Yeah, of course. Block size 1 and high sampling rate will make the
timing between control and audio super tight (ChucK does this, for
example). It will also eat the hell out of your CPU. It's a trade off.
This is because you start calling all the DSP functions once every
1/192k seconds instead of once every 1.45ms.
On 3/12/2015 1:06 PM, Alexandre Torres Porres wrote:
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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