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Paper ID 8333
An Advanced Implementation of a
Digital Artificial Reveberator
A. Primavera1, S. Cecchi1, L. Romoli1, P. Peretti1 and F. Piazza1
1
A3Lab - DIBET - Universit`a Politecnica delle Marche
Via Brecce Bianche 1, 60131 Ancona Italy
www.a3lab.dibet.univpm.it
Abstract
Reverberation is a well known effect particularly important for listening of recorded
and live music. In this paper we propose a real implementation of an enhanced
approach for digital artificial reverberator. Starting from a preliminary analysis of
the mixing time, the selected impulse response is decomposed in time domain con-
sidering the early and the late reflections. Therefore, a short FIR filter is used to
synthesize the first part of the impulse response, and a generalized recursive struc-
ture based on IIR filters is used to synthesize the late reflections, exploiting a min-
imization criterion in the cepstral domain. Several results are reported taking into
consideration different real impulse responses and comparing the results with those
obtained with a previous proposed technique in terms of computational complexity
and reverberation quality.
Introduction
Reverberation is probably the most used audio effect employed by musician during live performances
and recording session.
• MEASURED IR: the desired signal can be obtained by convolving the input signal with a mea-
sured impulse response.
Pros: accurate reproduction of the acoustic environment.
Cons: computational complexity bounded to the IR length.
• SYNTHETIC IR: the reverberation effect can be obtained using an IIR structure (e.g. comb
and/or allpass).
Pros: Great flexibility and high computational efficiency.
Cons: Low accuracy.
State of the Art
We propose a Hybrid Reverberator (HR) based on both approaches: this solution attempts to
create a parametric and realistic reverberation, with a low computational cost.
Pros: Great flexibility, low computational cost and audio quality comparable to the convolution
approach.
Cons: None.
Objective of this work
Proposed Hybrid Reverberator (1)
As previously discussed in [1] [2], the proposed Hybrid Reverberator is mainly composed of two parts
as the Moorer’s Reverberator [3].
Fig.1 Hybrid Reverberator block diagram for the single channel case.
Based on the convolution with a real IR for
the reproduction of the early echoes.
Early reflections device
Based on IIR filters network (e.g., comb
and/or all-pass) and a FDN matrix [4] for
the simulation of the reverberation tail.
Late reflections device
Fig.2 Late reflections device block diagram for the single
channel case.
Proposed Hybrid Reverberator (2)
Set the parameters of the Hybrid Reverberator in order to emulate a real environment starting from
its impulse response.
Aim of the work
An offline procedure has been developed in order to determine all the parameters of the Hybrid
Reverberator starting from the IR of a real environment.
Idea
Evaluation of the mixing time to set the
Early reflection device.
Early Reflections Partitioning Use of a minimization criterium, based on
Simultaneous Perturbation Stochastic Ap-
proximation (SPSA) to set all the para-
maters value of the Late reflections device.
Late Reflections Analysis
Proposed Hybrid Reverberator (3) - Early reflections partitioning
Two main approaches are used simultaneously to evaluate the mixing time:
Similarities between IR behavior and gaussian noise can be found in late reflections. Kurtosis (k)
and MAD/SD ratio (r) have been used [5].
k =
E (x − µ)4
σ4
− 3 → 0 r =
E (|x − µ|)
σ
→
2
π
(1)
Gaussianity Estimator
The unwrapped phase of the IR tends to become not linear with late reflections evolution [6].
Phase Distortion Evaluation
Fig.3 Mixing time evolution for medium room. Fig.4 Mixing time evolution for large room.
Proposed Hybrid Reverberator (4) - Late Reflection Analysis
An offline adaptation procedure, based on SPSA [7], has been used to iteratively find the parameters
set of the IIR structure (81 parameters for each audio channel).
Fig.5 General scheme of the adaption procedure using
SPSA procedure.
Fig.6 Evolution of the loss function L.
4 loss functions (It, If, Iff, IT60ω) and a threshold
system in [2].
New Approach
A single loss function L computed in cepstral
domain [8].
L = max



max



K
i=1
M
j=1
[Tr(i, j) − Ta(i, j)]2






where:
• Tr is a matrix representing the Mel-Frequency Cep-
stral Coefficients (MFCC) derived from the real IR.
• Ta is the MFCC obtained by the artificial IR.
Loss Function
Experimental Results - EDR
The automatic procedure for the parameters setting has been tested with 2 different real IRs, relative
to a Medium and a Large room.
Fig.7 Energy Decay Relief of the analyzed impulse responses for medium and large room.
Fig.8 Energy Decay Relief of the artificial impulse responses for medium and large room.
Experimental Results - Comparison with a previous method
The proposed method has been compared with a previous one proposed by the same authors [2].
A real-time implementation of the proposed algorithm has been realized on a DSP board (Omap-L137 TI).
The Workload results as the sum of two contributes:
• Early reflections device: it depends on the mixing
time value.
• Late reflections device: it is always equal to 13%.
WORKLOAD
Room Proposed approach Previous approach
medium 26% 21%
large 24% 19%
Computational Cost
The artificial reverberator has been compared with the approach proposed in [2] and with the real reverberation effect.
Listening Tests
Fig.9 Listening tests results.
As confirmed by the listening tests a low increase
of the required workload produces an evident im-
provement of the perceived reverberation effect.
Consideration
Conclusions
• A Hybrid Reverberator with an automatic procedure for the parameters setting has been proposed.
• The automatic procedure is based on the evaluation of the mixing time and the minimization of a single loss function
computed in the cepestral domain using the SPSA criterium.
• A real time implementation of the proposed algorithm has been realized on a DSP platform.
• Different tests have been carried out in order to evaluate reverberation quality, in terms of subjective evaluation and
objective measures.
• Listening test have been executed in order to compare the proposed algorithm with the performance obtained using
another approach proposed by the same authors.
• As confirmed by the listening tests the artificial effect generated sounds really similar to the natural one validating
the proposed approach.
• Future works will be oriented toward the refinement of the minimization criterium (Particle Swarm or Genetic algo-
rithms can be used in order to obtain better performance).
References
[1] R. Stewart and D. Murphy, “A Hybrid Artificial Reverberation Algorithm,” in Proc. 122nd Audio Engineering Society Convention (AES’07), Vienna, Austria, May 2007.
[2] A. Primavera, L. Palestini, S. Cecchi, F. Piazza, and M. Moschetti, “A Hybrid Approach for Real-Time Room Acoustic Response Simulation,” in Proc. 128th Audio Engineering Society Convention (AES’10),
London, UK, May 2010.
[3] J.A. Moorer, “About This Reverberation Business,” Computer Music Journal, vol. 3, no. 2, pp. 13–28, 1979.
[4] J. Jot, “Digital Delay Networks for designing artificial reverberators,” in Proc. 90th Audio Engineering Society Convention (AES’91), Paris, Feb 1991.
[5] R. Stewart and M. Sandler, “Statisical measures of early reections of room impulse responses,” in in DAFX 07), Bordeaux, France, Sep. 2007.
[6] G. Defrance and J.D. Polack, “Measuring the mixing time in auditoria,” in Proc. 155th Meeting of the Acoustical Society of America), Jun 2001, vol. 49, pp. 867–903.
[7] J.C. Spall, “Implementation of the Simultaneous Perturbation Algorithm for Stochastic Optimization,” in IEEE Transactions on Aerospace and Electronic Systems, 1998, vol. 34, pp. 817–823.
[8] S. Heise, M. Hlatky, and J. Loviscach, “Automatic Adjustment of Off-the-Shelf Reverberation Effects,” in Proc. 126th Audio Engineering Society Convention (AES’09), Munich, Germany, May 2009.

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An Advanced Implementation of a Digital Artificial Reverberator

  • 1. Paper ID 8333 An Advanced Implementation of a Digital Artificial Reveberator A. Primavera1, S. Cecchi1, L. Romoli1, P. Peretti1 and F. Piazza1 1 A3Lab - DIBET - Universit`a Politecnica delle Marche Via Brecce Bianche 1, 60131 Ancona Italy www.a3lab.dibet.univpm.it Abstract Reverberation is a well known effect particularly important for listening of recorded and live music. In this paper we propose a real implementation of an enhanced approach for digital artificial reverberator. Starting from a preliminary analysis of the mixing time, the selected impulse response is decomposed in time domain con- sidering the early and the late reflections. Therefore, a short FIR filter is used to synthesize the first part of the impulse response, and a generalized recursive struc- ture based on IIR filters is used to synthesize the late reflections, exploiting a min- imization criterion in the cepstral domain. Several results are reported taking into consideration different real impulse responses and comparing the results with those obtained with a previous proposed technique in terms of computational complexity and reverberation quality.
  • 2. Introduction Reverberation is probably the most used audio effect employed by musician during live performances and recording session. • MEASURED IR: the desired signal can be obtained by convolving the input signal with a mea- sured impulse response. Pros: accurate reproduction of the acoustic environment. Cons: computational complexity bounded to the IR length. • SYNTHETIC IR: the reverberation effect can be obtained using an IIR structure (e.g. comb and/or allpass). Pros: Great flexibility and high computational efficiency. Cons: Low accuracy. State of the Art We propose a Hybrid Reverberator (HR) based on both approaches: this solution attempts to create a parametric and realistic reverberation, with a low computational cost. Pros: Great flexibility, low computational cost and audio quality comparable to the convolution approach. Cons: None. Objective of this work
  • 3. Proposed Hybrid Reverberator (1) As previously discussed in [1] [2], the proposed Hybrid Reverberator is mainly composed of two parts as the Moorer’s Reverberator [3]. Fig.1 Hybrid Reverberator block diagram for the single channel case. Based on the convolution with a real IR for the reproduction of the early echoes. Early reflections device Based on IIR filters network (e.g., comb and/or all-pass) and a FDN matrix [4] for the simulation of the reverberation tail. Late reflections device Fig.2 Late reflections device block diagram for the single channel case.
  • 4. Proposed Hybrid Reverberator (2) Set the parameters of the Hybrid Reverberator in order to emulate a real environment starting from its impulse response. Aim of the work An offline procedure has been developed in order to determine all the parameters of the Hybrid Reverberator starting from the IR of a real environment. Idea Evaluation of the mixing time to set the Early reflection device. Early Reflections Partitioning Use of a minimization criterium, based on Simultaneous Perturbation Stochastic Ap- proximation (SPSA) to set all the para- maters value of the Late reflections device. Late Reflections Analysis
  • 5. Proposed Hybrid Reverberator (3) - Early reflections partitioning Two main approaches are used simultaneously to evaluate the mixing time: Similarities between IR behavior and gaussian noise can be found in late reflections. Kurtosis (k) and MAD/SD ratio (r) have been used [5]. k = E (x − µ)4 σ4 − 3 → 0 r = E (|x − µ|) σ → 2 π (1) Gaussianity Estimator The unwrapped phase of the IR tends to become not linear with late reflections evolution [6]. Phase Distortion Evaluation Fig.3 Mixing time evolution for medium room. Fig.4 Mixing time evolution for large room.
  • 6. Proposed Hybrid Reverberator (4) - Late Reflection Analysis An offline adaptation procedure, based on SPSA [7], has been used to iteratively find the parameters set of the IIR structure (81 parameters for each audio channel). Fig.5 General scheme of the adaption procedure using SPSA procedure. Fig.6 Evolution of the loss function L. 4 loss functions (It, If, Iff, IT60ω) and a threshold system in [2]. New Approach A single loss function L computed in cepstral domain [8]. L = max    max    K i=1 M j=1 [Tr(i, j) − Ta(i, j)]2       where: • Tr is a matrix representing the Mel-Frequency Cep- stral Coefficients (MFCC) derived from the real IR. • Ta is the MFCC obtained by the artificial IR. Loss Function
  • 7. Experimental Results - EDR The automatic procedure for the parameters setting has been tested with 2 different real IRs, relative to a Medium and a Large room. Fig.7 Energy Decay Relief of the analyzed impulse responses for medium and large room. Fig.8 Energy Decay Relief of the artificial impulse responses for medium and large room.
  • 8. Experimental Results - Comparison with a previous method The proposed method has been compared with a previous one proposed by the same authors [2]. A real-time implementation of the proposed algorithm has been realized on a DSP board (Omap-L137 TI). The Workload results as the sum of two contributes: • Early reflections device: it depends on the mixing time value. • Late reflections device: it is always equal to 13%. WORKLOAD Room Proposed approach Previous approach medium 26% 21% large 24% 19% Computational Cost The artificial reverberator has been compared with the approach proposed in [2] and with the real reverberation effect. Listening Tests Fig.9 Listening tests results. As confirmed by the listening tests a low increase of the required workload produces an evident im- provement of the perceived reverberation effect. Consideration
  • 9. Conclusions • A Hybrid Reverberator with an automatic procedure for the parameters setting has been proposed. • The automatic procedure is based on the evaluation of the mixing time and the minimization of a single loss function computed in the cepestral domain using the SPSA criterium. • A real time implementation of the proposed algorithm has been realized on a DSP platform. • Different tests have been carried out in order to evaluate reverberation quality, in terms of subjective evaluation and objective measures. • Listening test have been executed in order to compare the proposed algorithm with the performance obtained using another approach proposed by the same authors. • As confirmed by the listening tests the artificial effect generated sounds really similar to the natural one validating the proposed approach. • Future works will be oriented toward the refinement of the minimization criterium (Particle Swarm or Genetic algo- rithms can be used in order to obtain better performance). References [1] R. Stewart and D. Murphy, “A Hybrid Artificial Reverberation Algorithm,” in Proc. 122nd Audio Engineering Society Convention (AES’07), Vienna, Austria, May 2007. [2] A. Primavera, L. Palestini, S. Cecchi, F. Piazza, and M. Moschetti, “A Hybrid Approach for Real-Time Room Acoustic Response Simulation,” in Proc. 128th Audio Engineering Society Convention (AES’10), London, UK, May 2010. [3] J.A. Moorer, “About This Reverberation Business,” Computer Music Journal, vol. 3, no. 2, pp. 13–28, 1979. [4] J. Jot, “Digital Delay Networks for designing artificial reverberators,” in Proc. 90th Audio Engineering Society Convention (AES’91), Paris, Feb 1991. [5] R. Stewart and M. Sandler, “Statisical measures of early reections of room impulse responses,” in in DAFX 07), Bordeaux, France, Sep. 2007. [6] G. Defrance and J.D. Polack, “Measuring the mixing time in auditoria,” in Proc. 155th Meeting of the Acoustical Society of America), Jun 2001, vol. 49, pp. 867–903. [7] J.C. Spall, “Implementation of the Simultaneous Perturbation Algorithm for Stochastic Optimization,” in IEEE Transactions on Aerospace and Electronic Systems, 1998, vol. 34, pp. 817–823. [8] S. Heise, M. Hlatky, and J. Loviscach, “Automatic Adjustment of Off-the-Shelf Reverberation Effects,” in Proc. 126th Audio Engineering Society Convention (AES’09), Munich, Germany, May 2009.