I just upgraded my pc (my old hardware died during a move…that’s another story 
)…
I added a new MB, proc and mem. I also added a new graphics card and HD. the sytem is very stable (on everything else) and not overclocked or anything.
I am using the following:
abit nf7-s MB
AMD sempron 2400+
1gb ram
WD 80gb 7200rpm ide drive (main HD)
maxtor 40gb 7200rpm drive (2nd HD)
running: xp pro, sp2
I’m using 3.3 and the onboard sound on the nf7-s for now.
I’m having a problem with recorded tracks. I always add in a wav exported from fruity loops for drums, add bass, then guitars and then vocals…
what is happening: the drum wav track and bass track are fine. when I record the 1st guitar, everything is fine until I play it back…it gradually gets out of sync as the track plays…I noticed this adding a 2nd guitar just now. in a 5 1/2 min. song it’s all out of wack by 2min.
all of my drivers are up to date…there is nothing running in the background when I record.
I have the follwoing settings to choose from in audio devices:
preferences, audio devices (playback and recording):
MME: NVIDIA® nForce Audio
1-MME-Wave Mapper
NVIDIA ASIO
WDM: NVIDIA® nForce Audio
right now I have the defaults loaded:
1-MME-Wave Mapper for playback and recording. I tried the ASIO but got a warning that multiple I/O devices were selected…I ignored that and tried to play back the file…it was a mess (jarbled sounding).
I have the buffering set to high (default), sampling freq. at 44.1, normal program priority…everything else default.
any ideas? I know this sound card isn’t the best but surely I can record 8-10 tracks.
Thank you,
Chris
Hmm, I’ve seen this behaviour on some creative cards - they natively run at 48khz sampling frequency, so if you record in 44.1 it forces the card to convert on the fly, which it doesn’t handle so well.
Try recording at 48khz and see if the lag goes away. The pain point here is that you’ll have to convert mixdowns, etc when you want to burn to cd.
Also, I would try using the WDM drivers, they tend to work the best for me.
thanks for the quick reply…
I just double-checked the soundcard features (sisoft sandra and the manual) and the sampling rate is 44.1
i’ll give the wdm’s a go and report back.
thanks again.
Chris
I just tried the wdm for input and output…I got a message saying “multiple i/os selected…”, plus an error opening input devices…set it back to mme and it’s back to what it was before…
thanks for the try…
Check the n-Tracks preferences for which clock is being used. It’s under the menu Files/Settings/Preference/Options. Look for Use system time for and uncheck both Playback time and Recording time. n-Tracks defaults to using the System’s time (both options checked). That might not help, but making the switch has fixed a lot of things on some folks machines. It’s definitely worth toggling them no matter which way they are set now.
I had (and still sometimes have) the same problem. What I did, and this probably won’t work for everyone, is:
D/L and use EnditAll
drop your buffer levels till you get clicks/dropouts while recording—then edge them up till the dropouts go away.
Do as phoo said.
Set the priority level in the ‘preferences’ box to ‘High’ (not highest)
Don’t worry too much about the ‘multiple i/os selected’, I just click on ‘no’ and keep goin…you want WDM or asio, tho
Read the user manual again–it took me about four trips through to get dialed in, but I’m not too bright…
By the way, my puter is a p3 850mhz…you shouldn’t be having these problems.
Take everything I’ve said with a large grain of salt. This stuff worked for me, it may not work for you.
Peace,
-Ed
I’ve used the nforce audio with some success, just adding some vocals that needed to be done in a hurry (nforce2 ASUS a7N8x).
I got some sort of clock adjustment notice the first time I did it, and it was fine - I didn’t even pay attention to it, just clicked ok.
My setting are:
44.1khz
Recording: MME nvidia nforce audio (insert TM as required)
Playback: Same.
High buffering - 2 buffers@960 samples for recording,
6buffers@3584 for playback,
4 preload buffers,
disk loading buffers 6 times bigger
System timer is unticked
DSP thread priority = 1
And, surprisingly, I have “Keep audio devices open” ticked.
Hope that helps.
Willy.
hey guys…thanks. I’ll try unchecking playback and recording for the the “use system timer for…” setting and increase the program priority to high…see what that does.
i have the same increasing lag problem.
it seems like i’ve tried just about everything that has suggested to fix it or compensate.
my system is:
AMD Duron 700
Gigabyte 7zm (via chipset)
Windows ME
384mb ram
NVIDIA RIVA TNT2 Model 64
SoundBlaster PCI128
i’ve tried recording 4 minutes to test it, 2 tracks, and the second track is usually out by about 0.3 seconds at the end.
and wdm doesn’t work at all. comes up with: error opening wave input device etc.
thanks,
Nigel.
N-track Studio online help:
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With some soundcards new recordings don’t sound perfectly in sync with the previous tracks because of the inability of the soundcard to start the recording and the playback at the very same time. If you notice a sync problem in which recorded tracks sound constantly (i.e. with equal delay during the whole length of the recording) too early or too late with respect to existing tracks in the song you can use the compensation parameter in the Preferences/Wave devices/Advanced dialog box to compensate for the problem. Use a negative value if the new tracks sound delayed or a positive value if new tracks sound too early (the program will shift new tracks to the right by the amount specified, or if the compensation is negative the shift will be towards the left). The default value of 0 will work in almost all systems. Usually there’s no need to worry about the compensation settings, so read this only if you can hear a shift between the recorded tracks, If you experience sync problems try unchecking both the Preferences/Options/Use system timer for checkboxes, then set the Preferences/Recording settings/Hide lag indicator… parameter to 1. During a recording, notice the value that will appear near the “lag” writing on the time window on the program’s toolbar. If the lag value is constant throughout the duration of the recording it can be compensated with the compensation setting discussed above. If the lag value grows constantly during the recording the problem is most likely caused by the soundcard sampling frequency not being perfectly constant or by a difference in the soundcard’s recording and playback sampling frequencies. When this occurs the problem can’t be fixed using the compensation parameter and the program may not work properly so it may be necessary to change soundcard. Note: the lag values reported are based on the recording and playback position information reported to the program by the soundcard. This information may be incorrect so it may lead to think that sync problems exist while it’s just that the recording and/or playback times are being reported incorrectly. It’s very important to double check with your ears if the lag values reported make sense. |
My point being a bad soundcard: playback and recording won’t go in sync.
jps, have you run it at 48k sample rate?
i suppose it could be a bad soundcard, but this would be a defect, the sb pci 128 is listed as useable.
http://www.fasoft.com/soundcards.html
yep, i’ve tried it at 48k, and that was a little closer than 44.1
i’ve tried changing buffering settings, system timer, program priority, dsp thread priority, compensate plugins latency, some of the soundcard’s settings like sample rate conversion quality, i’ve reinstalled the latest drivers.
i should add the cpu usage never went above 5%.
Buffering affects hiccups and latency, but not the kind of sync problem you have. Program and DSP thread priority affect hiccups and sometimes workee/no-workee. (DSP prio isn’t meaningful unless you’re using plugins, and you shouldn’t use any during recording, at least until you have this figured out.)
Compensate Plugins latency should be set on and left alone, unless you have a plugin that lies and causes problems. Again, not a factor here.
“keep devices open” can have odd effects, because it can cause driver bugs to appear or disappear. So, it can affect almost anything.
So it looks like you did your homework, but you might make sure you try 48kHz and all 4 combinations of “system timer” checkboxes (with wave timer for both being best for most systems). The 48k thing is so often the cause of this problem, be sure to exhaust all possibilities there. That’s with the SBLive card; I don’t know how similar yours is to that.
Watching the “lag” indicator might help, but you already convinced me you have a gradual lag creep problem and not a “compensation”-fixable problem.
Just got me a M-audio firewire 1814. All my problems with sync, lag, buffer sizes etc. disappeared.
Repeat after me: No more crappy soundcards.
First impressions: easy to install, nice to work with, sounds good (no hiss nor hum). I’ve yet to test the mic preamps.
OK, no crappy soundcards!
Well, I’ve made good tracks with a builtin soundcard. It’s worthwhile to try, but no point going nuts trying to make a bad one work. So, “cheap” isn’t necessarily “crappy”.
I prefer my MOTU 828 now in any case. Neither cheap nor crappy.
using the wave timer seemed to help…but I don’t have confidence in the stability…and i have a demo project I have to get finished…
So I just bit the bullet and ordered a delta 44. I had been putting off the purchase of a good recording soundcard…I was just being cheap I guess . I’m excited to finally step up my rig a bit.
thanks for the help everyone.
oh…
OT: anyone have a sugestion for a decent stereo amp…75w per side at 4ohms?
thanks
I bet you don’t regret it