mirror of
https://github.com/saitohirga/WSJT-X.git
synced 2024-11-08 01:56:10 -05:00
5b9645bf09
git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/trunk@189 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
95 lines
3.0 KiB
Plaintext
95 lines
3.0 KiB
Plaintext
From: "Benno Senoner" <sbenno@gardena.net>
|
||
To: <music-dsp@shoko.calarts.edu>
|
||
Subject: Re: [music-dsp] coding realtime guitar efx on a "pc"
|
||
Date: Saturday, June 30, 2001 8:19 AM
|
||
|
||
|
||
Andrè,
|
||
you are solving your problem the wrong way:
|
||
you need to use a single threaded solution which does this:
|
||
|
||
- set the audio I/O parameters to fragnum=4 fragsize=128 bytes (=32samples) if
|
||
you use stereo or fragsize=64 bytes (=32 samples) if you use mono.
|
||
|
||
(do not forget to activate fulltuplex with using the _TRIGGER_ stuff)
|
||
(you need to frist deactivate audio and then start the trigger after the DAC is
|
||
prefilled (see below))
|
||
|
||
This will give you a total input to output latency of 4x32 samples
|
||
= 128 samples which at 44.1kHz correspond to 2.9msec latency.
|
||
|
||
now set your process to SCHED_FIFO (see man sched_setscheduler)
|
||
|
||
after the initialization your code should do more than less this:
|
||
|
||
- write() 4 x 32 samples to the audio fd in order to prefill the DAC.
|
||
Without this you will get dropouts.
|
||
|
||
while(1) {
|
||
read() 32 samples from ADC
|
||
perform_dsp_stuff() on the 32 samples
|
||
write() 32 samples to DAC
|
||
}
|
||
|
||
If you use a low latency kernel and pay attention to all the stuff above, then
|
||
you will get rock solid 3msec latencies (plus eventual converter latencies but
|
||
these are in the 1-2msec range AFAIK).
|
||
|
||
Using multiple threads , pipes etc, only complicates your life and often makes
|
||
it impossible to achieve these low latences.
|
||
|
||
Realtime/audio programming is not an easy task , this is why people often
|
||
fail to get the desired results even if their hardware is low-latency capable.
|
||
|
||
The problem is that the final latency depends on the hardware you use,
|
||
the application and the operating system.
|
||
|
||
cheers,
|
||
Benno.
|
||
|
||
http://www.linuxaudiodev.org The Home of Linux Audio Development
|
||
|
||
|
||
|
||
|
||
|
||
|
||
On Sat, 30 Jun 2001, you wrote:
|
||
> On 2001-06-29 21:38 +0200, Benno Senoner wrote:
|
||
>
|
||
> > OSS/Free refuses to use a low # of frags ?
|
||
> >
|
||
> > That's a myth.
|
||
>
|
||
> I hope it is. :-)
|
||
>
|
||
> The fact is that ioctl(SNDCTL_DSP_SETFRAGMENT) succeeds with
|
||
> values as low a 0x10007 (one 128-B fragment) but the latency is
|
||
> still high enough to be clearly noticeable, which suggests that
|
||
> it's *way* above 2/3 ms. This is on an otherwise idle machine
|
||
> equipped with a SB PCI 128.
|
||
>
|
||
> But maybe it's me who's doing something wrong. I've been careful
|
||
> to flush stdio buffers or use unbuffered I/O (write(2)) but I
|
||
> may have let something else through.
|
||
>
|
||
> For example, since the signal processing and the I/O are done by
|
||
> two different vanilla processes communicating via pipes, it may
|
||
> be a scheduling granularity problem (E.G. the kernel giving the
|
||
> I/O process a time slice every 20 ms).
|
||
>
|
||
> --
|
||
> André Majorel <amajorel@teaser.fr>
|
||
> http://www.teaser.fr/~amajorel/
|
||
>
|
||
> dupswapdrop -- the music-dsp mailing list and website: subscription info,
|
||
> FAQ, source code archive, list archive, book reviews, dsp links
|
||
> http://shoko.calarts.edu/musicdsp/
|
||
--
|
||
|
||
|
||
dupswapdrop -- the music-dsp mailing list and website: subscription info,
|
||
FAQ, source code archive, list archive, book reviews, dsp links
|
||
http://shoko.calarts.edu/musicdsp/
|
||
|
||
|