Reverberation Reverberation Reverberation

Reverberation Der everberation Dereverberation

Dereverberation Dereverberation
By Staff Technical Writer


n this article we summarize recent AES convention papers dealing with reverberation and its artificial generation, analysis, and enhancement. How can some of the characteristics of reverberation be measured in a perceptually relevant way? How can reverberation be removed successfully from other wanted audio signals? It is also important to consider ways in which reverberation may be implemented in platforms designed for interactive virtual environments, where computational resources may be shared with visual processing and numerous changing sources have to be processed in real time.
ARTIFICIAL REVERBERATION Although artificial reverberation processors have been in existence for many years, research is still continuing into more efficient and better sounding algorithms. Vickers et al., in “Frequency Domain Artificial Reverberation Using Spectral Magnitude Decay” (AES 121st paper 6926), explore the concept of producing artificial reverberation in the frequency domain. Most extant algorithms, they point out, are either based on feedback-delay networks in the time domain or on convolution in the frequency domain. The former have a relatively low computational cost and provide control over some perceptually relevant parameters; but it is expensive to implement multiband equalizers in the feedback paths, so any control over frequency-dependent decay tends to be limited to a few bands. On the other hand, those based on convolution in the frequency domain are very effective at simulating specific physical spaces, but it is more difficult to control individual parameters, and such systems are computationally expensive for long decay times. In theory, the method described by the authors would allow for detailed control over the frequency-dependent decay time and require less memory than feedback-delay networks. The authors use a technique inspired by the phase vocoder, shown in Fig. 1, which essentially consists of a time-toJ. Audio Eng. Soc., Vol. 55, No. 3, 2007 March

frequency transform based on the shorttime fourier transform (STFT), followed by modifications to the phase and magnitude components of the frequency spectrum, followed by an inverse transform and reconstruction of the time-domain signal. They investigate a range of techniques for generating reverberation and time-freezing effects, mainly based on the accumulation of successive frames of spectral magnitude and phase information and the successive attenuation of the magnitude components over time. They find that the processing of phase information in successive frames during the decay is crucial to the generation of a perceptually natural reverberation. For this reason they have to generate an artificial phase signal that can be combined with the the accumulated magnitude response, such that the reverb’s impulse response resembles noise with an exponential decay. Some of the issues to be overcome here include establishing the right tradeoff between phase coherence and randomization, as well as the avoidance of roughness and periodicity in the decay structure. The authors find that an algorithm with phase randomization applied at the output works quite effectively although the echo density does not necessarily increase with time. However, they argue that since late reverberation is normally said to begin where individual reflections cannot ➥



Rectangular to Polar




Polar to Rectangular




Fig. 1. Phase vocoder (courtesy Vickers et al.)

3. though not as high as that of the best reverberation devices based on feedback-delay networks in the time domain or on convolution in the frequency domain. The quality of the output is found to be reasonably good. The impulse response. Soc. stemming from the early days of audio effects. Typical spring reverb impulse response and spectrogram a b Fig. Helical spring with driver and receiver at either end Fig. 2). shown in Fig. 2–6 courtesy Abel et al. leading to scattering at the junctions. (b) longitudinal. so they can be studied by observing their impulse responses. as shown in Fig. Springs can propagate waves in longitudinal... 5. 55. Spring propagation modes: (a) transverse. This appears to be caused by J. An alternative to the above types of digital reverberation. 3. They explain that springs are approximately linear and time invariant at typical operating levels for audio systems. 4. No. in “Spring Reverb Emulation Using Dispersive Allpass Filters in a Waveguide Structure” (AES 121st paper 6954). is the spring reverberator. (Figs. Abel et al. A magnetic driver at one end turns the spring so as to set up a propagating torsional wave through the spring. and attenuation filters. 190 be heard. For this reason they plan further work on aspects such as the control of modal density and the elimination of unwanted perceptual artifacts. 2007 March . Springs were originally used because they gave rise to delays between the signal applied to an exciter at one end and a receiver at the other end of a spring. or one part of the spring is wound in the opposite direction to the other.) Fig. (a) Left. transverse. attempt to analyze and emulate the performance of these classic devices using digital waveguide models. tends to show a series of decaying repetitions.and right-going waves are processed separately using delay. 2. or torsional modes (as shown in Fig. Sometimes there are multiple elements connected in series. (c) torsional. but with each reflected impulse having considerable and increasing smearing in the time domain. 3. Audio Eng. dispersion. which is detected by a similar pickup at the other end. 4. Vol.Reverberation and Der everberation Fig. (b) Separate sections of the spring reverb are connected using scattering junctions. this does not seem to be a problem. Modern devices typically use the torsional mode with two or three independent springs operating in parallel.

Reverberation and Der everberation have been tried. 55. using digital signal processing. The degree of kurtosis describes the peakedness of the distribution. 2007 March frequency . Comparison between measured (upper) and modeled (lower) impulse response spectrograms for one particular spring reverb Fig.KHz successful. Soc. One of these is known as blind dereverberation. with lower kurtosis. consisting of a number of spring sections connected using scattering junctions. are quite the opposite—they have non-white characteristics and have positive kurtosis. but they speculate ➥ 191 Fig. in that the speech signal tested had less reverberation. The authors found that this process improved upon the results of a previous study in which they had used a conventional CMA-based method.) Many audio signals. and good perceptual equivalence is claimed between the models and the original devices they aimed to emulate. had a structure based on that shown in Fig. Linear predictive coding attempts to predict the current sample on the basis of a weighted sum of previous samples. A constant modulus algorithm (CMA) is used in conjunction with a linear predictive coding (LPC) filter to dereverberate a monophonic audio signal. Because of the linear nature of these algorithms under the circumstances employed here. Vol. which tends to turn impulses into a chirp and eventually into a noiselike sound. 6. and that it has a sub-Gaussian distribution with negative kurtosis. Many different methods . The residual signal is then used as the input to the modified CMA algorithm used for dereverberation. for the Accutronics Type 8. No. One of the problems that they found in their previous work on this topic was that typical constant modulus algorithms assume that the input signal is a statistically-independent and identically-distributed sequence. DEREVERBERATION Dereverberation is the identification and removal of reverberation from other wanted audio signals. 4). This process is shown in Fig. which in this case is used in the dereverberation process because the LPC residual values tend to have a “whiter” statistical characteristic than the audio signal itself. Little energy seems to propagate above 4 kHz through most of the springs that the authors tested.KHz frequency . Huang and Kyriakakis try a novel approach based on blind deconvolution in “Blind Dereverberation of Audio Signals Using a Modified Constant Modulus Algorithm” (AES 121st paper 6974). The results were reasonably J. 6. in which the algorithm only has access to the received signal and has no knowledge of the dry signal or acoustical environment giving rise to the reverberation.. Audio Eng. tried by the authors. the blind deconvolution filter thereby derived can be applied directly to the reverberant speech signal. The error or residual in this process is the difference between the prediction and the actual value of the sample. however. One example of the comparison between the spectrograms of measured and modeled devices is shown in Fig. (A Gaussian distribution is like the statistical distribution of a white noise signal. 7. For this reason the authors employed LPC analysis. 7. 3. Overall system diagram for modified-CMA-based monaural dereverberation (courtesy Huang and Kyriakakis) low frequencies propagating faster than high frequencies (see the spectrogram in Fig. 5. The models used to emulate spring reverbs. Statistical distribution essentially describes the likelihood that a signal will have a certain amplitude.

. is a good way of finding out whether there are repetitive features in the reverberant decays. They work on a similar premise to Bitzer and Extra and colleagues. The latter two were found to be useful coarse predictors of the quality of selected reverberation algorithms. Dereverberation using a computational auditory model (CAMM) (Courtesy Zarouchas et al. a Gaussian probability distribution. In order to split the 192 Fig. but in this case it is applied to binaural impulse responses separated into 40-ms overlapping blocks. They attempted to find physical metrics that could be used to predict the perceived quality of artificial reverberation devices. as well as lateral early decay time (LEDT). then a decision device makes a subtraction of the “distortion” due to reverberation on the basis of evaluating the just-noticeable difference between the internal perceptual representations of the two versions. RLE is a measure that was introduced by Trautmann and enhanced by Blauert. Bitzer and Extra attempted to find metrics that would model some of the spatial characteristics of artificial reverberation. counting the number of reflection “taps” that lie outside this standard deviation. No. During the later. the source and reverberant versions of the signal are both processed by the CAMM. Examining binaural measures in a simular way. 55. The metrics employed included the energy-decay curve and the energydecay relief. 3. 2007 March . This is the result of a suitably diffuse. the latter being based on Jot’s 3-D surface plots of decay time against frequency. In “A Perceptual Measure for Assessing and Removing Reverberation from Audio Signals” (AES 120th paper 6702). which appears to be successful at reducing the perceived reverberance of a range of different material. Audio Eng. filters based on third octaves were used. particularly on drum sounds. Initial results suggest that there is a relationship between the objective measures and perceived quality. treat reverberation as a distortion or degradation of an otherwise dry monophonic audio signal. but these metrics cannot currently replace a listening test. Abel and Huang provide further insight into the development of metrics for predicting reverberation quality in “A Simple. They also conducted a listening test to get some initial information about the sound quality of different reverbs. Echo density was also measured as this has also been shown to be an important factor governing the perceived quality of reverberation. The authors used these metrics on a range of free and commercial reverberation devices and plug-ins and found that they were able to observe the degree of complexity in different algorithms. These were used to derive the reverberation time and the early decay time in different bands. the process they employ for dereverberation needs access to the source signal and the reverberant version of the same. and a time-angle phase scope were employed. Unlike the method of Huang and Kyriakakis. It attempts to measure the energy in each segment at a dominant angle of incidence. diffuse part of the decay there will be a dense pattern of overlapping reflections approximating Gaussian noise. the standard deviation is large so a smaller number of reflections are classified as outliers. Robust Measure of Reverberation Echo Density” (AES 121st paper 6985). Zarouchas et al.) audio frequency range into bands. They also note that real room impulse responses are nonminimum phase. The result is a representation of the energy and direction of the signal over time. mixed soundfield. room level (early) (RLE). The autocorrelation function (the similarity of the signal to itself at different time intervals). 8. Objective clarity and time variance were also measured. The authors also examined short-time histograms of the probability distribution of the decays to determine their “whiteness” (see above). as this is known to correspond to good-sounding reverberation. they assert. which could present difficulties with the type of algorithm employed here. In this way only the perceptually-significant reverberation is removed. as proposed by Griesinger. A graininess and metallic quality was observed in the sounds of some reverbs. compared with the frontal energy. and normalizing by that expected for Gaussian noise. except that Bark-scaled bands were employed below 500 Hz. such that the late part of reverberation has a smooth quality. The time-angle phase scope is based on the traditional Lissajous figure display sometimes used to analyze the coherence of stereo audio channels. Soc. and a high echo density. first of all using traditional monaural measures of the reverberation impulse response. by looking at the standard deviation of the impulse response in each time window. A computational auditory masking model (CAMM) is employed to model the perception of reverberant signals in a similar way to sound quality measurement approaches such as NMR (noiseto-mask ratio) and PAQM (perceptual audio quality measure) used for measuring the perceptual significance of codec distortions. The authors report preliminary success with this approach. As shown in Fig. Bitzer and Extra and colleagues present the results of monaural and binaural analysis tools applied to artificial reverberation. as these tend to result in annoying coloration. which pointed the authors toward future development ideas for metrics. ANALYZING REVERBERATION In two papers from the AES 121st Convention (6928 and 6981). Vol. IACCbased measures.Reverberation and Der everberation that coloration and nonstationary properties of the speech signal might be two problematic factors. 8. It is based on an analysis of short time segments of the first 80 ms of the binaural impulse response. In the early part of the decay where there are relatively few but prominent reflections. Echo density can be measured as a function of time. The authors found that a time window of 20 to 30 ms worked well as it is long enough to contain at least a few reflections and short enough J.

9) and be robust to different equalization and decay parameters. extracting less correlated signals. 0. in “A New Upmixer for Enhancement of Reverberance Imagery in Multichannel Loudspeaker Audio Scenes” (AES 121st paper 6965). The resulting plots of echo density according to the new measure were found to discriminate well between different diffusion settings of artificial reverberators (see Fig. and 1. some stereophonic recording techniques use spaced microphones or time-delay panning. The system he developed uses an adaptive filter and time-delay combination that is able to adjust input signals in both frequency and time before the difference signal between the stereo channel pair is calculated (see Fig. (The application in ques. His conjecture that this would maximize the spatial fidelity of the front source image while locating the reverberation images to the sides was largely supported by listening tests.. (3) the new system should not be dispreferred to a conventional 2/0 system. 0. By means of this approach it was possible to upmix a range of different stereo recordings to four channel surround (using two additional loudspeakers at ±120°). No. Soc. assuming this represents front sources. this new approach could be a useful indicator of the time-domain quality or texture of a reverberant signal. He devised three design criteria for a new upmixer: (1) spatial distortion of the source image in the upmixed audio scene should be minimized. attempts to deal with the problem that most upmixing approaches are designed on the assumption that stereo images are created using amplitude-panned sources. 10). assuming they are diffuse sound or reverberation. They try to differentiate between the reverberation and reflection modeling that might be required for interactive virtual acoustics and that required for applications such as architectural acoustics. reverberance image directional strength should be high from lateral (±90) directions. 3. and upmixed results were generally preferred to the original two-channel versions. 10. as Usher points out.2 (red). It is claimed that since traditional acoustical parameters for measuring reverberation have not included anything that accounts for the temporal structure of reflections. 55. REVERBERATION IN ARTIFICIAL SCENE SYNTHESIS Jot and Trivi take a novel approach to the synthesis of reverberation for interactive virtual environments in “Scene Description Model and Rendering Engine for Interactive Virtual Acoustics” (AES 120th paper 6660).Reverberation and Der everberation should have a homogenous distribution in the horizontal plane.0s and roughly highpass equalization (courtesy Abel and Huang) to have sufficient time resolution for psychoacoustic purposes. REVERBERATION ENHANCEMENT IN UPMIXING Upmixing is the term commonly applied to systems that attempt to derive multichannel stereo signals from two-channel or matrixed multichannel program material.4 (green). and these recordings are not handled well by conventional upmixers.0 (black). Echo density profiles of feedback delay network impulse responses at different diffusion settings: 0. J. 2007 March Such upmixers tend to look for highly correlated material between left and right channels. such that diagonally opposite loudspeakers showed almost zero correlation but side pairs had non-zero correlation. 9. in particular. with RT60 = 1. Usher. Vol. However. Audio Eng. (2) reverberance imagery Fig. Conceptual diagram of Usher’s upmixer 193 . and using them to drive rear/side loudspeakers.6 (blue).➥ Fig.

Reverberation and Der everberation tion is the Creative EAX environmental audio programming interface. Editor’s note: The papers reviewed in this article.cfm.aes. 55. Auditory cues should be sufficiently valid and believable to support the accompanying visual information. 194 .org/publications/ preprints/search. and all AES papers. emerging field of audio for mobile and handheld devices. an individual reflection rarely has a critical effect on perception compared to direct path components. (3) reverberation from other environments (such as adjacent spaces). go to www. 3-D audio. J. virtual audio environments are not attempting to simulate existing situations so the requirement becomes one of plausibility rather than accuracy. (4) refine early or discrete reflections. they assert.aes. for example. can be purchased online at www. 2007 March T HE P RO C E EDINGS th OF THE A E S 29 IN T E R NATIONA L C O N F ER E NC E A u d i o f o r M o b i l e a nd H and he l d D e v i c e s The 24 papers in this proceedings address the important areas of research and practical applications—coding. that when attempting to process large numbers of virtual sound sources. speech processing. Purchase online at www. Also available on CD-ROM. 188 pages. Vol. Class-D amplifiers. Soc. Audio Eng.cfm For more information call Donna Vivaro at +1 212 661 8528.cfm and www. implementations. They The priority for resources therefore is ordered like this: (1) direct path and synthetic audio—in the fast-paced..aes. only attempting to improve the simulation of early or discrete reflections when necessary and if resources allow. is out of the question. They therefore opt to prioritize the allocation of resources toward improving the shared reverberation process and the control of per-source reverberator feeds. with accurate rendering of individual reflections. For this reason a reverberation system built on a physical model of the scene. Furthermore. so they aim to reduce the complexity of the algorithms that might be used based on the principle of journal/search. (2) reverberation of listener’s environment. audio accuracy is not as crucial as in some other applications because visual cues often dominate. ext. In realistic virtual worlds. The authors prefer to contemplate traditional algorithms based on feedback-delay networks or possibly frequency-domain convolution with a measured or synthetic reverb impulse response. No.) In particular the authors are keen to develop a means of generating reflections and reverberation that has minimal computational costs. 42. 3. AES members also have free access to a large number of past technical review articles such as this one and other tutorials from AES conventions and conferences.