@anna Welcome to the forum...!
Like this......... sync.zip
The Pd Extended version works well..... the Pd Vanilla version I cannot test as I don't have Gem working on my machine. But the only change I have made for Vanilla is to use the right outlet of [soundfiler] to get the sample rate of the audio file. In Pd Extended we used [soundfile_info]
If you need an explanation please ask.
David.

@jakobschauer Here is m-pd.h from pd 0.54.1 .... m_pd.h .... I haven't compared it to older versions so if you are trying to build for another version of Pd please ask.

Here are all the files from the source folder of that version...... Pd source 0.54.1.zip
I hope that helps but I have no idea how to compile correctly. They are from a windows version but I think .h files are the same.
If not then download Pd without installing it and go to the /src folder to find them.
David.

@porres In windows Pd can address 4 devices as you say.
Using ASIO4All more soundcards can be addressed and specific inputs and outputs removed from the final list that is passed to Pd as a single ASIO4All device..
David.

@gcerberv This should be useful....... it used Pd Extended (it was made in 2012) so there are some libraries you need to add to Pd vanilla )listed in the help file).
ulltrasonic sensors Pd.zip
It was posted here......... http://codelab.fr/3842 ...... and is all in French.
But, designed specifically to deal with ultrasonic sensors, smoothing and scaling data, it should be useful to you.
David.

@starpea223 I don't know of course what you are trying to achieve, but here is the full documentation of the Pd file format..... https://puredata.info/docs/developer/PdFileFormat
I hope that helps, but the best way to understand it is to open an existing patch in a simple text editor and read that alongside the documentation.
Not easy though.... using the GUI is much simpler.

I see from your other thread that you had a problem with the Gem window. Have you fixed that?
David.

@EEight The REW wizard is great software..... I would recommend that especially as it is free and powerful with great forum support
Phase inversion of treble frequencies can make you feel dizzy as you move your head...... we are much more sensitive to phase than other audio information.
Just one treble speaker in a stack is instantly horrible...... not so noticeable if the speakers are further apart though.
If you had more than one sub you would need much more tuning...... moving subs around until it sounds good everywhere is an everyday challenge on tour....... a few inches..... or less... can make a big difference.
David.

@EEight My mixer was for in-ear live on stage, the musicians using TouchOSC to control their earphones. So everything was mono. You are welcome to try it of course, and change it as you need. But although there was input and output eq there were no effects. It might be useful though for you to have Wi-Fi control while you are setting up.

I think you will find what you need (mostly) in Automatism..... all the eq and effect modules..... but I haven't seen an automatic room flattener. It is easy enough by ear though using a sweep generator.... sweep-explained.zip or easier still by humming, singing, whistling and "tssss'ing" with a microphone. You will hear the room resonance sing out at the frequencies you need to cut, and it's better than using pink noise as a source. You get the inaccuracies of the speakers at the same time.

You will have to spend some time setting up though. The real problem eq'ing a room is the comb filtering at lower frequencies.... much worse than the room or the speaker response because you can hear a note loud and then take one step and you can't hear it at all. The more speakers you have the worse it will be, and speaker placement is the most important factor. Really you need a room with very little reflective surface and the public should be listening to only one speaker wherever they are. It's ok'ish in a home cinema because you sit in one place and delays can be set to adjust the relative timing between speakers, but you will have noticed that public cinemas have soft furniture and each sound comes pretty much from one speaker as delays don't work for an audience sitting everywhere in the room.

It's probably worth at least slowing down the sweep, or having manual control, as it takes time for the room to get humming sometimes.
The vocal method works really well..... like when you are humming in the shower..... you really hear those room resonances coming back strong....
David.

@jakobschauer Maybe you should PDINCLUDEDIR=../pure-data/src ..... without the forward slash at the end, as it is looking for ../pure-data/src//m_pd.h ..... which presumably doesn't work with the double slash.

I am pretty certain that m_pd.h is in that directory.
David.

@polle Have a look at the doc folder in your Pd folder.
doc/3/audio.examples/G05.execution.order.pd ...... has information about how to use delays.
Also there was a very recent thread about the feedback problem....... https://forum.pdpatchrepo.info/topic/14471/dsp-loop-with-buffer-abstraction

And more...... because it can get complicated if using interpolation..... and the [send~] [receive~] 64 sample fixed block limit is gone in the latest Pd........ https://forum.pdpatchrepo.info/topic/14549/feedback
David.

@brendanmac If I remember correctly from school the power of a signal is the area under the curve.... which is why a compressed signal (then normalized) sounds louder and compression is often used when mastering a recording.
A square wave has a large area under the curve..... it is always at +- 1 ....... and will be 6db louder than a sine wave for the same peak values.
A saw contains all the higher harmonics (like the square wave) but the perceived loudness has more to do with the richness of the timbre than the actual power I think. The square wave of course also has that perceived increase in power from the harmonics.
A saw wave actually has less power than a sine wave, so the effect is purely psychoacoustic.

The harshness comes from the higher harmonics (especially the highest), so you cannot change that. A low pass filter will progressively return the waveform to a sine as the cut off frequency is reduced, because the higher harmonics are being reduced...... and it will introduce a phase shift (a slight lag in the waveform).
A wavetable will produce the same effects as your maths...... it is in the nature of the waveform and there is nothing wrong with your method.
You can use a [lop~] to reduce the effect though if you actually want a less harsh sound.
David.

@Ossicone You can use any receive name you wish, but it is a good idea to add a $0 to the name as in @FFW's example above. $0 is translated to a number as the patch is opened. If you then open the patch again...... (because you are using it multiple times.. maybe as an abstraction) the number assigned to $0 will be different, so the receive names will be different and there will be no crosstalk.

So something like $0-button1.
Here is the help file for sending messages to a canvas or a GUI....... help.zip
Open [pd Changing_GUI_Properties] from within that file to see the messages.
I have added an example [allatonce] in that zip.
David.

@evankelo Welcome to the forum...!
You could use a [delay] to close and then open a [spigot] and so stop the stream of values for the time of the [delay].
David.

If you want the notes to be sent faster as you move the mouse faster you could [count] the numbers arriving and for example output every 10th one.
But the result will be slightly "musical" changing more when you change direction (up > down or down > up) than when you are only going in one direction.
Here is the [count] object..... count.zip

Thank you all for your contributions. Yes @oid I suppose I do have moderating powers. I only generally use them to ban users that are using the forum for inappropriate advertising (unrelated to Pd, music, etc.). I have deleted inappropriate (in my opinion) posts in this thread, and locked it, trying to preserve up votes that were merited and useful information that members took the time to publish.
@Eeight is the Forum owner and so he is the final arbiter. He and I can still read those deleted posts.
David.

@Aliam.Sigsaly @porres I suggest you disengage, at least on this forum.
David.

@mbbaker I don't know why, but I know that you are not the first to bang your head against this problem (I think it probably does have to do with the [block 64] communication between the patch and the sub-patch).
Have a look at these threads, which looked at solving the problem in a different way.......
Hann window.... https://forum.pdpatchrepo.info/topic/3649/how-to-zero-pad-an-fft-with-block
Multiply by zero..... https://forum.pdpatchrepo.info/topic/12342/zero-padding-fft-to-increase-frequency-resolution/3
David.

@ionicle I agree with oid....... windows wish is not great.
To make up for my bad post above....... here is your patch using abstractions...... seqo.zip
It automatically saves your patterns, and you can easily add more drum sounds without the screen becoming cluttered.
You might find it hard to understand how the abstractions work. There is some help for that here...... https://forum.pdpatchrepo.info/topic/9774/pure-data-noob and then especially further down that thread... here....https://forum.pdpatchrepo.info/topic/9774/pure-data-noob/4
Once you understand the use of abstractions you will be able to build patches faster and more easily..!
David.

@oid My mistake....... opened in extended by double-clicking..... so no tempo message for [metro].
I thought it was trying to run rather fast........
David.

@ionicle Welcome to the forum...!
Stopping the [metro 10] that is banging [snapshot~] seems to relax Pd.
I was having trouble even editing your patch on my I7 laptop, even with DSP turned off.
I modified this part so that it does not run unless you start the metro...

Here is your patch with the change made.......
sequ.zip

I also changed the counter...... as at some point the number going into [%16] would become very large.
You need to keep the [count] object in the same folder as your sequencer. It is what we call an abstraction.... you can use it many times..... like a sub-routine.
I hope that helps get you going again with your patch.
David..

@brendanmac You need (in vanilla) to use [oscformat] to send a properly formatted osc message through [netsend], and [oscparse] to "decode" a received osc message from [netreceive].
A float as part of the message address can cause problems though, as it should contain only symbols
Here is a little help...... vanilla osc.zip
Or you can use the MrPeach library, which is a little easier.
David.

@ddw_music [pix_colormatrix] will pass just one colour or a combination set by a 4x4 matrix, and then [pix_grey] will convert the output to greyscale, but you will need to use it 4 times (one for each colour + alpha).
David.