I read http://designingsound.org/2013/06/tutorial-a-compressor-in-pure-data/ about compressor working and I tried 3 different (or similar) compressors: the one from the page, the one from DIY2 by Hardoff and the one from rjdj by Jwif. I thought that compressors works different...

I use visual arrays to see the difference from the original sound (that is very loud) and the compressed one. With the first compressor, there was NO DIFFERENCE. No at all... I don't know if I was using wrong, but it was the same graphic. With DIY2/st-compressor.pd, there's difference and it seems to be good... untill I make a kick. Again, the waves goes to the sky (well, not so much). The third compressor can compress a lot (low threshold), but all the rest of the sound does too.

I though that only the sounds that are higher the threshold are compressed and the rest stay the same. Am I wrong? Am I confused? I thought that a compressor is like tanh(), but with editable controls.

Can someone explain this to me?

El 30/09/13 17:23, Charles Z Henry escribió:

Then, you haven't picked the right computer for your composition :)

If I could, I would buy another notebook... and throw away this with-non-working-in-Linux-hybrid-graphic-cards fucking HP notebook. But, our economy is fatal. One dollar is ten argentine pesos. I bought this computer 3-4 years ago, at ARS $6.000. Think spending U$S 6.000 in a computer. Now, a good computer, is above $10.000.

El 30/09/13 17:43, mr sgg escribió:

ask 4 persons and you probably will get 4 different answers. first thing i would notice the setup seems dirty. for example use an external preamp instead of digital amplification (e.g. *~4) if you want proper sound quality. secondly i think you should try out and trust your ears. it really depends on what you want it to sound. tanh most probably will lead to more distortion while compression will eat more cpu.

I will use a wireless system with this mic, so, it has gain. I'll check that, because I think that it is important a preamp there.

to avoid your kicks being louder than vocals you should refer to beatboxers. maybe you do something wrong there. or you should use 2 mics. i admit i am not familiar with mixing rappers who beatbox at the same time

The mic is very good and it sound very good. I want to clarify my doubts, because I "see" waves beyond 1 in visual arrays... and I want to understand that.

El 30/09/13 17:23, Charles Z Henry escribió:

On Mon, Sep 30, 2013 at 2:35 PM, Mario Mey <mariomey@gmail.com mailto:mariomey@gmail.com> wrote:

Now, I am checking volumes of my looper patch. I had to raise [*~
4] the volume of the mic, to get a razonable volume, compared to a
song file, for example. But, using this looper patch, I make
beatbox. So, kicks and snares from my mouth get in the mic. And,
using a visual array to test it, I realize that the kicks and
snares are so much higher volume than the vocals.

The patch has FXs with feedbacks, so, they can make signal > 1.
So, at the end of the patch, there's [expr tanh($v1)] to to avoid
that...  tanh() is simpler than a a compressor, but it loose some
sounds (I think). Or I should trust in tanh()?

Multiple choice:
1- Use tanh() in the input, after adc~ and before dac~.

This will cause distortion and change the shapes of your waveforms, and introduce extra harmonics. It may be an interesting effect, but it will change the sound of your beatboxing.

2- Use a compressor patch in the input, after adc~ (and tanh()
before dac~)

This would be the preferred method.

3- Use a compressor at the end of the patch, before dac~

If you have multiple instruments or voices in the output, you'll be decreasing the volume globally and throwing off your mix.

4- Stay as it is now...

Also, I can't spend more CPU process...

Then, you haven't picked the right computer for your composition :)

Really, I don't think compression should be a cpu-expensive process.
Plus, you have only one mic, right?

What do you recommend me to use?

Thanks.

*AND THE WINNER IS...*

El 01/10/13 17:28, katja escribió:

Did you try [zexy/limiter~]? It's not so CPU-intensive. Isn't that what you want: only attenuate the sounds which exceed a limit, and leave the rest untouched?

*This is the one.* After trying with its LIMIT, CRACK and COMPRESS options, LIMIT just does the trick. I don't know if it is the "perfect" solution, but I can "see" a good result saving to 32bits file and open it in Audacious, as you see in the attached image. And it spends 1% of CPU.

El 30/09/13 23:58, i go bananas escribió:

if the kick is not compressing properly, then one idea might be to lowpass filter the signal that goes to the compressor's detection stage. that way, you'd filter out a lot of the energy from the hats, snares, etc.

I put [0-[lop~] just before [pd amp-factor] and try all values... but the peaks are still there. I don't understand why the first peak is always there... Can't be detected at the right time to compress it, because of the window?

El 01/10/13 11:38, Antoine Villeret escribió:

if CPU is a matter, you could try to use an external analog (or digital) compressor (and an eq) between your mic and your soudcard.

Well... I'm looking for a cheap way to do it. As I told Charles, my country economy is not favorable. The idea is not to buy a compressor device... or an EQ... or a new computer... or a new microphone... or...

The table display in Pd is linear while sound intensity feeling is logarithmic, so it's not surprising to have such peaks.

Well, that explains why is so different between the vocal and the kicks. But it is still a bit confusing to me to understand logarithmic against linear. Also, I would like to "see" a logarithmic table display. Maybe a linear to log signal conversion only for represent sound? I should be something already done...

When you try to write value > 1 to disk or to output it, it's clipped to 1.

I discovered "-bytes 4" (32 bits) flag on [soundfiler]. So, it saves almost all the information.

Thanks everyone!

El 01/10/13 00:54, Brian Fay escribió:

Are you using a pop filter or windscreen or anything in front of the mic? Maybe that could help filter out some of the super-loud plosive and fricative sounds.

No beatboxer uses a pop filter or windscreen... unless he/she is at a recording studio.

Moving further away from the mic would definitely lower the volume, too (the SM58 is a cardioid mic, so it has proximity effect, where bass gets emphasized the closer you are to the mic). I guess the proximity effect will actually sound pretty nice on the kicks, but it also would explain why the peaks are so high.

Yes, the proximity is very important. Unfortunatly, I can't move further away the mic.

Well, I wrote in DIY2 thread in Forum, because I want to use Hardoff's compressor and I wanted his opinion, too. Because, after several tests, I realized that I can't get rid of the first peak of a kick (more or less, signal = 4). So, I thought about using the compressor with attack and release at lowest and, inmediatly after that, something like [expr~ (tanh($v1/1.5))*1.5]. So, tanh() "compress" the highest peaks to 1.5, but it doesn't distort so much the sound.

Can anybody tell me if I am very wrong? Is the math correct?

As when writting the arrays to disk, Pd has to normalized them to 1, I can't see really the difference in a sound editor, between the original and the compressed one.

On Mon, Sep 30, 2013 at 3:35 PM, Mario Mey <mariomey@gmail.com mailto:mariomey@gmail.com> wrote:

Now, I am checking volumes of my looper patch. I had to raise [*~
4] the volume of the mic, to get a razonable volume, compared to a
song file, for example. But, using this looper patch, I make
beatbox. So, kicks and snares from my mouth get in the mic. And,
using a visual array to test it, I realize that the kicks and
snares are so much higher volume than the vocals.

The patch has FXs with feedbacks, so, they can make signal > 1.
So, at the end of the patch, there's [expr tanh($v1)] to to avoid
that...  tanh() is simpler than a a compressor, but it loose some
sounds (I think). Or I should trust in tanh()?

Multiple choice:
1- Use tanh() in the input, after adc~ and before dac~.
2- Use a compressor patch in the input, after adc~ (and tanh()
before dac~)
3- Use a compressor at the end of the patch, before dac~
4- Stay as it is now...

Also, I can't spend more CPU process...

What do you recommend me to use?

Thanks.


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hey,

if CPU is a matter, you could try to use an external analog (or digital) compressor (and an eq) between your mic and your soudcard.

The table display in Pd is linear while sound intensity feeling is logarithmic, so it's not surprising to have such peaks. When you try to write value > 1 to disk or to output it, it's clipped to 1.

++ a

-- do it yourself http://antoine.villeret.free.fr

2013/10/1 Mario Mey mariomey@gmail.com

How does Pd or the soundcard outputs the signal > 1? Does it compress it?

I started this thread because I saw the high peaks in an array... but I never listened the sound distorted.

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On 2013-10-01 16:30, Mario Mey wrote:

How does Pd or the soundcard outputs the signal > 1? Does it compress it?

depends on the actual backend and how you are using it. e.g. some backends only support values that cannot possibly exceed +-1; in these cases, the signal will be hard clipped! with some other backends (that support floating point values natively) you might get away with a compressed/limited signal.

if you don't know, it's best to assume that anything exceeding +-1 will be clipped. even if you do know, it's better to assume that they will be clipped.

I started this thread because I saw the high peaks in an array... but I never listened the sound distorted.

with very transient sounds (like clicks and pops) you might not hear the distortion, as these values are very broadband anyhow.

btw, make sure that your signal is not already distorted on the soundcard input. you cannot undo that...

fgamsdr IOhannes