Re: [PD] Gem with Intel HD 4000 F/MSAA
hi, I think it depends on the driver. I heard it doesn't work on Ubuntu 12.04 with crapy intel driver (and unfortunately this is not the only issue on Linux with Intel cards). + a -- do it yourself http://antoine.villeret.free.fr 2013/3/6 pured...@11h11.com Is it possible to use anti-aliasing with an Intel HD 4000? http://communities.intel.com/**thread/31912http://communities.intel.com/thread/31912 Intel graphics built into the last two generations support MSAA I tried to send MSAA 1 || 2 || 4 || 8 = not working. I think Gem only support FSAA? __**_ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/** listinfo/pd-list http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] wireless audio from Pd to PA system (katja)
Netjack docs look promising. It binds Jack clients on the transmitter
computer (called 'slave') to the sample clock on the receiver computer
('master'). Netjack supports transmission over wifi, and over internet
(using the CELT codec). It is primarily designed for 'distributed
music'. Here is a paper introducing Netjack:
http://www.google.nl/url?q=http://lac.linuxaudio.org/2009/cdm/Saturday/22_Hohn/22.pdfsa=Uei=0AM3UajlJIfAO5qDgIgFved=0CDgQFjAJusg=AFQjCNF47yIrp3T6cuIXCKhl0AUIaeiA2w
For anyone using Pd on a mobile device as sound generator, Netjack
could be a great way to deliver audio over wifi. For my purpose
though, I still need access to the sound card at the transmitter side
to capture audio from a mic. An audio rate conversion routine called
AlsaInOut is provided with Netjack so you can use the slave's sound
card for playback. However, this can not be used for capturing or
monitoring, due to large processing delay. So, clock drift is the show
stopper here.
Clock drift is not related to network transmission but to the slight
differences in clock speeds of sound cards even if they are set to the
same nominal sample rate. It also happens with two sound cards used
together on one computer, like when using an USB mic for input and the
system sound card for output. Core Audio can handle such a setup, but
at the expense of increased and fluctuating latency. Jack can combine
soundcards, but buffer under- or overruns are audible.
What is so hard about compensating clock drift? If it must be done in
a theoretically correct way, pitch should not change so it will be a
time stretching routine which will indeed introduce considerable
latency. However, clock speeds often differ by a very small amount,
like a factor 1/1000 or less. What if we just accept the pitch
difference? You would get playback speeds like 1.001 or 0.999,
perfectly acceptable for my purpose at least. This would greatly
simplify the task of resampling.
Katja
On 3/5/13, Charles Goyard c...@fsck.fr wrote:
Hi,
see if netjack fills the bill :
Netjack is a Realtime Audio Transport over a generic IP Network. It is
fully integrated into JACK.
When you transport Audio Signals over a Network, you will always have
the problem, that two Computers are bound to two different Sample
Clocks. The other Network Transports face the Problem of compensating
for the drift between these Sample Clocks. This is very hard to do.
NetJack works around this problem, by syncing all Clients to one
Soundcard. So there will be no resampling or glitches in the whole
network. Packet loss is now also handled gracefully. By using the celt
codec, its even possible, that single packet losses get masked by the
Packet Loss Concealment Code.
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack2
Cheers,
--
Charles
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Re: [PD] wireless audio from Pd to PA system (katja)
Hi Katja,
Regarding all this with...
Digital clocks need to be synchronized to work together or else you will
get clicks. As you have said it can be done in a driver or it can be
done on-chip if your hardware has that feature. It does indeed add to
the latency. On pro sound cards you would find word clock to sync cards
together or possibly time code input/output for synchronising with video
cameras. Consumer hardware does not have these features since most
people wouldn't have any use for this. Some equipment can hold a very
stable clock even after being disconnected from the others. This feature
comes with added costs.
Super expensive digital radio mics have a noticeable latency even though
they are highly specialised equipment built to minimise this very
latency. It's about 3ms for a good setup.
Anyhow the lowest latency you will find is using analog wireless packs.
The only point of digital radio mics is Hollywood can't probably cope
with non-encrypted dialogues of your favourite TV series during the shoot.
Don't go for the cheapest, and you might not need the high end. You can
rent that stuff if it's not needed for such a long time.
The Sennheiser e100 (gen3) is pretty decent for the price. That would be
about 450~500€ for a kit.
There is a pro level audio-over-ethernet distribution system called
ethersound. It's proprietary, but worth looking into. As with other
digital format, there is a master clock somewhere in the system to which
every one syncs. Once again digital converters are the main culprits
when it comes to latency.
http://www.ethersound.com/technology/overview.php
Cheers
Pierre-Olivier
On 06/03/2013 12:41, katja wrote:
Netjack docs look promising. It binds Jack clients on the transmitter
computer (called 'slave') to the sample clock on the receiver computer
('master'). Netjack supports transmission over wifi, and over internet
(using the CELT codec). It is primarily designed for 'distributed
music'. Here is a paper introducing Netjack:
http://www.google.nl/url?q=http://lac.linuxaudio.org/2009/cdm/Saturday/22_Hohn/22.pdfsa=Uei=0AM3UajlJIfAO5qDgIgFved=0CDgQFjAJusg=AFQjCNF47yIrp3T6cuIXCKhl0AUIaeiA2w
For anyone using Pd on a mobile device as sound generator, Netjack
could be a great way to deliver audio over wifi. For my purpose
though, I still need access to the sound card at the transmitter side
to capture audio from a mic. An audio rate conversion routine called
AlsaInOut is provided with Netjack so you can use the slave's sound
card for playback. However, this can not be used for capturing or
monitoring, due to large processing delay. So, clock drift is the show
stopper here.
Clock drift is not related to network transmission but to the slight
differences in clock speeds of sound cards even if they are set to the
same nominal sample rate. It also happens with two sound cards used
together on one computer, like when using an USB mic for input and the
system sound card for output. Core Audio can handle such a setup, but
at the expense of increased and fluctuating latency. Jack can combine
soundcards, but buffer under- or overruns are audible.
What is so hard about compensating clock drift? If it must be done in
a theoretically correct way, pitch should not change so it will be a
time stretching routine which will indeed introduce considerable
latency. However, clock speeds often differ by a very small amount,
like a factor 1/1000 or less. What if we just accept the pitch
difference? You would get playback speeds like 1.001 or 0.999,
perfectly acceptable for my purpose at least. This would greatly
simplify the task of resampling.
Katja
On 3/5/13, Charles Goyard c...@fsck.fr wrote:
Hi,
see if netjack fills the bill :
Netjack is a Realtime Audio Transport over a generic IP Network. It is
fully integrated into JACK.
When you transport Audio Signals over a Network, you will always have
the problem, that two Computers are bound to two different Sample
Clocks. The other Network Transports face the Problem of compensating
for the drift between these Sample Clocks. This is very hard to do.
NetJack works around this problem, by syncing all Clients to one
Soundcard. So there will be no resampling or glitches in the whole
network. Packet loss is now also handled gracefully. By using the celt
codec, its even possible, that single packet losses get masked by the
Packet Loss Concealment Code.
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack
https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_NetJack2
Cheers,
--
Charles
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Re: [PD] Gem with Intel HD 4000 F/MSAA
I forgot to specify my OS. I am using Ubuntu Studio 12.04 and found this posted: http://www.phoronix.com/scan.php?page=news_itempx=MTE0MjY ...Intel's Linux graphics driver is finally onto supporting 8x multi-sample anti-aliasing (MSAA)... The question: Is GEM supporting MSAA? I looked in GEM source and found this: if (m_fsaa!=0) setenv(__GL_FSAA_MODE, svalue, 1); // this works only for NVIDIA-cards Also this: if(GLEW_ARB_multisample) glEnable (GL_MULTISAMPLE_ARB); if(GLEW_NV_multisample_filter_hint) glHint (GL_MULTISAMPLE_FILTER_HINT_NV, GL_NICEST); I am wondering how to turn in on? ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] Gem with Intel HD 4000 F/MSAA
Hi, Doesn't 12.04 mean you have a intel driver that is rather old ? Or at least predates the announcement you're referring to ? Just asking, I know nothing about Ubuntu. pured...@11h11.com wrote: I forgot to specify my OS. I am using Ubuntu Studio 12.04 and found this posted: http://www.phoronix.com/scan.php?page=news_itempx=MTE0MjY ...Intel's Linux graphics driver is finally onto supporting 8x multi-sample anti-aliasing (MSAA)... The question: Is GEM supporting MSAA? I looked in GEM source and found this: if (m_fsaa!=0) setenv(__GL_FSAA_MODE, svalue, 1); // this works only for NVIDIA-cards Also this: if(GLEW_ARB_multisample) glEnable (GL_MULTISAMPLE_ARB); if(GLEW_NV_multisample_filter_hint) glHint (GL_MULTISAMPLE_FILTER_HINT_NV, GL_NICEST); I am wondering how to turn in on? ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] wireless audio from Pd to PA system (katja)
Hi all -
It's correct that if you're using non-synchronized udio devces, soner
or later some FIFO will over- or under-flow and you'll get dropped
audio samples. But lots of people seem happy to live with this - for
instance, I believe anyone using Jacktrip for multi-site performances is
doing precisely that. (Me too - at home, for driver-ish reasons, I have
to use one audio device for input and a different one for output, and
no way to sync them).
If the clocks in question are accurate to a part in a million you'll get
1 msec of slip every 20 minutes or so. And a well-designed audio driver
could even see the trouble coming and do a quick cross-fade when necessary,
or choose a moment of silence to resynchronize every 10 minutes or so.
And/or, one could actually resample things. I believe this is how
Cobranet (one of the 'professional' solutions) does it. Resampling does
add a few msec of latency if done at 44K1 or 48K, but if you run at higher
rates (and if you don't mind non-flat frequency responses above 20kHz) I
believe you can do much much better, around 0.1 msec. I remember hearing a
good paper on this at the 2012 LAC (Stanford).
cheers
Miller
On Wed, Mar 06, 2013 at 01:51:12PM +0100, Pierre-Olivier Boulant wrote:
Hi Katja,
Regarding all this with...
Digital clocks need to be synchronized to work together or else you
will get clicks. As you have said it can be done in a driver or it
can be done on-chip if your hardware has that feature. It does
indeed add to the latency. On pro sound cards you would find word
clock to sync cards together or possibly time code input/output for
synchronising with video cameras. Consumer hardware does not have
these features since most people wouldn't have any use for this.
Some equipment can hold a very stable clock even after being
disconnected from the others. This feature comes with added costs.
Super expensive digital radio mics have a noticeable latency even
though they are highly specialised equipment built to minimise this
very latency. It's about 3ms for a good setup.
Anyhow the lowest latency you will find is using analog wireless
packs. The only point of digital radio mics is Hollywood can't
probably cope with non-encrypted dialogues of your favourite TV
series during the shoot.
Don't go for the cheapest, and you might not need the high end. You
can rent that stuff if it's not needed for such a long time.
The Sennheiser e100 (gen3) is pretty decent for the price. That
would be about 450~500€ for a kit.
There is a pro level audio-over-ethernet distribution system called
ethersound. It's proprietary, but worth looking into. As with other
digital format, there is a master clock somewhere in the system to
which every one syncs. Once again digital converters are the main
culprits when it comes to latency.
http://www.ethersound.com/technology/overview.php
Cheers
Pierre-Olivier
On 06/03/2013 12:41, katja wrote:
Netjack docs look promising. It binds Jack clients on the transmitter
computer (called 'slave') to the sample clock on the receiver computer
('master'). Netjack supports transmission over wifi, and over internet
(using the CELT codec). It is primarily designed for 'distributed
music'. Here is a paper introducing Netjack:
http://www.google.nl/url?q=http://lac.linuxaudio.org/2009/cdm/Saturday/22_Hohn/22.pdfsa=Uei=0AM3UajlJIfAO5qDgIgFved=0CDgQFjAJusg=AFQjCNF47yIrp3T6cuIXCKhl0AUIaeiA2w
For anyone using Pd on a mobile device as sound generator, Netjack
could be a great way to deliver audio over wifi. For my purpose
though, I still need access to the sound card at the transmitter side
to capture audio from a mic. An audio rate conversion routine called
AlsaInOut is provided with Netjack so you can use the slave's sound
card for playback. However, this can not be used for capturing or
monitoring, due to large processing delay. So, clock drift is the show
stopper here.
Clock drift is not related to network transmission but to the slight
differences in clock speeds of sound cards even if they are set to the
same nominal sample rate. It also happens with two sound cards used
together on one computer, like when using an USB mic for input and the
system sound card for output. Core Audio can handle such a setup, but
at the expense of increased and fluctuating latency. Jack can combine
soundcards, but buffer under- or overruns are audible.
What is so hard about compensating clock drift? If it must be done in
a theoretically correct way, pitch should not change so it will be a
time stretching routine which will indeed introduce considerable
latency. However, clock speeds often differ by a very small amount,
like a factor 1/1000 or less. What if we just accept the pitch
difference? You would get playback speeds like 1.001 or 0.999,
perfectly acceptable for my purpose at least. This would greatly
simplify the task of resampling.
Katja
On 3/5/13, Charles Goyard c...@fsck.fr wrote:
Re: [PD] file format for GEM
Le 06/03/2013 19:09, Stephan Elliot Perez a écrit : Hi, I am working on a computer with two 2,66 GHz Dual-Core Intel Xeon processors (so, slightly weaker than the other) and when I play the 800x450 videos in 1920 x 1080 format, I again have the problem that, while the CPU usage is fine, PD itself slows down considerably. The cues are executed too late and the video is of course slower. Does anyone know what causes this problem specifically and what can be done about it? pd time shift from real time. Your computer can't do what pd ask it to do. if cpu is fine, then the bottleneck is not the cpu. maybe the bus between cpu and gpu. you can reduce video size, or change computer. cheers c Many thanks, Stephan On Mon, Mar 4, 2013 at 7:31 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com mailto:dreamoftheshoreofanotherwo...@gmail.com wrote: Well, I do not know why, but if I use auto for everything (instead of line objects into the right inlet of pix_film), it does fine with one or two videos, but, when a third one is activated, it skyrockets to over 100. And this is using 800x450 videos. With this format, I had no such CPU problems using line objects. -Stephan On Mon, Mar 4, 2013 at 6:01 PM, chris clepper cgclep...@gmail.com mailto:cgclep...@gmail.com wrote: Use the 'rate $1' message to change the playback speed. On Mon, Mar 4, 2013 at 11:31 AM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com mailto:dreamoftheshoreofanotherwo...@gmail.com wrote: I am using pix_film to play back the files, do I need to use another object? I ask because when I send the messages auto -1, 1 or 2 into the first inlet, the playback is always the same (1), and if I send auto .5, it turns off (as with 0). On Sun, Mar 3, 2013 at 8:45 PM, chris clepper cgclep...@gmail.com mailto:cgclep...@gmail.com wrote: A laptop drive will probably not play more than two HD ProRes files at once. An external Firewire or USB drive might help if half the clips are on it and the other half on the internal. Auto can play at any speed forward or backward: try 'rate -1' or 'rate 1.5' etc. That sets the Quicktime clock for playback to that rate. It's more efficient to let QT do the tasking internally. You can select the starting frame of playback using the frame number into the second inlet. On Sun, Mar 3, 2013 at 2:34 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com mailto:dreamoftheshoreofanotherwo...@gmail.com wrote: I am not using the auto message because I often want to play the files backwards or only play a certain part of them. Unless there is a variant of auto for this? When I use the patch for a performance it will be on a laptop, which I am not certain will have multiple hard drives. Would a solid state drive definitely fix my problem? Otherwise, I suppose I will be stuck with a lower resolution for the videos. -Stephan On Sun, Mar 3, 2013 at 7:22 PM, chris clepper cgclep...@gmail.com mailto:cgclep...@gmail.com wrote: Are you using 'auto 1' to play the files? That uses Quicktime to determine the current position which is more efficient than sending a frame number. You may also want to try 'frame 60' into the gemwin with the auto message for smoother looking output. That basically syncs the render output with the screen and 'auto 1' only loads a new frame at the rate of the file. Those CPU figures seem about right. The ProRes HQ files are probably much larger on disk than the AIC files, so the extra CPU time might be waiting for the disk. That might also be why the AIC files spike at certain points because the drive is working more. Disk speed is an important factor here: I would spread the files over multiple drives if possible. Obviously, SSD is a good option too. Chris On Sun, Mar 3, 2013 at 1:05 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com mailto:dreamoftheshoreofanotherwo...@gmail.com wrote: So, I added the -nomidi -noaudio and -nrt commands as instructed by Mr. Clepper. This seems to help stop lag in Pure Data itself (so cues are executed punctually) when using HD-files. However, I then converted my files to 1920 x 1080 at 100% using the prores 422 (HQ) codec. The CPU load still climbs to into the 80s and 90s (even over 100 once) with two videos and into the 120s-160s with three. Oddly, with the Intermediary Codec, the load is sometimes much lower (in the 50s for two files) but sharply climbs at other moments (higher than with the pro-res codecs).
Re: [PD] file format for GEM
Double check that the following Startup flags are set: -nrt -noaudio -nomidi On Wed, Mar 6, 2013 at 1:09 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com wrote: Hi, I am working on a computer with two 2,66 GHz Dual-Core Intel Xeon processors (so, slightly weaker than the other) and when I play the 800x450 videos in 1920 x 1080 format, I again have the problem that, while the CPU usage is fine, PD itself slows down considerably. The cues are executed too late and the video is of course slower. Does anyone know what causes this problem specifically and what can be done about it? Many thanks, Stephan On Mon, Mar 4, 2013 at 7:31 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com wrote: Well, I do not know why, but if I use auto for everything (instead of line objects into the right inlet of pix_film), it does fine with one or two videos, but, when a third one is activated, it skyrockets to over 100. And this is using 800x450 videos. With this format, I had no such CPU problems using line objects. -Stephan On Mon, Mar 4, 2013 at 6:01 PM, chris clepper cgclep...@gmail.comwrote: Use the 'rate $1' message to change the playback speed. On Mon, Mar 4, 2013 at 11:31 AM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com wrote: I am using pix_film to play back the files, do I need to use another object? I ask because when I send the messages auto -1, 1 or 2 into the first inlet, the playback is always the same (1), and if I send auto .5, it turns off (as with 0). On Sun, Mar 3, 2013 at 8:45 PM, chris clepper cgclep...@gmail.comwrote: A laptop drive will probably not play more than two HD ProRes files at once. An external Firewire or USB drive might help if half the clips are on it and the other half on the internal. Auto can play at any speed forward or backward: try 'rate -1' or 'rate 1.5' etc. That sets the Quicktime clock for playback to that rate. It's more efficient to let QT do the tasking internally. You can select the starting frame of playback using the frame number into the second inlet. On Sun, Mar 3, 2013 at 2:34 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com wrote: I am not using the auto message because I often want to play the files backwards or only play a certain part of them. Unless there is a variant of auto for this? When I use the patch for a performance it will be on a laptop, which I am not certain will have multiple hard drives. Would a solid state drive definitely fix my problem? Otherwise, I suppose I will be stuck with a lower resolution for the videos. -Stephan On Sun, Mar 3, 2013 at 7:22 PM, chris clepper cgclep...@gmail.comwrote: Are you using 'auto 1' to play the files? That uses Quicktime to determine the current position which is more efficient than sending a frame number. You may also want to try 'frame 60' into the gemwin with the auto message for smoother looking output. That basically syncs the render output with the screen and 'auto 1' only loads a new frame at the rate of the file. Those CPU figures seem about right. The ProRes HQ files are probably much larger on disk than the AIC files, so the extra CPU time might be waiting for the disk. That might also be why the AIC files spike at certain points because the drive is working more. Disk speed is an important factor here: I would spread the files over multiple drives if possible. Obviously, SSD is a good option too. Chris On Sun, Mar 3, 2013 at 1:05 PM, Stephan Elliot Perez dreamoftheshoreofanotherwo...@gmail.com wrote: So, I added the -nomidi -noaudio and -nrt commands as instructed by Mr. Clepper. This seems to help stop lag in Pure Data itself (so cues are executed punctually) when using HD-files. However, I then converted my files to 1920 x 1080 at 100% using the prores 422 (HQ) codec. The CPU load still climbs to into the 80s and 90s (even over 100 once) with two videos and into the 120s-160s with three. Oddly, with the Intermediary Codec, the load is sometimes much lower (in the 50s for two files) but sharply climbs at other moments (higher than with the pro-res codecs). -Stephan On Sun, Mar 3, 2013 at 10:10 AM, Peter Venus n...@petervenus.dewrote: Hello! i did not know, that you wanted to playback HD-material. with HD material, i notice problems as well, also with the mjpeg codec. anyone having experience with fullHD and other codecs? under OSX i found, that apples ProRes 422 codec works best for that matter. The only thing being, that its comes with final cut. right now, i am running a show, where i use mjpeg in 720p resolution with no problems for simultaneous playback of 3 videos. cheers, peter Am 01.03.13 21:39, schrieb Stephan Elliot Perez: So, I reduced the resolution of the files from 1920 x 1080 to 800 x 450 with the Apple Photo -Jpeg codec and now I have no
[PD] crude pd interface for omxplayer/Rpi
Hi list Looking for a way to control video playback on the Raspberry Pi outside Gem's realm, i stumbled across omxplayer and this topic http://www.raspberrypi.org/phpBB3/viewtopic.php?f=38t=7987. So based on it i've made a very crude [shell]-based omxplayer controller in Pd, see attached. As omxplayer can be controlled via keystrokes, you can expand this patch's possibilities very easily. For now, you can choose a file, play it, pause and resume, restart the video, quit omxplayer, increase and decrease volume. I encounter an annoying bug when restarting a video : one can briefly see what's behind the window, be it a console or lxde's desktop. Happily, omxplayer has also a counter-bug thats prevents the desktop to refresh itself (it stays black : you have to move the mouse all over the screen to refresh). So to keep the restart prettier, you have to launch the desktop manager. #N canvas 145 350 426 301 10; #X obj 81 136 openpanel; #X obj 81 44 bng 15 250 50 0 empty empty file 17 7 0 10 -262144 -1 -1; #X msg 242 60 pause; #X msg 242 79 restart; #X msg 242 98 quit; #X obj 81 200 print; #X obj 192 202 bng 15 250 50 0 empty empty done 17 7 0 10 -262144 -1 -1; #X msg 242 41 go; #X msg 242 117 vol+; #X msg 242 136 vol-; #N canvas 188 254 918 528 omxplayercontrol 0; #X obj 60 234 shell; #X msg 214 182 mkfifo /tmp/cmd; #X msg 446 188 echo . /tmp/cmd; #X obj 446 234 shell; #X msg 241 163 rm /tmp/cmd; #X obj 60 25 inlet file; #X obj 214 125 t b b; #X msg 60 121 omxplayer \$1 /tmp/cmd; #X obj 446 4 inlet commands; #X obj 446 29 symbol; #X obj 473 324 outlet done; #X obj 60 324 outlet stdout; #X msg 473 287 bang; #X obj 214 144 delay 1000; #X msg 564 188 echo -n \$1 /tmp/cmd; #X msg 539 126 i; #X msg 585 125 q; #X msg 492 126 p; #X msg 632 124 +; #X msg 678 124 -; #X obj 564 169 symbol; #X obj 446 58 select go pause restart quit vol+ vol-; #X obj 214 75 loadbang; #X obj 558 79 t b b; #X obj 360 135 delay 500; #X connect 0 0 11 0; #X connect 0 1 12 0; #X connect 1 0 0 0; #X connect 2 0 3 0; #X connect 3 0 11 0; #X connect 3 1 12 0; #X connect 4 0 0 0; #X connect 5 0 7 0; #X connect 6 0 13 0; #X connect 6 1 4 0; #X connect 7 0 0 0; #X connect 8 0 9 0; #X connect 9 0 21 0; #X connect 12 0 10 0; #X connect 13 0 1 0; #X connect 14 0 3 0; #X connect 15 0 20 0; #X connect 16 0 20 0; #X connect 17 0 20 0; #X connect 18 0 20 0; #X connect 19 0 20 0; #X connect 20 0 14 0; #X connect 21 0 2 0; #X connect 21 1 17 0; #X connect 21 2 15 0; #X connect 21 3 23 0; #X connect 21 4 18 0; #X connect 21 5 19 0; #X connect 22 0 6 0; #X connect 23 0 24 0; #X connect 23 1 16 0; #X connect 24 0 4 0; #X restore 81 155 pd omxplayercontrol; #X connect 0 0 10 0; #X connect 1 0 0 0; #X connect 2 0 10 1; #X connect 3 0 10 1; #X connect 4 0 10 1; #X connect 7 0 10 1; #X connect 8 0 10 1; #X connect 9 0 10 1; #X connect 10 0 5 0; #X connect 10 1 6 0; ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
[PD] found how to reproduce Pd-ext 0.43.4 Tcl Invalid Command Name error
hey, dunno if you remember, but I still have this error (below) and now I managed to make a small patch that reproduces it (attached). It seems related to the Hide flag for a 2nd level nested GOP patch. it'd be great if somebody could test it on linux and mac. I'm on Ubuntu Lucid 10.04, pd-ext 0.43.4 how to reproduce: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - close the subpatch - close MAIN-graph-bug.pd at this point Pd throws the error as below. Only the GUI freezes, the patch is unusable and have to kill it, by closing pd. (Tcl) INVALID COMMAND NAME: invalid command name .x996ebd0.c while executing .x996ebd0.c delete graph996f4b0i0 (uplevel body line 1) invoked from within uplevel #0 $cmds_from_pd How to avoid it: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - open the subpatch - open the further subpatch anlz.scope~ - flag hide object name and argument - save -close pd - restart the patch and the error disappear It is worth noting that the error I get with the Xth Sense software looks similar but has different tags (see below). And I can't reproduce this one error using a subpatch including a graph or iem_image (which I use in the Xth Sense) (Tcl) INVALID COMMAND NAME: invalid command name .x9c4d3b0.c while executing .x9c4d3b0.c create image 900 776 -image a4304c0PHOTOIMAGE -tags a4304c0PHOTO (uplevel body line 283) invoked from within uplevel #0 $cmds_from_pd \ should i submit a different bug report or add to the one I did already? thanks in advance for any hint, this is forcing me to still use pd-ext 0.42.5 during my workshop, which is a shame :) -- Marco Donnarumma New Media + Sonic Arts Practitioner, Performer, Teacher, Director. Embodied Audio-Visual Interaction Research Team. Department of Computing, Goldsmiths University of London ~ Portfolio: http://marcodonnarumma.com Research: http://res.marcodonnarumma.com Director: http://www.liveperformersmeeting.net invalid-command-name.tar.gz Description: GNU Zip compressed data ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] found how to reproduce Pd-ext 0.43.4 Tcl Invalid Command Name error
FWIW, I've done a lot of work at cleaning up gop-related bugs in pd-l2ork. This is one of them (in other words, pd-l2ork is not affected by this). It may not be a bad idea to do some code comparison between g_editor.c g_canvas.c and g_graph.c files where most of these reside in hope of merging this into regular pd/extended. On 03/06/2013 08:02 PM, Marco Donnarumma wrote: hey, dunno if you remember, but I still have this error (below) and now I managed to make a small patch that reproduces it (attached). It seems related to the Hide flag for a 2nd level nested GOP patch. it'd be great if somebody could test it on linux and mac. I'm on Ubuntu Lucid 10.04, pd-ext 0.43.4 how to reproduce: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - close the subpatch - close MAIN-graph-bug.pd at this point Pd throws the error as below. Only the GUI freezes, the patch is unusable and have to kill it, by closing pd. (Tcl) INVALID COMMAND NAME: invalid command name .x996ebd0.c while executing .x996ebd0.c delete graph996f4b0i0 (uplevel body line 1) invoked from within uplevel #0 $cmds_from_pd How to avoid it: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - open the subpatch - open the further subpatch anlz.scope~ - flag hide object name and argument - save -close pd - restart the patch and the error disappear It is worth noting that the error I get with the Xth Sense software looks similar but has different tags (see below). And I can't reproduce this one error using a subpatch including a graph or iem_image (which I use in the Xth Sense) (Tcl) INVALID COMMAND NAME: invalid command name .x9c4d3b0.c while executing .x9c4d3b0.c create image 900 776 -image a4304c0PHOTOIMAGE -tags a4304c0PHOTO (uplevel body line 283) invoked from within uplevel #0 $cmds_from_pd \ should i submit a different bug report or add to the one I did already? thanks in advance for any hint, this is forcing me to still use pd-ext 0.42.5 during my workshop, which is a shame :) -- Marco Donnarumma New Media + Sonic Arts Practitioner, Performer, Teacher, Director. Embodied Audio-Visual Interaction Research Team. Department of Computing, Goldsmiths University of London ~ Portfolio: http://marcodonnarumma.com Research: http://res.marcodonnarumma.com Director: http://www.liveperformersmeeting.net ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list -- Ivica Ico Bukvic, D.M.A Composition, Music Technology Director, DISIS Interactive Sound Intermedia Studio Director, L2Ork Linux Laptop Orchestra Head, ICAT IMPACT Studio Virginia Tech Department of Music Blacksburg, VA 24061-0240 (540) 231-6139 (540) 231-5034 (fax) disis.music.vt.edu l2ork.music.vt.edu ico.bukvic.net ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] found how to reproduce Pd-ext 0.43.4 Tcl Invalid Command Name error
start pd with the flag -stderr i had the same plague with my GEM patches for a week until Iohannes ahd Hans noted a solution pp From: pd-list-boun...@iem.at [pd-list-boun...@iem.at] on behalf of Marco Donnarumma [de...@thesaddj.com] Sent: Wednesday, March 06, 2013 8:02 PM To: pd-list@iem.at Subject: [PD] found how to reproduce Pd-ext 0.43.4 Tcl Invalid Command Name error hey, dunno if you remember, but I still have this error (below) and now I managed to make a small patch that reproduces it (attached). It seems related to the Hide flag for a 2nd level nested GOP patch. it'd be great if somebody could test it on linux and mac. I'm on Ubuntu Lucid 10.04, pd-ext 0.43.4 how to reproduce: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - close the subpatch - close MAIN-graph-bug.pd at this point Pd throws the error as below. Only the GUI freezes, the patch is unusable and have to kill it, by closing pd. (Tcl) INVALID COMMAND NAME: invalid command name .x996ebd0.c while executing .x996ebd0.c delete graph996f4b0i0 (uplevel body line 1) invoked from within uplevel #0 $cmds_from_pd How to avoid it: - launch MAIN-graph-bug.pd - click the bang to open a subpatch (if it doesn't at startup) - open the subpatch - open the further subpatch anlz.scope~ - flag hide object name and argument - save -close pd - restart the patch and the error disappear It is worth noting that the error I get with the Xth Sense software looks similar but has different tags (see below). And I can't reproduce this one error using a subpatch including a graph or iem_image (which I use in the Xth Sense) (Tcl) INVALID COMMAND NAME: invalid command name .x9c4d3b0.c while executing .x9c4d3b0.c create image 900 776 -image a4304c0PHOTOIMAGE -tags a4304c0PHOTO (uplevel body line 283) invoked from within uplevel #0 $cmds_from_pd \ should i submit a different bug report or add to the one I did already? thanks in advance for any hint, this is forcing me to still use pd-ext 0.42.5 during my workshop, which is a shame :) -- Marco Donnarumma New Media + Sonic Arts Practitioner, Performer, Teacher, Director. Embodied Audio-Visual Interaction Research Team. Department of Computing, Goldsmiths University of London ~ Portfolio: http://marcodonnarumma.com Research: http://res.marcodonnarumma.com Director: http://www.liveperformersmeeting.net ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
