@katjav said:

@sumidero said:

Could it be a choice/alternative for what you want to do?

I have not heard of this technique yet. Do you know if it is mathematically described somewhere? How many frequencies, are those sine shots windowed, how is the inverse calculated, etc?

As far as I can remember, there were no phase calculations involved in my calculi, just relative amplitudes for each frequency bin, in dB. I had set the loudspeakers outdoors, side by side and heading upwards to the sky, placed the mic a meter above them and got the 'reference curve' trusting the PA system response (a pair of close field reference monitors), then did the same inside the room. Response over difference in dB and hands on craftmanship to make bass traps. I know nowadays that I maybe should have done something better, with better equipment, I know it was sort of rough, but at that time I just wanted a room with a flatter response, asap.

Nowadays my job has gotten a little hectic, sorry for the delay in giving you feedback.
Best regards,

sumidero

@bassik said:

---

Sumidero wrote: "I saw, many years ago, a PC-XP based ISA card for acoustic measurement of loudspeakers that instead of using a sine sweep it sent short trains of frequency fixed sines, opening the record just when the wave train was passing through the microphone."

is this system you are referring to?

http://www.mlssa.com/

IF yes, MLS (Maximum Lenght Signal) measurement system have been the industry standard until the sweep sine method comes out (expecially the log swept sine).

According to my knowledge, It is a very good way of measuring IRs but it has some problems in handling the non-linearities of the reproduction and recording chain.

Yes, I think it was it. It was shown on an old CGA monitor but it looks very similar.

I think that one of the main drawbacks of that method might be the influence of the transients in each sine train. If you don't cut them out they would introduce artifacts in the spectra. At the time I did my home measurements, I checked these transients throughout the frequency spectrum and made an algorithm to slice them out before doing dB calculi. I wander where that Octave code has gone to...

I wish you all a good week.

sumidero

Made some good progress on the topic at last.

Attached patch has a log chirp generator for max 20 seconds, and computes the inverse. As a test, the chirp is convolved with it's own inverse, using [partconv~]. The result should approximate a single pulse of value 1 at time L = chirplength.

With stop frequency set at sr/2, that pulse is 0.997393 and surrounding values are in the order of 1/1000. With lower stop frequency, results quickly drift away from the ideal case. The imprecision artefacts as we perceived them, seem to do relatively little harm.

I think that the mathematical routines, though not perfect, are far more precise than physical test equipment (speakers, microphones) will ever be.

So it is now a matter of including record / load / save routines, to collect some real-life responses.

Katja

http://www.pdpatchrepo.info/hurleur/logsweep&inv-test.pd

Hello Katja,

that is great news!!!

I was looking into the scaling factor today but get stopped by work stuff and I am planning to have a look tonight.

I will also test the patch as well.

If you want I can do a quick real world test here at work as I have all the equipment.

In my first attachment there was a sort of implementation of reproduce and recording of the signal (if I remember well), would this be of help??

Bassik

Also forgot to mention that looking around for another reason I found this software made in Java:

http://www.hometheatershack.com/roomeq/

It does IR measurements as well.

Cheers

Bassik

Included some recording-, save- and load-routines to the log sweep deconvolution, and done a first real world test in my room. Dreadful standing waves oh boy.

Status quo patch attached.

Some weighted amplitude normalisation should be done on the sweep response. Now I have done an easy peak normalisation, but this will always lead to IR's having too low power.

Anyway, it is already possible to see the effect of the log sweep method (by the way this works with a linear sweep as well): non-linear distortion and pollution is shifted left of time zero and thus separated from the impulse response proper. Use vertical zoom to see this.

It would be useful to include an elementary wave editor for trimming. I have now done this with Audacity. Hardoff has done a wave editor in Pd, we could use a subset of it and include vertical zoom.

Katja

http://www.pdpatchrepo.info/hurleur/logsweepdeconv-test.pd.zip

@bassik said:

I found this software made in Java:

http://www.hometheatershack.com/roomeq/

It's great and it does all you need plus more.

Our Pd patch may not grow to such a professional tool, if only because it would require years of developement. But from an educational perspective a Pd implementation may be more interesting. I remember very well how IR measurement was demonstrated by our dsp teacher in Sonology class. We went through the mathematics, and I was excited about how you can expand and compress time dimension with help of a sweep.

With our Pd patch, the whole process can be visualised. At any point in it, you can test how (im)precise your method and result is. You can then decide whether it is good enough for your purpose, or should be improved. You can read Ben Saylor's [partconv~] code to see how fast convolution is done in practice. The benefits of open source. I see good reason to continue our log sweep deconv project.

Katja

Hello Katjav,

The new patch is great, it works well and already does more than I was expected to reach in such a small amount of time.

I would like to do some tests and comparison with a commercial software with the same functionality but I have just realised that the laptop with the software installed is not available until monday; nevermind.

I have two questions on the patch:

1. there is a strange behaviour in PD, if i turn on dsp the patch doesn't allow me to move the horizontal slides or better it doesn't visualize the movements while with dsp off there is no problem. It is a bit of a problem as you don't see the computation progress slider moving.
   Do you know why?
   No error messages and the last PD installed.

2. would you please expand a bit more about weighted amplitude normalisation?
   in the case of IRs normalising the main peak of the file will make all the IR information readable, or probably I am missing the point here.

so regarding the to-do list:

• Multichannel: it is quite important it could be multichannel input (i.e. ambisonic mic or stereo pairs) while it is much less important as multichannel output (1 signal would be common to all speakers you are testing, no need for different signal).

• fft analysis of IR: great idea but would be great to derive also additional acoustic parameters, I will provide maths for all the parameters we need to implement (ISO 3382 part 1 and part2, sorry I need to send the papers privately as I would get in trouble if I share on the web).

Also I would like to add in the future, a-format to b-format encoding (http://pcfarina.eng.unipr.it/Public/B-format/A2B-conversion/A2B.htm).
This is why 4 capsules tetrahedral mics can be used to analyse 3D soundscapes including directional analysis (along 3 axes) and understand issues in room such as room modes directions, flutter echoes and first reflections direction among others.

I am glad you have done a clean-up of the scaling constant calculation as in the previous patch it was a bit confusing to me.

I will have a more in depth play around tomorrow or friday...i am also in the middle of moving house...:(

Thank you

Bassik

@bassik said:

would you please expand a bit more about weighted amplitude normalisation? in the case of IRs normalising the main peak of the file will make all the IR information readable

The sweep itself has RMS 1/sqrt(2) as it is purely sinusoidal. Deconvolved, it translates as a unit pulse which, as a filter, would have neutral amplitude effect. A sweep response however, fluctuates with peaks and dips over the frequency range. When peak-normalised, it has only dips and it's RMS will be lower than 1/sqrt(2), assuming it still has sinusoidal character. This also translates to an IR with low power. When used as a filter, such an IR can reduce overall signal output substantially. In my first tests, by some 10 dB or more. See what I mean?

Therefore I think there should be an option to somehow normalise RMS of the sweep response / impulse response. I have no clue how this is handled in regular analysis software. Maybe in frequency domain, after the IR is trimmed to it's useful length?

Katja

@bassik said:

there is a strange behaviour in PD, if i turn on dsp the patch doesn't allow me to move the horizontal slides

This quirk has popped up a few times recently on Pd forum

http://puredata.hurleur.com/sujet-4893-number-slider-boxes-freeze

The deconvolution patch is in no way optimized. The [Scope~] showing sweeps is running at highest possible refresh rate. The deconvolution subpatch does crazy lots of fft's continuously. These things need switches to turn them off occasionally.

Katja

Hello Katja,

thank you very much for your answer.

Unfortunately in the next couple of weeks I would not be able to be present on the web due to work commitments and moving house with my wife (no internet at home...).

So I will try to check the forum and have a look at the amplitude normalisation but I am not sure I will have enough time to study it properly.

Sorry about that.

I will get in contact as soon as everything will be sorted.

CHeers

Bassik

Hello Katja,

Fortunately I had a conversation with a colleague today about IR normalisation for another reason and probably we got our answer.

In simulation softwares (Acoustics 3D models simulations), the auralisation module when deriving an IR and then the convolution of the anechoic file, applies a gain at the two stages and stores the info in the file.

In the case of the IR it applies a gain in order to have the maximum peak digital value (0.9999999999) and then consequently the same gain to all the rest of the IR (24-bit integer).

When doing the auralisation it applies another normalization gain to prevent clipping of the auralised file and takes into account that auralisation is made at 16-bit integer.

a simple program for the IR normalisation (converting the IR to PCM) can be:

Variable definitions

FP(N) = floating point impulse response

PCM(N) = linear impulse response

M = sample length

Algorithm

MAX = 0

For N = 1 to M

If abs (FP(N))> Max then MAX = abs(FP(N))

Next N

GAIN = (2^16)/MAX

For N = 1 to M

PCM(N) = FP(N) * GAIN

Next N

Store GAIN

As you can see the gain is arbitrary and depends on the IR (or the recorded sweep response) and there isn't a general rule as far as I know.

IN our case we should do these normalisation of the file and store somewhere (metadata in the wav file) the gain applied that would be useful in future applicaitons.

IN the case of multichannel IRs this can be a bit more complex but one thing at time is better.

Hope this helps

Bassik

Hello Katja,

one other comment:

gain formula should be:

GAIN = (2^15)-1/MAX for a 16-bit file.

or GAIN = (2^23)-1/MAX for a 24-bit file

CHeers

Bassik

@bassik said:

In the case of the IR it applies a gain in order to have the maximum peak digital value (0.9999999999) and then consequently the same gain to all the rest of the IR (24-bit integer).

Ah, so normalisation of the IR is done in such a way that the resolution of the integer format is maximally employed. That makes sense.

And then for convolution of the anechoic signal, a second normalisation factor should be applied, in such a way to prevent clipping. This is the difficult part. How to compute this factor from an arbitrary IR? I don't see how it could be computed from the IR array itself, since the convolution output magnitude depends on input frequency. But I can imagine a test method: convolve a sweep with the IR and find the maximum output, then scale down to have maximum output 0.999999 when expressed in floats. Is that how it is done?

All taken together, lots of routines to be written in Pd. Oh yeah, there is something else: for proper analysis, the impulse response of the test set (speaker&mic) should be factored out. Taking close-mic sweep response, invert magnitude spectrum, compensate, etc. Is this regularly done in IR measurement practice? If so, how?

Katja

@katjav said:

How to compute this factor from an arbitrary IR? I don't see how it could be computed from the IR array itself, since the convolution output magnitude depends on input frequency.

Oops, this is of course very simple to find out in frequency domain since the trimmed impulse response will easily fit in one FFT frame. This thread is now getting full with my stupid remarks, sorry.

Katja

I have now written an object class [chirp~], generating linear or logarithmic chirps calculated with 64 bit float precision. It has user parameters for start / stop frequency and chirp length. It has two outlets, one for the chirp and one for the inverse chirp.

The chirps from this object have neglegible numerical error, as opposed to the earlier chirp-generating patches. It is a pity that you can no longer see how the chirps are calculated, without reading the C-code. But this is the only road to quality IR measurement within Pd.

Attached is a zip containing chirp~.c, chirp~.pd_darwin, and chirp~-help.pd. If you happen to be on OSX, you can check [chirp~] rightaway. Otherwise you may compile the class for your platform, or wait till I have done that on someone else's computer.

Katja

http://www.pdpatchrepo.info/hurleur/chirp~files.zip

Hello Katjav,

I am finally back even if still with some problems as I don't have internet at home and work is manic....

Anyway let's put everything in the right order:

1. [chirp~]: I am not on mac but I can test it on mac at home but it would be good to compile it for all platforms...do you have any reference where I can understand how to do the compiling?
   also when you consider linear sweep, are you considering a different inverse filter which only the time reversal of the test signal?

2. In attachment there is an extract of the CATT acoustics manual on calibration of IR and convolution; It is related to audio convolution for auralisation simulations but the same principle can be applied to our purposes to calibrate IR and convoluted responses.

3. Katjav wrote: Oh yeah, there is something else: for proper analysis, the impulse response of the test set (speaker&mic) should be factored out. Taking close-mic sweep response, invert magnitude spectrum, compensate, etc. Is this regularly done in IR measurement practice? If so, how?

Normally in IR measurement practices with Swept sine, the convoluted result will appear in the software with all the non-linearities on its left hand side. In all common software (as far as I know) the removal of the non linearities is done by hand just by changing the start point of the result IR and by windowing the end part of it....after this you start analyzing all the parameters you want to derive from the IR.
I hope I answered your question.

4. Calibration of IR and resulted convoluted IR
   as far as I know there isn't anything in literature that explains exactly the routine for calibrations but the discussed process in the previous posts will do the job with the important task of storing the gain information in order to be able to calibrate the convoluted anechoic file later.

also please have a look at this paper: http://pcfarina.eng.unipr.it/Public/Papers/238-NordicSound2007.pdf

from page 13 to 24 it explains all the issues that can occurs with sweep sine method.

I will try to start the tests soon and to try to find more info in literature.

Hope this helps

Thank you

Bassik

http://www.pdpatchrepo.info/hurleur/CATT_acoustics_IR_calibration.pdf

Thanks Bassik for pointing to the Farina article. Particularly interesting is the section on Kirkeby inverse filters for compensating spectral deficiencies in measurement equipment.

Meanwhile, I have made a patch to inspect characteristics of [chirp~] output in detail, with horizontal and vertical zoom options on the IR plot, and spectrum magnitude and phase plots. Real-signal chirps can never have a completely white spectrum. It turns out that logarithmic chirps are quite sensitive to parameter settings, while linear chirps are less so. Therefore, a log chirp may have considerable ripple near start and stop frequency, but I have also found settings which induce low ripple (+/- 0.3 or 0.4 db). It may have been coincidence that I stumbled upon such favourable settings. There is probably no simple rule to derive them numerically.

Because the log chirp is so sensitive to parameter settings, it needs special attention. For an IR measurement tool, we could provide 'approved' chirps in stereo wave files (32 bit floating point) together with their inverse. Chirp generator / tester and IR measurement tool would then be two different Pd root patches which can be developed independantly.

So, the chirp generator / tester is attached to this post. Again, there is only an OSX binary for [chirp~] (same one as before). Sorry. Building on other platforms is easy pie if you have developer tools installed and know how to operate them, but otherwise, don't spoil your time on it. I'll do it later on a friend's computer.

Katja

http://www.pdpatchrepo.info/hurleur/chirp~test.zip

Hello Katjav,

hope you are well.

I have performed some test on a real case and I have to say that despite the first 2 measurements worked very well, after that the deconvolution process stopped to work properly.

Attached there is one of the wrong results I have obtained from the soft.

you can clearly hear that instead of the IR you get the inverse filter at the same time where the IR should be.

Is it a user problem?
Is it a multi measurement issue with the software?

I will investigate further before any conclusion.

thanks

Bassik, there may be some confusion. This last patch chirp~test.pd is only a patch to test the qualities of the generated chirp itself.

I am kind of swallowed by details of the method. Instead of quickly producing a working IR measurement tool, I am now trying to refine the method, starting with refining the chirp itself. Last time I mentioned the sensitivity for parameter settings. Now I have found a systematic method to optimize a chirp. The formula is different from the regular ones as found in texts. This has lead to yet another chirp object [expochirp~] (yes it's exponential, not logarithmic, also see Farina's 2007 article).

The regular exponential chirp formula's make a chirp start at sine phase to avoid a step, but they do not care about the phase at the end of the function, therefore it may well end with a step and produce excessive ripple at the high spectrum end. This is traditionally relieved by windowing, or by manually cutting the chirp at the last zero-crossing as Farina suggests. My new formula computes the chirp in such a way that it will start and stop in sine phase. No need for windowing or cutting, and the ripples are minimized.

Still there will always remain ripples in a chirp's spectrum, and I am now trying to produce inversion-filters to clean up a system's measured IR. For the high spectrum end, this is fairly well possible. But in the low spectrum end, where it is more important, the ripple character of exponential chirps is such that conventional methods fail. I did find some solution but it's not without compromise. Therefore I am investigating alternative methods.

I am sorry that proceedings are slow now. But my hope is to contribute to a good quality tool in the end. I'll post details on the new chirp formula soon.

Katja