for the simple answer, i would say that :
-from inside a patch, pd clock accuracy is "infinite"
- from the outside, clock accuracy is related to block size and sample rate.

cheers
c

Le 12/03/2015 21:06, Alexandre Torres Porres a écrit :
Well, anyway, I was hoping for a simpler answer, and my original guess was that 
the control rate limit might be at the block size, now I'm all confused :)

I'm seeing that maybe we can't really measure the speed efficiently in a patch, 
and that Pd might actually be able to manage tiny and fasty clocks, but that 
there is also a limit on the way they are computed that depends on the block 
size, right?

What about a block size of one, and a large sample rate? I wonder if we could 
get control messages to work at 192Khz for example, huh?

One way or another, I guess there might be a pretty straightforward answer to 
this. I didn't find any in Miller's book yet.

cheers

2015-03-12 16:42 GMT-03:00 David Medine <dmed...@ucsd.edu 
<mailto:dmed...@ucsd.edu>>:


    My understanding of this patch and the guts of Pd tells me that this patch 
isn't really going to measure how long it takes between each control message. 
What it can do is show the time resolution of calls to sys_getrealtime, which 
is Pd's method of querying the CPU clock:

    double sys_getrealtime(void)
    {
    #ifndef _WIN32
         static struct timeval then;
         struct timeval now;
         gettimeofday(&now, 0);
         if (then.tv_sec == 0 && then.tv_usec == 0) then = now;
         return ((now.tv_sec - then.tv_sec) +
             (1./1000000.) * (now.tv_usec - then.tv_usec));
    #else
         LARGE_INTEGER now;
         QueryPerformanceCounter(&now);
         if (nt_freq == 0) sys_initntclock();
         return (((double)(now.QuadPart - nt_inittime.QuadPart)) / nt_freq);
    #endif
    }

    The code is from s_inter.c. It is apparent (at least in the non-Windows 
part of the code) that there is a microsecond resolution, hence 1e-6, but I 
could misunderstanding this. I was able to put 1e-7 on a Windows machine and it 
still worked -- I haven't had a chance to do try it in Unix/BSD land and I 
don't actually want to know what Windows is doing with this QuadPart of a 
LARGE_INTEGER. Still, (on Linux and Mac, anyway) 1us is the smallest unit of 
time that Pd's clocks keep track of, so that should be the limit of what 
[delay] can do.

    The actual time it takes for Pd to deal with messages depends on a great 
many things. Symbols in Pd are stored in a hash table, so I would guess that 
the size of the table (which needs to be searched) is the main factor 
controlling the rate at which those messages can be handled. However, I suspect 
that the number of symbols needed to slow Pd down on a modern computer is 
impractically large. Then there are control messages that don't have hashed 
symbols associated with them (like floats and bangs). Also, some external 
controls -- especially mouse/keyboard events and MIDI -- can be badly timed. 
These tend to queue up and get spit out at the OS's whim. Pd then simply does 
what it can with what it gets. So measuring the exact time it takes to /do/ 
control in Pd is pretty hairy. I don't believe that meaningful measurements of 
this can be done with a Pd patch.

    The other thing is that control messages get rolled up between dsp ticks 
and are then applied immediately on the start of the next tick. This means that 
two messages that are, say, .05ms apart somewhere in the midst of a 1.45ms 
block, get applied simultaneously at the start of the next block. This also 
means that at 44.1kHz with a block size of 64 samples, both of them may be 
anywhere from 0.02 ms to 1.45ms late -- depending on where they fall in 
relation to block boundaries. This also, also means that if one control message 
happens near the end of one block and the other happens near the start of the 
next block, their distance of .05ms in physical time will be expanded to 
1.45ms. This is a very big, teeny-tiny problem in real-time audio programming 
because under certain conditions there can be serious (audible) repercussions. 
This is why there is [vline~], by the way.

    If anyone else is interested in this stuff, I recommend these lectures 
(Miller's is the first in the series):
    http://repmus.ircam.fr/mutant/rtmseminars

    -David


    On 3/12/2015 10:25 AM, Alexandre Torres Porres wrote:
    > timer and realtime compute 2 different things
    > (logical time and real time). i don"t know what
    > your want.

    I know they are different, and I don't really know what I want either :)

    I just wanted to measure how long it takes between each control message.

    you were using [realtime], and then Roman came in and said that'd be kinda 
random and how [timer] was best for it. So I tried with [timer] and got a very 
nice result indeed. But I'm not sure now if that actually relates to whats 
going on... or how it is actually working.

    > what i don't understand is your intention with
    > the spigot in the patch.

    just wanted to have a way to close the message stream, but you can forget 
about it

    cheers

    2015-03-12 14:14 GMT-03:00 Cyrille Henry <c...@chnry.net 
<mailto:c...@chnry.net>>:



        Le 12/03/2015 18:04, Alexandre Torres Porres a écrit :

            "/i don't understand your patch.

            using [timer], a delay 0 will give a 0 delay...
            logical time will always be consistent./"

            well, I thought you were disucussing here and reaching the 
conclusion that [timer] is the one to be used to calculate this...

        timer and realtime compute 2 different things (logical time and real time). 
i don"t know what your want.

        what i don't understand is your intention with the spigot in the patch.

        cheers
        c


            So you mean this result is actually inconsistent? And the 
implication is that it is not going at that super fast rate at all? Please help 
me understand better about how to measure this.

            thanks


            2015-03-12 11:55 GMT-03:00 Cyrille Henry <c...@chnry.net <mailto:c...@chnry.net> 
<mailto:c...@chnry.net <mailto:c...@chnry.net>>>:

                hello,

                i don't understand your patch.

                using [timer], a delay 0 will give a 0 delay...
                logical time will always be consistent.


                cheers
                c


                Le 12/03/2015 15:41, Alexandre Torres Porres a écrit :

                    ok, so the metro at 1ms is because I'm using extended.

                    as for the minimum time pd can process and send data, 
what's the final word on it?

                    something like 1.4013e-45 ms?

                    cause that's a lot more than an audio rate at 44.1khz :)

                    I thought there was a limit control rate that was below the 
audio rate, but curiously it can go over.

                    1 sample at 44.1khz gives us 0.0226757 ms, and I was able 
to send bangs at 1e-06 ms, according to [timer]

                    check my patch attached, based on the one that was sent 
here on the thread.

                    thanks

                    2015-03-12 10:04 GMT-03:00 Cyrille Henry <c...@chnry.net <mailto:c...@chnry.net> 
<mailto:c...@chnry.net <mailto:c...@chnry.net>> <mailto:c...@chnry.net <mailto:c...@chnry.net> 
<mailto:c...@chnry.net <mailto:c...@chnry.net>>>>:

                         hello,

                         Le 12/03/2015 10:12, Roman Haefeli a écrit :

                             On Thu, 2015-03-12 at 09:17 +0100, Cyrille Henry 
wrote:

                                 hello

                                 this patch show the same behaviors for a delay 
based metro and a [metro].
                                 (both can do faster than 1ms period)


                             You're right. More recent versions of Pd (>= 
0.45?) have an updated
                             [metro] that supports many more ways to specify 
time and the restriction
                             was lowered. However, the [metro] in any available 
version of
                             Pd-extended is still limited to 1ms.

                         sorry, i was not aware of this old limitation.


                             I don't understand why you use [realtime] and not 
[timer] to illustrate
                             your point. [timer] gives you consistent values 
(logical time) while
                             [realtime] is very jittery and shows just some 
random value depending on
                             the current cpu usage and probably other factors. 
When you render a
                             soundfile, the logical time is actually the one 
that matters.

                         yes, for things that stay in pd, logical time is 
better.
                         but if you want to send midi note, [realtime] is more 
related to what happens.
                         it's just the way i understand the original question.

                         cheers
                         c



                             Roman



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