hello list,
i just bought a hammerfall DIGI96/8 card, and now i'm trying to configure pd to get the lowest possible latency out of it. i first tried with pd-0.36, then with pd-0.37. with both some strange things happened. (<ashamed to admit> using win ME</> on a DELL pentium III 64meg RAM)
pd0.36 seems to change the number of audio buffers i ask for, sometimes lowering them to 1!
the details:
when i try: "pd -asio -listdev -blocksize 32 -audiobuf 4:
pd first says "4 audio buffers", then "bufs 8" in stead
quoting the console: Number of devices = 1 ---------------------------------------------- #1 DefaultInput DefaultOutput Name = ASIO DIGI96 Serie Max Inputs = 16, Max Outputs = 16 Sample Rates = 44100.00, 48000.00, 88200.00, 96000.00, Native Sample Formats = paInt32, ----------------------------------------------
4 audio buffers
not using MIDI input (use 'pd -midiindev 1' to override)
using default input device number: 0
using default output device number: 0
nchan 2, flags 3, bufs 8, framesperbuf 32
as pd seems to prefer 8 buffers: " pd -asio -listdev -blocksize 32 -audiobuf 8"
now the console first says "4 audio buffers", then changes it to "bufs 16" latency (measured with the patch from the doc files) is +- 27ms - this works fine, but i'd expect about 12ms here - (32 * 16 / 44100 = .0116")?
(snipped the beginning as it stays the same..) 4 audio buffers nchan 2, flags 3, bufs 16, framesperbuf 32
another try: "pd -asio -listdev -blocksize 64 -audiobuf 4"
pd doesn't change the params here - latency 16ms (but the audio is fucked up, so i can't use this setting)
4 audio buffers
nchan 2, flags 3, bufs 4, framesperbuf 64
"pd -asio -listdev -blocksize 128 -audiobuf 4"
even weirder - pd sets bufs to 1, than says it's less than recommended..
4 audio buffers
nchan 2, flags 3, bufs 1, framesperbuf 128
warning: number of buffers 1 less than recommended minimum 2
(tried some other settings also, getting similar results as above..)
with 0.37 i first tried "pd -asio -listdev -blocksize 32 -audiobuf 8"
choosing media from the menu: everything seems ok. the delay (i suppose this the same as latency?) field is set to 8 ms, but the actual latency is about 25 ms
next i tried with a higher sampling rate, as according to the hammerfal docs this should allow for lower latency..: "pd -asio -r 96000 -listdev -blocksize 32 -audiobuf 8"
the displayed delay is again 8ms, the actual delay is 17.3ms (without jitter!?) but now metro works at about half speed and the osc~ freq is about two times (probably 2.18 times) too low, so somewhere something goes wrong with the samplerate.. when i set delay to 4 msec in the media window i get a latency of 12 ms. (with no -r 96000 flag this doesn't work, as the audio then is completely fucked up when i try to set delay to 4)
the 12ms seems to be the lowest i can get.. is there any chance i can get lower if i'd try it on linux with ALSA? i need to put real time effects on sharp-attack-no-sustain-instruments such as harpsichord and lute, so i really want the latency to be as low as possible..
thanx for reading this far - now i hope someone can answer my questions..
kind regards,
kristof
kristof.lauwers@logosfoundation.org
http://kristoflauwers.de.vu ____________________________________