I found this solution and it worked

@sdaau_ml said:

Sorry to necro this thread, but I finally found out how to run PureData under Pulseaudio (which otherwise results with "ALSA output error (snd_pcm_open): Device or resource busy").

First of all, run:

pd -alsa -listdev

PD will start, and in the message window you'll see:

audio input devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
audio output devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
API number 1

no midi input devices found
no midi output devices found

... or something similar.

Now, let's add the pulse ALSA device, and run -listdev again:

pd -alsa -alsaadd pulse -listdev

The output is now:

audio input devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
3. pulse
audio output devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
3. pulse
API number 1

no midi input devices found
no midi output devices found

Notice, how from the original two ALSA devices, now we got three - where the third one is pulse!

Now, the only thing we want to do, is that at startup (so, via the command line), we set pd to run in ALSA mode, we add the pulse ALSA device, and then we choose the third (3) device (which is to say, pulse) as the audio output device - and the command line argument for that in pd is -audiooutdev:

pd -alsa -alsaadd pulse -audiooutdev 3 ~/Desktop/mypatch.pd

Yup, now when you enable DSP, the patch mypatch.pd should play through Pulseaudio, which means it will play (and mix) with other applications that may be playing sound at the time! You can confirm that the correct output device has been selected from the command line, if you open Media/Audio Settings... once pd starts:

As the screenshot shows, now "Output device 1" is set to "pulse", which is what we needed.

Hope this helps someone!

EDIT: I had also done changes to /etc/pulse/default.pa as per https://wiki.archlinux.org/index.php/PulseAudio#ALSA.2Fdmix_without_grabbing_hardware_device beforehand, not sure whether that makes a difference or not (in any case, trying to add dmix as a PD device and playing through it, doesn't work on my Ubuntu 14.04)

@johnwbyrd I have never looked at the source code, but I run a live performance with Pd....... 30 input channels, 8 out..... with live mixing and using [readsf~] for up to 15 minute backing tracks, and I have never had Pd hang even momentarily.
I use extended for that although I doubt the [readsf~] implementation is different.
However, I use windows7....... and is it possible that Pd and the OS treat disk access slightly differently on other systems? I doubt that.

[readsf~] requires an "open" and a "start"...... so it is not forward looking as it might be in an audio editor. However if you program the "open" in advance.... or even at load, then the start is instantaneous (in ram) and you can emulate an instant player.
If Pd is stumbling during playback..... look elsewhere for the problem....... not in [readsf~]......

A multi-GHz processor with rapid disc access should not have problems with audio or Pd...... they use a paltry fraction of the resources available.

Yes..... Pd prioritises audio..... like all programs that use audio.... over video, screen drawing...... everything. So if your patch is struggling in any way you could see the Pd gui stumble.
Check thoroughly your audio and latency settings.... any mismatch with the capabilities of the soundcard could cause what you are seeing.
More here about setting up your computer for audio......... https://forum.pdpatchrepo.info/topic/10125/audio-and-video-why-do-i-have-problems ..... with a link to a good read for problem solving!
You often just need to tell your OS to do as instructed......"and I mean it!"........... Otherwise it prefers to go and peruse it's Facebook page for likes!

Maybe, otherwise, use FL studio or something similar.
Programs written specifically for audio application will have been written in assembly language and be more appropriate for your purposes.
Communication with Pd is usually facilitated through at least midi or OSC messaging.
David.

Sorry to necro this thread, but I finally found out how to run PureData under Pulseaudio (which otherwise results with "ALSA output error (snd_pcm_open): Device or resource busy").

First of all, run:

pd -alsa -listdev

PD will start, and in the message window you'll see:

audio input devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
audio output devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
API number 1

no midi input devices found
no midi output devices found

... or something similar.

Now, let's add the pulse ALSA device, and run -listdev again:

pd -alsa -alsaadd pulse -listdev

The output is now:

audio input devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
3. pulse
audio output devices:
1. HDA Intel PCH (hardware)
2. HDA Intel PCH (plug-in)
3. pulse
API number 1

no midi input devices found
no midi output devices found

Notice, how from the original two ALSA devices, now we got three - where the third one is pulse!

Now, the only thing we want to do, is that at startup (so, via the command line), we set pd to run in ALSA mode, we add the pulse ALSA device, and then we choose the third (3) device (which is to say, pulse) as the audio output device - and the command line argument for that in pd is -audiooutdev:

pd -alsa -alsaadd pulse -audiooutdev 3 ~/Desktop/mypatch.pd

Yup, now when you enable DSP, the patch mypatch.pd should play through Pulseaudio, which means it will play (and mix) with other applications that may be playing sound at the time! You can confirm that the correct output device has been selected from the command line, if you open Media/Audio Settings... once pd starts:

As the screenshot shows, now "Output device 1" is set to "pulse", which is what we needed.

Hope this helps someone!

EDIT: I had also done changes to /etc/pulse/default.pa as per https://wiki.archlinux.org/index.php/PulseAudio#ALSA.2Fdmix_without_grabbing_hardware_device beforehand, not sure whether that makes a difference or not (in any case, trying to add dmix as a PD device and playing through it, doesn't work on my Ubuntu 14.04)

I had the same difficulty. No sound cards were available according to

aplay --list-devices

I found somewhere (I forget where) the suggestion to add to

/boot/config.txt 

the line:

dtparam=audio=on

and from then everything went swimmingly. I set up some (possibly pretty dangerous) values in

/etc/security/limits.conf

like this:

### realtime permission for audio
@audio   –  rtprio     99
@audio   –  memlock    unlimited
@audio   –  nice      -10
@audio   -  priority   8
# End of file

The audio priority of 8 seems to be necessary to stop annoying messages in the puredata console when it's started. I'm not an expert on these values, they might not be wise, safe or ideal but they worked for me.

And I made sure that my user is in the audio group:

usermod -a -G audio david

And I've had glitch free performance since out of the mini jack.

Hi there, i also posted to the manjaro forum but i guess here i can find more expertise on that topic: i want to install pd-l2ork on manjaro (arch-based) linux and I get this error building flite:

Insert Co> Found flite1.patch
==> Validating source files with md5sums...
    flite-1.4-release.tar.bz2 ... Passed
    flite1.patch ... Passed
==> Extracting sources...
  -> Extracting flite-1.4-release.tar.bz2 with bsdtar
==> Starting prepare()...
patching file config/common_make_rules
==> Starting build()...
checking build system type... x86_64-unknown-linux-gnu
checking host system type... x86_64-unknown-linux-gnu
checking target system type... x86_64-unknown-linux-gnu
checking for gcc... gcc
checking for C compiler default output file name... a.out
checking whether the C compiler works... yes
checking whether we are cross compiling... no
checking for suffix of executables... 
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking for ranlib... ranlib
checking for a BSD-compatible install... /usr/bin/install -c
checking for ar... ar
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /usr/bin/grep
checking for egrep... /usr/bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking whether byte ordering is bigendian... no
checking for mmap... yes
checking sys/soundcard.h usability... yes
checking sys/soundcard.h presence... yes
checking for sys/soundcard.h... yes
checking machine/soundcard.h usability... no
checking machine/soundcard.h presence... no
checking for machine/soundcard.h... no
checking sys/audioio.h usability... no
checking sys/audioio.h presence... no
checking for sys/audioio.h... no
checking mmsystem.h usability... no
checking mmsystem.h presence... no
checking for mmsystem.h... no
configure: creating ./config.status
config.status: creating config/config
config.status: creating config/system.mak
making in ...
making in include ...
making in src ...
making in src/audio ...
gcc -fPIC  -D_FORTIFY_SOURCE=2 -I. -DCST_AUDIO_LINUX -I../../include -O3 -Wall    -c auclient.c -o /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/auclient.os
gcc -fPIC  -D_FORTIFY_SOURCE=2 -I. -DCST_AUDIO_LINUX -I../../include -O3 -Wall    -c auserver.c -o /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/auserver.os
gcc -fPIC  -D_FORTIFY_SOURCE=2 -I. -DCST_AUDIO_LINUX -I../../include -O3 -Wall    -c audio.c -o /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/audio.os
gcc -fPIC  -D_FORTIFY_SOURCE=2 -I. -DCST_AUDIO_LINUX -I../../include -O3 -Wall    -c au_streaming.c -o /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/au_streaming.os
au_streaming.c: In function ‘audio_stream_chunk’:
au_streaming.c:77:9: warning: variable ‘n’ set but not used [-Wunused-but-set-variable]
     int n;
         ^
gcc -fPIC  -D_FORTIFY_SOURCE=2 -I. -DCST_AUDIO_LINUX -I../../include -O3 -Wall    -c au_oss.c -o /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/au_oss.os
au_oss.c: In function ‘audio_open_oss’:
au_oss.c:86:25: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
     ad->platform_data = (void *)afd;
                         ^
au_oss.c: In function ‘audio_close_oss’:
au_oss.c:181:11: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
     ioctl((int)ad->platform_data, SNDCTL_DSP_SYNC, NULL);
           ^
au_oss.c:182:16: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
     rv = close((int)ad->platform_data);
                ^
au_oss.c: In function ‘audio_write_oss’:
au_oss.c:189:18: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
     return write((int)ad->platform_data,samples,num_bytes);
                  ^
au_oss.c: In function ‘audio_flush_oss’:
au_oss.c:194:18: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
     return ioctl((int)ad->platform_data, SNDCTL_DSP_SYNC, NULL);
                  ^
au_oss.c: In function ‘audio_drain_oss’:
au_oss.c:199:18: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
     return ioctl((int)ad->platform_data, SNDCTL_DSP_RESET, NULL);
                  ^
ar: ../../..//tmp/yaourt-tmp-kalimerox/aur-flite1/lib/libflite.shared.a: No such file or directory
make[2]: *** [../../config/common_make_rules:116: /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/audio/.build_so] Error 1
make[1]: *** [../config/common_make_rules:133: /tmp/yaourt-tmp-kalimerox/aur-flite1/obj/src/.make_build_dirs] Error 2
make: *** [config/common_make_rules:133: /tmp/yaourt-tmp-kalimerox/aur-flite1/obj//.make_build_dirs] Error 2
==> ERROR: A failure occurred in build().
    Aborting...
==> ERROR: Makepkg was unable to build flite1.
de 

anyone having the same issues or an advice=? ?

thanks!!

http://www.2shared.com/photo/mA24_LPF/820_am_July_26th_13_window_con.html

I conceptualized this the other day. The main reason I wanted to make this is because I'm a little tired of complicated ableton live. I wanted to just be able to right click parameters and tell them to follow midi tracks.

The big feature in this abstract is a "Midi CC Module Window" That contains an unlimited (or potentially very large)number of Midi CC Envelope Modules. In each Midi CC Envelope Module are Midi CC Envelope Clips. These clips hold a waveform that is plotted on a tempo divided graph. The waveform is played in a loop and synced to the tempo according to how long the loop is. Only one clip can be playing per module. If a parameter is right clicked, you can choose "Follow Midi CC Envelope Module 1" and the parameter will then be following the envelope that is looping in "Midi CC Envelope Module 1".

Midi note clips function in the same way. Every instrument will be able to select one Midi Notes Module. If you right clicked "Instrument Module 2" in the "Instrument Module Window" and selected "Midi input from Midi Notes Module 1", then the notes coming out of "Midi Notes Module 1" would be playing through the single virtual instrument you placed in "Instrument Module 2".

If you want the sound to come out of your speakers, then navigate to the "Bus" window. Select "Instrument Module 2" with a drop-down check off menu by right-clicking "Inputs". While still in the "Bus" window look at the "Output" window and check the box that says "Audio Output". Now the sound is coming through your speakers. Check off more Instrument Modules or Audio Track Modules to get more sound coming through the same bus.

Turn the "Aux" on to put all audio through effects.

Work in "Bounce" by selecting inputs like "Input Module 3" by right clicking and checking off Input Modules. Then press record and stop. Copy and paste your clip to an Audio Track Module, the "Sampler" or a Side Chain Audio Track Module.

Work in "Master Bounce" to produce audio clips by recording whatever is coming through the system for everyone to hear.

Chop and screw your audio in the sampler with highlight and right click processing effects. Glue your sample together and put it in an Audio Track Module or a Side Chain Audio Track Module.

Use the "Threshold Setter" to perform long linear modulation. Right click any parameter and select "Adjust to Threshold". The parameter will then adjust its minimum and maximum values over the length of time described in the "Threshold Setter".

The "Execution Engine" is used to make sure all changes happen in sync with the music.

IE>If you selected a subdivision of 2, and a length of 2, then it would take four quarter beats(starting from the next quarter beat) for the change to take place. So if you're somewhere in the a (1e+a) then you will have to wait for 2, 3, 4, 5, to pass and your change would happen on 6.

IE>If you selected a subdivision of 1 and a length of 3, you would have to wait 12 beats starting on the next quater beat.

IE>If you selected a subdivision of 8 and a length of 3, you would have to wait one and a half quarter beats starting on the next 8th note.

http://www.pdpatchrepo.info/hurleur/820_am,_July_26th_13_window_conception.png

original post by Sumidero, cut and pasted from here:

http://puredata.hurleur.com/sujet-5605-compile-install-real-time-kernel-debian

Hello everybody:
After some exchange of posts in this thread:

http://puredata.hurleur.com/sujet-5578-old-toughbook-useful-live-events

I told @katjav and @PonZo that soon I was going to post a log of a Real-time Kernel compilation and installation for Debian based machines, so that Pd would run fast and smooth in any machine, even a small netbook. I only found the log of the compilation of a desktop machine, but the process is the same, adjusting the choices to the equipment features. I hope this encourages you to compile your own Real-time kernel.
You can download the pdf file from here:

http://dl.dropbox.com/u/6086343/HowToCompileARealTimeKernelForDebian.pdf

Extra data:
You can find out the chipset name of your audio card using this command:

# cat /proc/asound/cards

There are some extra tweaks needed to develop real-time performance:

Install jack with its graphic front end:

# aptitude install qjackctl

Once installed, it prompts us to configure automatically this file:
/etc/security/limits.d/audio.conf

We agree and then we tweak this other file opening it with any text editor (always in root mode).

# gedit /etc/security/limits.conf

Then we add these lines at the bottom of the limits.conf file to assign high and real-time priorities to the @audio group:

@audio - rtprio 99
@audio - memlock unlimited
@audio - nice -10

Don't forget to add your user to the @audio group:

# adduser <userName> audio

I hope it works for you too. Best regards to all of you.

Sumidero

Hey

Once again I'm back on vista. My brief journey to puredyne goes something like this.

from puredyne
partitioned new hard drive with 4 partitions 1-swap , 2 ext4 200Gib, 1 NTFS 200GB . Copied Vista to NTFS partition with gparted(started to do this with dd but with new Gibabits stuff I was skeered but hey a platter is a platter and a cluster is a cluster right). Actually there was a 5th partition in there which was the vista restoration partition but I eventually deleted it for obvious reasons.

Removed old drive ,
took machine to ethernet connection with internet,
booted to vista ,
installed Norton virus scanner - only 1 virus, 2 tracking cookies, and one gain trickler after over a year with no virus scanner, Thats pretty good and shows you can be selective about what you download and where you visit on the web and not get any viruses.

Was trying to get windows movie maker to work with mov codec which it doesnt.
Installed auto updates for the first and only time. this took about 3 hours and still needed to install service pack 1, 2. Aborted trying to update win movie maker

****Problem 1***** MS movie maker does not work with GEM mov video output codec.

Installed latest Nvidia drivers for vista.
Woohooo OpenGl hardware acceleration GEM works so much better in vista now.

*****Problem 2********ASIO for all does not allow 2 applications to run with audio and midi at the same time. If I wanted to run Widi and play my guitar so puredata can receive midi input from guitar ASIO will not allow it but ASIO for all is the only way I can get latency down so this would be feasible.

Got frustrated.

Installed fresh copy of puredyne onto 2nd ext4 partition.
ran puredata audio latency good with fresh install. gem video bad.

used synaptics to install nvidia drivers. v185 I think.
ran GEm error no opengl config.

downloaded gdeb , flash player, google chrome, nvidia drivers from nvidia site.
installed gdeb from terminal, flash and chrome from gdeb.
rebooted to log in as root and iunstalled nvidia drivers.

woohoo hardware acceleration for opengl

******problem 3************
gem works great but audio glitches with puredata window visible.
when only gem window is visible i can unaudioglitchily move the mouse around to control same parameters that would normaly be controled with sliders in pd window.

Seems like before i upgraded nvidia video drivers hw:do o whatever it is, was listed in JackD as realtek HD Audio, now it is listed as NVIDIA HD Audio.

So Gem works great but not when Pd window is open and I think my audio drivers were replaced with Nvidia.

I would really like to get back to my patch and stop wasting time on these operating systems and hardware configurations I thought in the last 15 years all this hardware acelleration and autodetection would be figured out but no all we got is rc.0, rc.01, rc.02, rc, etc...... instead of just rc.d.

Ok I'm getting an error when I exit puredyne that says pid file not found is jackd running. anyone know what that means?

Also does anyone have a keymap for gem keyboard for linux, windows and mac so I don't have to record them.

Any suggestions about what to do about my ASIO problems in windows?
Any suggestions about my audio driver in Linux?
Any suggestions about my GEM mov video / windows incompatibilities?

I can import the mov video and audio to SF Acid 3.0 and output a 2Gb avi then import it into windows movie maker but this is crazy to do for a 7 minute 12fps 320x240 video.

Ahhhhhhhhh!
Any suggestions are welcome!

PS I believe someone could rewrite puredata's audio and video interface in assembly language to control their hardware before figuring out how to get these operating systems to do it.

hello

i have hard times on osx here...
system: osx (10.6) on a macbook pro

i downloaded pd, checked it out from git and svn, but so far there is no version i compiled which is working

when i compile the checked out version from git-repository, pd compiles, but when i start it i get this error:

2011-02-18 17:21:56.182 pd[72376:903] Error loading /Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin/Contents/MacOS/DVCPROHDAudio: dlopen(/Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin/Contents/MacOS/DVCPROHDAudio, 262): no suitable image found. Did find:
/Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin/Contents/MacOS/DVCPROHDAudio: no matching architecture in universal wrapper
2011-02-18 17:21:56.184 pd[72376:903] Cannot find function pointer NewPlugIn for factory C5A4CE5B-0BB8-11D8-9D75-0003939615B6 in CFBundle/CFPlugIn 0x100600a70 </Library/Audio/Plug-Ins/HAL/DVCPROHDAudio.plugin> (bundle, not loaded)
2011-02-18 17:21:56.189 pd[72376:903] Error loading /Library/Audio/Plug-Ins/HAL/JackRouter.plugin/Contents/MacOS/JackRouter: dlopen(/Library/Audio/Plug-Ins/HAL/JackRouter.plugin/Contents/MacOS/JackRouter, 262): no suitable image found. Did find:
/Library/Audio/Plug-Ins/HAL/JackRouter.plugin/Contents/MacOS/JackRouter: no matching architecture in universal wrapper
2011-02-18 17:21:56.190 pd[72376:903] Cannot find function pointer New_JackRouterPlugIn for factory 7CB18864-927D-48B5-904C-CCFBCFBC7ADD in CFBundle/CFPlugIn 0x100602d20 </Library/Audio/Plug-Ins/HAL/JackRouter.plugin> (bundle, not loaded)
Error in startup script: couldn't read file "5400": no such file or directory

i dont know what to do.

thanks for any help.
ingo

Hi everyone,

I'm having an issue with audio output in PD : when setting audio to output via portaudio, all the audio that gets out of PD has audio glitch in it, some kind of random-spaced crackles or clicks. It is completly useless as such. Increasing the delay in PD's audio settings doesn't change a thing.
However, using Jack as an audio outlet supresses all this problem, and the audio is clean that way.

Would anyone know the reason for this? I would be glad to know it as there may be a link with another issue about audio on my hardware : system audio of mac os X has the same kind of glitches that I got with PD on portaudio, only it's only occasional glitches so it's bearable. Solving this issue would be a big relief to me.

My hardware is a hackintosh (pc with osX installed on it), and mbox2 as audio interface (but I have tested with an m-audio transit and I have the same crackling issue). My soft is osX Snow Leopard 10.6.5.

Any help appreciated!