26 1.06

Approach presented in [2] 4.23 3.41 0.81 1.29

Clarity measures: mean and the standard deviation (STD) error computed asdifference in corresponding receiver positions of the synthesized and measured IRs

exploiting the proposed approach and the one described in [2].

The similar values obtained as mean and standard deviation evaluationconfirms the validity of the approach.





Informal listening tests have been performed:

• The movement of the listener position along one dimension has beensimulated.

• The effectiveness of the approach has been confirmed since listenersare not able to hear any difference between the presented approachand the one described in [2].




ConclusionQuestionsBibliography

In conclusion:

• A novel approach for the reproduction of moving listener positionexploiting time variant hybrid reverberation algorithm has beenpresented.

• As confirmed in several papers the employment of IIR filter networkfor the approximation of convolution operation allows to reduce thecomputational cost required in the auralization operation, moreoverthe approach also allow to decrease the IR database size reducingthe information required for the late reflection reprodution.

• The effectiveness of the approach has been proved taking intoaccount a real IR database providing comparison with the existingstate-of-art techniques in terms of objective and subjective measures.





QUESTIONS?





B. Dalenback and M. Stromberg, “Real time walk-throughauralization - the first year,” in Proc. Institute of Acoustics,Amsterdam, NL, Mar. 2006.

R. Stewart and M. Sandler, “Generating a spatial averagereverberation tail across multiple impulse responses,” in Proc. 35thAudio Engineering Society Conference, London, UK, Dec. 2009.

B. Blesser, “An interdisciplinary synthesis of reverberationviewpoints,” J. Audio Eng. Soc., vol. 49, no. 10, pp. 867–903, Oct.2001.

A. Primavera, S. Cecchi, P. Peretti, L. Romoli, and F. Piazza, “AnAdvanced Implementation of a Digital Artificial Reverberator,” inProc. 130th Audio Engineering Society Convention, London,UK,May 2011.

A. Primavera, M. Gasparini, S. Cecchi, L. Romoli, and F. Piazza,“Hybrid Reverberation Algorithm: a Practical Approach,” inAIA-DAGA Conference, Merano, Italy, Mar. 2013.





R. Stewart and M. Sandler, “Statisical measures of early reections ofroom impulse responses,” in in DAFX 07), Bordeaux, France, Sep.2007.

G. Defrance and J. Polack, “Measuring the mixing time inauditoria,” in Proc. 155th Meeting of the Acoustical Society ofAmerica), vol. 49, Jun 2001, pp. 867–903.

J. Jot, “Digital Delay Networks for designing artificial reverberators,”in Proc. 90th Audio Engineering Society Convention, Paris, Feb1991.

J. Spall, “Implementation of the Simultaneous PerturbationAlgorithm for Stochastic Optimization,” in IEEE Transactions onAerospace and Electronic Systems, vol. 34, 1998, pp. 817–823.

S. Heise, M. Hlatky, and J. Loviscach, “Automatic Adjustment ofOff-the-Shelf Reverberation Effects,” in Proc. 126th AudioEngineering Society Convention, Munich, Germany, May 2009.





“OpenAIR, Audiolab, University of York.” [Online]. Available:http://www.openairlib.net/


http://www.openairlib.net/

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Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Improvement