• whale-av

    @JackPD Here is how to make Pd and Python talk on a RPI..... https://guitarextended.wordpress.com/2012/11/03/make-python-and-pure-data-communicate-on-the-raspberry-pi/
    And you will find more info in the same blog as the author has built a similar project.
    David.

    posted in libpd / webpd read more
  • whale-av

    @ingox Excellent! ...... I really don't know why I was thinking in the audio domain.
    Here it is with an auto complete..... once the threshold is set... trim_silence.pd
    (threshold must be greater than 0...... a little tiny bit!)
    David.

    posted in technical issues read more
  • whale-av

    @phil123456 It depends whether there is any background noise (and any silence in the middle of the sample). You could set a threshold for data received from [env~] and read the array. Count the number of audio blocks that return zero (or a value below the threshold). Stop counting and send that value, multiplied by the block size, to a -skip message for reading the array next time. Continue reading and stop counting blocks when you get another zero. Use the new value, the total of all blocks to this point including the previous "zeros" (multiplied by the block size again) to set a -resize message for the array.
    If you don't build this (or find a better way to do it) then I will do it myself...... as it will for sure be useful at some point.
    Call it "auto_trim_silence" maybe so it can be found.
    Maybe easier to write to a new array using the first positive bit of data to start the write, and stop writing when the first zeros arrive..... but you might miss the first samples? You still need [env~] as you cannot rely on only one sample.
    David.

    posted in technical issues read more
  • whale-av

    @H.H.-Alejandro Hello Ale......
    If you redraw the timbre curves for the higher frequencies (something) like this then you will have a brighter timbre all the time regardless of the note/timbre curve "Z" value......
    David.

    Capture.JPG

    posted in technical issues read more
  • whale-av

    @svanya All credit to those that put in the work.... unless it was me :neutral_face: ABSTR VANILLA.zip
    Abstractions only. I also dump .DLL's from various libraries in there for everyday work...... but they will do you no good for Mobmuplat.
    Also, many zexy replacements depend on [list tosymbol] and [list fromsymbol]. Are they implemented in Mobmuplat?
    David.

    posted in technical issues read more
  • whale-av

    @svanya You will be most likely to find what you want in the PMPD library..... in the examples folder.
    David.

    posted in technical issues read more
  • whale-av

    @H.H.-Alejandro Hello Ale.
    Not finished....... just for testing..... Ale4.zip

    All effects and audio meters etc. removed...... and only the first 2560 sines have micro adjustment.
    In 1_vod_12_2560 it takes curves 3 to 18.
    When running iannix I seem to get some distortion...... but if I stop sending the curves from iannix it is better.
    This suggests that the messages are interfering with the audio..... but I am not sure...... in fact I think not....... something else.... maybe high frequency beating.
    When you stop iannix....... or it gets to the end of the curves.... you can hear the beating at different rates of the sine waves....... it is interesting..... but starting the curves again makes it less audible.

    Maybe massive sines controlled by curves are less interesting than massive sines on their own?
    I will build one with only one curve for every sine..... to hear what Pieter suurmond expected..... so with a fixed note.

    vod_13_20480 and vod_14_50960 are there for you to try..... (20480 sines and 50960 sines).
    They take a while to load...... and for me they just distort..... but they really are a massive "ask" of the computer.
    As I said above they are not finished. A lot of work needs to be done for all the curves to be used....... and as I think they will not work I have not done the work yet.

    I really like 1_vod_12_2560 when iannix stops...... it is endlessly changing and interesting......
    David.

    posted in technical issues read more
  • whale-av

    @Ch4n Sorry, I have these......... round.zip which suggest that you might find what you need googling..... "libpd iemlib github"
    David.

    posted in technical issues read more
  • whale-av

    @Ch4n In the menu...... help..... find externals....... show all......
    For any you don't find... download them and put their folder in the Pd/extra folder.
    Follow any instructions you find in any "readme".
    If necessary declare the name in ...... edit... preferences..... startup..... using the -lib flag...... (i.e. -lib zexy)
    Normally you do not need to put the path to the folder if it is in "extra".
    David.

    posted in technical issues read more
  • whale-av

    @atcq1227 In extended there was [streamout~].... (ggee library)
    But un-buffered.
    David.

    Capture.JPG

    posted in technical issues read more
  • whale-av

    @SystemaComplicado A complete guess..... inlet.zip that its in there somewhere.. (no guarantees..... its way beyond me).
    David.

    posted in technical issues read more
  • whale-av

    @Jocketor I might have grabbed the stick by the wrong end...... but this patch will output instant data as the audio arrives............ decile.zip
    Open [mother_live].
    Do any math you want inside [test_decile], click "continuous" and connect the output to your oscsend.
    It should output the same as [tab_sum] for the window size.
    Window side has to be 4096 samples though unfortunately..... to be as close as possible to 1/10 of a second at 44.1K. Exactly 1/10 is not possible like this......

    It can be adapted to examine a large file already written by [soundfiler] or [tabwrite]......... by getting the total arraysize and scanning chunks of samples in order. Then it would not need to run at audio rate and the chunks could be 4410 samples (or 4800)....... but that should be the same as [tab_sum] and such.

    The differences you are getting might be because of the different window sizes in your patch between audio and data analysis...... because you have selected a 1/10 second chunk size.

    If you want to post your patch then someone might spot why you get different data depending on your approach, and it will be easier to try to help by modification.
    What I have posted above contains no FFT analysis...... so it is not yet what you are looking for.... but it might be a stepping stone.
    David.

    posted in technical issues read more
  • whale-av

    @Jocketor I mean that it can be done very fast as an analysis once you have all the data in an array..... so if you load an audio file into an array... or you have written a live input to an array..... then the analysis will be much quicker than doing it "live" in the audio domain where you are waiting all the time for audio blocks to complete their processing.
    Do you want to show the user the results "live" while they are recording the audio..... is that why you want to use audio objects for analysis?
    If you want amplitude values during a "live" [adc~] recording you can use [env~] to output those values (although in dB) at the end of each audio block (or set a wider..... or a narrower window if you wish for [env~]...... right down to a single sample if you set the block size to 1).
    David.

    posted in technical issues read more
  • whale-av

    @cixxx You need the hcs library. Does help/externals not find it?
    But the required patches are all abstractions........ square.zip
    It uses a patch from the purepd library..... which I have included.
    You will need to change [purepd/float_argument] to [float_argument] in the objects within [square~]
    (drill down to find them both)
    David.

    posted in technical issues read more
  • whale-av

    @H.H.-Alejandro It seems that your computer is quite powerful....... that is good news.
    [voice_gen128microtonal] contains 128 sine wave oscillators...... so you have 3840 running if you have 30 voicegens......... impressive.
    You need 30 curves ....... 3 to 32..... to drive that.
    David.

    posted in technical issues read more
  • whale-av

    @weightless Ah.... I thought you needed to know if they were connected "outside" the patch...... not inside.... i.e if a signal cord is connected to the patch.....
    David.

    posted in technical issues read more
  • whale-av

    @LiamG @weightless As I understand [canvasconnections] you would need to use it for the container patch interrogating inlet "one" (if its the second inlet) for your object [grainy-whotsit]........
    So from within [grainy-whotsit] use [canvasconnections 1]....... maybe, to look "up-one-level"?
    David.

    posted in technical issues read more
  • whale-av

    @ddisciglio Use [select 0]
    Any zeros will drop out the left outlet...... and any other float values will drop from the right outlet....
    David.

    posted in technical issues read more
  • whale-av

    @weightless [env~] is already the average of a window (and can match the block if you wish). You can change the windowing of [env~] to make sure....... see it's help file.
    Or if you wish you could "lock" the switch once it has seen an input...... if that is appropriate to this use?
    That depends whether you are connecting and disconnecting all the time......
    David.

    posted in technical issues read more
  • whale-av

    @weightless Maybe....... but of course audio input of zero will look like no input connected.
    It should work because an audio inlet sends a stream of zeros when not connected (if my memory serves me well), and with a window of 1024 it should not pick up on zero crossings.
    Connect the toggle to a [spigot] or two maybe to get the desired result?
    Capture.JPG

    posted in technical issues read more
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