Sudden increase in audio interface latency in Live...
Sudden increase in audio interface latency in Live...
Hi everybody - I'm wondering if anyone else is having or has had this issue, or if there already exists some info on it (I've already tried searching a bit).
My current setup: MacBook Intel Core 2 Duo, 2GHz, 1GB RAM, OS X 10.4.10; Presonus Inspire 1394 (firewire); Live 6.0.7
So my problem is sometimes there's a sudden increase in latency. When I first open Live, the latency is normal (not much at all), but then sometimes it increases suddenly, to where there's a significant latency. If I go into Live's preferences and change to No Audio, and then back (to CoreAudio), it works fine again. But, I don't know what could be causing this. Thanks for the input.
My current setup: MacBook Intel Core 2 Duo, 2GHz, 1GB RAM, OS X 10.4.10; Presonus Inspire 1394 (firewire); Live 6.0.7
So my problem is sometimes there's a sudden increase in latency. When I first open Live, the latency is normal (not much at all), but then sometimes it increases suddenly, to where there's a significant latency. If I go into Live's preferences and change to No Audio, and then back (to CoreAudio), it works fine again. But, I don't know what could be causing this. Thanks for the input.
I am experiencing the same or at least very similar problem. I can record one audio track and it turns out fine. But when I go to record a second it starts getting super laggy. It even messes with the metronome. After a second audio track is recorded the metronome no longer matches up with the song.
My guess is that it has to do with memory, either system, soundcard, or virtual in Live. Have you updated your sound card drivers? I haven't but I'm about to and will let you know if it fixes anything. Such a frustrating problem...
My guess is that it has to do with memory, either system, soundcard, or virtual in Live. Have you updated your sound card drivers? I haven't but I'm about to and will let you know if it fixes anything. Such a frustrating problem...
Alright I was looking around some more (new drivers didn't help) and I changed my audio input device to ASIO and it seems to help alot. Try this and see if it helps you as well. It was kind of tricky to set up for me so if you have any questions post em and I'll try and help you
Right now my settings in the audio tab are
ASIO
48000 In/Out Sampel Rate, standard conversion rate
1024 Buffer Size
21.3 input and output latency
-43ms driver compensation for a total of -.33ms
Right now my settings in the audio tab are
ASIO
48000 In/Out Sampel Rate, standard conversion rate
1024 Buffer Size
21.3 input and output latency
-43ms driver compensation for a total of -.33ms
Hey there, thanks for the reply and suggestions. Well, I can't really change the drivers because I'm using a Mac and it just uses a class compliant driver.
Anyway, something you mentioned could also be it - driver error compensation. I recently started putting in a driver error compensation value, when before I just kept it at the default of 0.00 ms. I'm going to set mine back to zero and see if that clears it up...
Anyway, something you mentioned could also be it - driver error compensation. I recently started putting in a driver error compensation value, when before I just kept it at the default of 0.00 ms. I'm going to set mine back to zero and see if that clears it up...
I'm having similar problems with Ableton. My setup is: Live 6 on an AMD Laptop. Clock speed - 2GHz. RAM - 1GB. Soundcard - MOTU Traveler (Firewire).
At a certain point in the session, when there are a few plug-ins and tracks, each time I hit play, I have to wait for a few seconds of coughing and spluttering before the audio output is smooth. Sometimes even in the middle of playback, the CPU usage can suddenly jump from 35% to 140% and cause the engine to start coughing again for 5-10 seconds.
Changing audio buffer size hasn't helped. (And I need relatively low latency if I'm recording with many tracks running alongside.)
Is there a solution? Am I going wrong somewhere?
At a certain point in the session, when there are a few plug-ins and tracks, each time I hit play, I have to wait for a few seconds of coughing and spluttering before the audio output is smooth. Sometimes even in the middle of playback, the CPU usage can suddenly jump from 35% to 140% and cause the engine to start coughing again for 5-10 seconds.
Changing audio buffer size hasn't helped. (And I need relatively low latency if I'm recording with many tracks running alongside.)
Is there a solution? Am I going wrong somewhere?
MacBook Pro 15", Mac OSX 10.5.5, Logic Studio 8, Ableton Live 8.1.5, MOTU Traveler, Dynaudio BM6A, novation ZeRO SL MkII, novation Launchpad, M-Audio Ozonic, M-Audio MK449C
What sample rate are you using, 44100kHz?
Try setting Buffer Size (under Latency on the Audio tab) to 512 or so (mine works fine at 256). Then, set Plug-In Buffer Size (on the CPU tab) to 128 or so instead of "As Audio Buffer" (mine's set at 64)...
Also, be sure of course that you've specified your firewire interface as the device for both Audio Input Device and Audio Output Device on the Audio tab, and that you're using an ASIO driver (which I'm sure you're doing both of these things). Also, it might help to disable any inputs and outputs you don't use by clicking the Input Config and Output Config buttons on the Audio tab...I don't know, just some suggestions for now, good luck.
Try setting Buffer Size (under Latency on the Audio tab) to 512 or so (mine works fine at 256). Then, set Plug-In Buffer Size (on the CPU tab) to 128 or so instead of "As Audio Buffer" (mine's set at 64)...
Also, be sure of course that you've specified your firewire interface as the device for both Audio Input Device and Audio Output Device on the Audio tab, and that you're using an ASIO driver (which I'm sure you're doing both of these things). Also, it might help to disable any inputs and outputs you don't use by clicking the Input Config and Output Config buttons on the Audio tab...I don't know, just some suggestions for now, good luck.
Thanks. Yes, sure I'm already following these points.
I've tried different buffer settings. Everything from 512 to 2048! It didn't change the situation; only increased the latency. Though my Plug-In Buffer size setting was at "As Audio Buffer". (However, with a large audio buffer, even this should correspond in change.)
I wonder if this is normal or something incorrect about my setup. And therefore do I have to do something like - format my computer (which is probably overdue now). Or what?
Thanks.
But if all these precautions for load management are taken, shouldn't it allow me to run a session comfortably even at 24/96 (or 192)?dn83 wrote: Also, be sure of course that you've specified your firewire interface as the device for both Audio Input Device and Audio Output Device on the Audio tab, and that you're using an ASIO driver (which I'm sure you're doing both of these things). Also, it might help to disable any inputs and outputs you don't use by clicking the Input Config and Output Config buttons on the Audio tab...
I've tried different buffer settings. Everything from 512 to 2048! It didn't change the situation; only increased the latency. Though my Plug-In Buffer size setting was at "As Audio Buffer". (However, with a large audio buffer, even this should correspond in change.)
I wonder if this is normal or something incorrect about my setup. And therefore do I have to do something like - format my computer (which is probably overdue now). Or what?
Thanks.
MacBook Pro 15", Mac OSX 10.5.5, Logic Studio 8, Ableton Live 8.1.5, MOTU Traveler, Dynaudio BM6A, novation ZeRO SL MkII, novation Launchpad, M-Audio Ozonic, M-Audio MK449C
Hello again - Yes, well you might in fact need to do a back up, format, and restore proccess. However, that most likely won't help unless your system is REALLY bogged down with a lot of junk.
I would add more RAM if possible, max it out even. But your laptop model probably only supports RAM up to 1GB or 2GB I'm guessing.
Yes, you can record/produce at up to 24/32 bit @ 96/192 kHz in Live, but this is very hard on the computer to process and rather unnecessary. I used to want to record at high sample rates too, but after reading a lot of articles about how after like, 88 kHz it doesn't make much difference, I've stopped. It causes a ton of constant sample conversion by the computer, which it wouldn't have to do if you were to just set your sample rate at a lower, more standard one like 44.1 or 48 kHz. Also, files recorded at 96/192 kHz 24 bit are HUGE - so using these sample rates will also cause you to run out of hard-drive space faster.
I have my set up set at 44.1kHz 24 bit, which is what a lot of people use. Many argue that this is the best balance of quality, compatibility, hard-drive space saving and CPU performance. Really, unless you're going to save and resave the recorded file over and over (which is what degrades the sound because of file compression), then 44.1 or 48 kHz is all you need. Again, especially if you're just going to keep the recorded sound file in the Live set and only resample once when you Render to Disk. Plus, again, you will gain better computer performance in Live and will save disk space.
I would add more RAM if possible, max it out even. But your laptop model probably only supports RAM up to 1GB or 2GB I'm guessing.
Yes, you can record/produce at up to 24/32 bit @ 96/192 kHz in Live, but this is very hard on the computer to process and rather unnecessary. I used to want to record at high sample rates too, but after reading a lot of articles about how after like, 88 kHz it doesn't make much difference, I've stopped. It causes a ton of constant sample conversion by the computer, which it wouldn't have to do if you were to just set your sample rate at a lower, more standard one like 44.1 or 48 kHz. Also, files recorded at 96/192 kHz 24 bit are HUGE - so using these sample rates will also cause you to run out of hard-drive space faster.
I have my set up set at 44.1kHz 24 bit, which is what a lot of people use. Many argue that this is the best balance of quality, compatibility, hard-drive space saving and CPU performance. Really, unless you're going to save and resave the recorded file over and over (which is what degrades the sound because of file compression), then 44.1 or 48 kHz is all you need. Again, especially if you're just going to keep the recorded sound file in the Live set and only resample once when you Render to Disk. Plus, again, you will gain better computer performance in Live and will save disk space.
PS - Yeah, anything above about 512 for buffers will cause a significant, audible latency. What I meant was, your computer should be able to support setting the buffer size to between 256 and 512 and the plug-in buffer to either 64 or 128. This has worked the best for me. Using the test tone, I verified that my computer can handle 256 for the main audio buffer and, setting the plug-in buffer to 64 guarantees that any plug-ins used will not have an audible latency...
I have the same latency problem with Live 6 on my Macbook Pro with 2 gigs of ram whil using either Kore AI or the Macbooks Built in output (which actually has BETTER latency than the Kore AI, pfff).
Mostly noticable on playback and stop. There is a VERY noticable lag between hitting spacebar and playback start. And When hitting stop there's always a little lag and then a sound stutter as it stops.
I've tried all the different Audio Buffer latency settings, although havn't messed with driver error compensation OR audioplugin buffer settings (set as the default "As Audio Buffer"), because I'm not very familiar with either.
The Lag becomes even more noticable when I start adding more plugins to a project even though my cpu meter is still low (under 30% usually), which is wierd because on my PC's the latency never changes. Even with the cpu meter at Very high levels there will only be audio glitches, but no latency or play/stop Lag.
Maybe a difference the way Live is programmed between Mac and Pc?
Mostly noticable on playback and stop. There is a VERY noticable lag between hitting spacebar and playback start. And When hitting stop there's always a little lag and then a sound stutter as it stops.
I've tried all the different Audio Buffer latency settings, although havn't messed with driver error compensation OR audioplugin buffer settings (set as the default "As Audio Buffer"), because I'm not very familiar with either.
The Lag becomes even more noticable when I start adding more plugins to a project even though my cpu meter is still low (under 30% usually), which is wierd because on my PC's the latency never changes. Even with the cpu meter at Very high levels there will only be audio glitches, but no latency or play/stop Lag.
Maybe a difference the way Live is programmed between Mac and Pc?
audiovoid.net
.............
Macbook pro,
MacPro 8 core
running OSX, &
WinXP through Bootcamp and Fusion
.............
Macbook pro,
MacPro 8 core
running OSX, &
WinXP through Bootcamp and Fusion
Hmm...well for me, my latency is great, only sometimes it suddenly increases, seemingly mostly when I leave my desk for a bit, leaving Live open and the computer idle, then come back to it, and find that the latency is increased.
But again, if I restart Live or change my audio device in preference to No Audio, and then back again, it resets and is fine again.
But again, if I restart Live or change my audio device in preference to No Audio, and then back again, it resets and is fine again.
I guess I'm in the same boat. I'm fine at 512-ish buffer until I sync Live to an external sequencer. Then I have to bump it up to about 1,000 just to be able to play two audio clips at once without the sound crackling. Also, stuttering artifacts are introduced with some of the sound input on my Firewire, and that's with no clips playing! All this only when external sync is enabled. I do have the latest driver update too.
Not to brag, but I'm running a 1.42 GHz PowerPC G4 with 768 MB.
Not to brag, but I'm running a 1.42 GHz PowerPC G4 with 768 MB.
"It's more fun to compute". Yeah, when you know what you're doing!
Hm I see, well part of your problem is your computer I'm afraid...especially if you're running Live 6. The G4 processors are quite old by now (they first came out in 1999). There have already been G5's and now Apple is no longer using the Motorola PowerPC processor architecture (as I'm sure you probably know).
So, I'd say get an more up-to-date computer but if you can't, then add the maximum amount of RAM, if you haven't already.
So, I'd say get an more up-to-date computer but if you can't, then add the maximum amount of RAM, if you haven't already.
I was afraid of a response like that. Still no need for a computer yet tho, since this only happens when ext sync is activated. The more important thing is that the sync is constantly accurate. Besides, it's unlikely that my fellow producing colleages will be listening to me perform. Instead these people will be like, "Dude! How does he make certain drum hits stutter like that in perfect time?!"
I'm afraid you're right tho, and that's the reason I haven't upgraded to Live 6 yet. I won't do it until I get a new machine. Next year when my warranty runs out. "If it ain't that badly broken, then don't fix it."
And no, I was not aware that Apple changed its processing architecture. Thanks for the input!
I'm afraid you're right tho, and that's the reason I haven't upgraded to Live 6 yet. I won't do it until I get a new machine. Next year when my warranty runs out. "If it ain't that badly broken, then don't fix it."
And no, I was not aware that Apple changed its processing architecture. Thanks for the input!
"It's more fun to compute". Yeah, when you know what you're doing!