Re: [music-dsp] Reverb removal

2011-07-29 Thread Massimiliano Tonelli

Hi there

Blind dereverberation is a very difficult task. I have spent the last 
years of my research working on this topic.


Even though several approaches have been proposed, a possible 
discrimination into two classes can be accomplished by considering 
whether or not the inverse IR needs to be estimated. In fact, all 
de-reverberation algorithms attempt to obtain dereverberation by 
attenuating the IR effects or by undoing it. In a simplistic view, one 
approach tries to alleviate the “symptoms” of the signal degradation, 
while the other attempts to address its “cause”. Due to the spatial 
diversity and temporal instability that characterize the IRs, the first 
class of algorithms can offer, at the current state, more effective 
results in practical conditions. However, the algorithms belonging to 
the second class can potentially lead to ideal performances. It must be 
noted that practical de-reverberation is still largely an unsolved problem.


“Reverberation suppression methods” are based on diverse set of 
techniques such as: beamforming, spectral subtraction, temporal 
envelope, LPC enhancement .


“Blind reverberation cancellation methods” can be distinguished into two 
sub-classes: the techniques that are based on the IR blind estimation 
followed by its inversion and the ones that attempt to directly estimate 
the inverse system. While the first methods have the benefit of 
providing the access to the IR estimation, and this is of interest for 
the extraction of many acoustic parameters, the calculation of the 
inverse system is not trivial even in the non blind case and it might 
lead to inaccuracies. Therefore, it is probably more consistent for the 
de-reverberation purpose to achieve a direct estimation of the inverse 
system.


It is worthy to notice that blind reverberation cancellation and 
suppression methods can be combined to offer hybrid strategies.


Even if cesptrum based techniques are popular in speech recognition, 
they generally perform poorly since, speech and the acoustics cannot be 
clearly separated in the cepstral domain. Furthermore, even if this 
separation could be achieved, estimation of the clean speech signal is 
still problematic since all phase information are removed. Therefore, 
cepstral based methods can only be successfully applied in simple 
reverberation scenarios (i.e. signal degraded by simple echoes).


you can find further information on:

www.dereverberation.org


Hope this is of help
Massimiliano Tonelli

Il 28/07/2011 11:24, Jerry ha scritto:

Alexandros, I haven't looked at your dissertation but it sounds very cool. Is 
cepstral processing appropriate for removing reverberation? Just wondering.

Jerry


On Jul 21, 2011, at 10:08 PM, Alexandros Tsilfidis wrote:


Dear Kenneth,

Dereverberation is still an open and largely complicated research issue. 
Depending on the application context (and on your needs) there are many methods 
that you can chose from. However,  from the mathematical point of view blind 
dereverberation in realistic scenarios is a hard (or impossible) to solve blind 
deconvolution problem and most methods make assumptions that produce 
significant processing artifacts. You can download my PhD thesis on 
dereverberation here:

http://www.wcl.ece.upatras.gr/audiogroup/alexandros/publications/Tsilfidis_PhD.pdf

where you can find a short literature review in p. 31-37.

Moreover, if interested you can download some of my personal publications and 
listen to the corresponding audio demos at my personal website:

http://www.wcl.ece.upatras.gr/audiogroup/alexandros/

If you provide me with some more details on the specific application (e.g. how 
important is the perceptual quality of the final results? do you have any 
(other) prior knowledge than the recorded sound file? is the computational 
complexity an issue for you?) probably I would be able to point you out the 
specific method(s) that (may) suit your needs.

I hope that was helpful!

Alexandros Tsilfidis, MPhil, PhD
Post Doc Researcher
Department of Electrical and Computer Engineering
University of Patras
Greece


On 22 Jul 2011, at 06:50, Kenneth Ciszewski wrote:


Any suggestions about the best way to remove reverb from a sound recording 
(voice/speech)?
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Re: [music-dsp] Factorization of filter kernels

2011-01-25 Thread Massimiliano Tonelli

Hi,

Maybe this can be of help.

http://www.musicdsp.org/archive.php?classid=3#65

All the best
Massimiliano

Il 19/01/2011 15.56, Uli Brueggemann ha scritto:

Hi,

thanks for the answer so far.
A polyphase filter is a nice idea but it does not answer the problem.
The signal has to be demultiplexed (decimated), the different streams
have to be filtered, the results must be added to get the final output
signal.

My question has a different target.
Imagine you have two system (e.g. some convolution  boards with DSP).
Each system can just run a 512 tap filter. Now I like to connect the
two systems in series to mimic a desired 1024 tap filter. The 1024
kernel is known and shall be generated by the two 512 tap filters.
So what's a best way to decompose the known kernel into two parts ? Is
there any method described somewhere?

Uli


2011/1/19 João Felipe Santosjoao@gmail.com:

Hello,

a technique that allows something similar to what you are suggesting
is to use polyphase filters. The difference is that you will not
process contiguous vectors, but (for a 2-phase decomposition example)
process the even samples with one stage of the filter and the odd
samples with another stage. It is generally used for multirate filter
design, but it makes sense to use this kind of decomposition if you
can process the stages in parallel... or at least it is what I think
makes sense.

You can search for references to this technique here [1] and here [2].
A full section on how to perform the decomposition is presented on
Digital Signal Processing: a Computer-based approach by Sanjit K.
Mitra.

[1] 
http://www.ws.binghamton.edu/fowler/fowler%20personal%20page/EE521_files/IV-05%20Polyphase%20FIlters%20Revised.pdf
[2] https://ccrma.stanford.edu/~jos/sasp/Multirate_Filter_Banks.html

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João Felipe Santos



On Tue, Jan 18, 2011 at 5:46 AM, Uli Brueggemann
uli.brueggem...@gmail.com  wrote:

Hi,

a convolution of two vectors with length size n and m gives a result
of length n+m-1.
So e.g. two vectors of length 512 with result in a vector of length 1023.

Now let's assume we have a vector (or signal or filter kernel) of size
1024, the last taps is 0.
How to decompose it to two vectors of half length? The smaller vectors
can be of any arbitrary contents but their convolution must result
must be equal to the original vector.

It would be even interesting to factorize  given kernel into n
smaller kernels. Again the smaller kernels may have any arbitrary but
senseful contents, they can be identical but this is not a must.

Is there a good method to carry out the kernel decomposition? (e.g.
like calculating n identical factors x of a number y by x =
Exp(Log(y)/n) with x^n = x*x*...*x = y)

Uli
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