Ableton Forum

dinaiz wrote:

leedsquietman wrote:Most latency issues are down to computer performance and audio card settings anyway, rather than choice of DAW.

Very true. Actually I think he means that live 8 is more CPU hungry than Live 7, so he needs a bigger buffer, which induces more lattency. At least, that would make sense

not to mention the user. lots of complainers are just lazy.

dinaiz wrote:

leedsquietman wrote:Most latency issues are down to computer performance and audio card settings anyway, rather than choice of DAW.

Very true. Actually I think he means that live 8 is more CPU hungry than Live 7, so he needs a bigger buffer, which induces more lattency. At least, that would make sense

Actually, you're both right... but wrong as well. I think I may have it. Someone else said it somewhere (maybe earlier here) which led me to what I think is a fix. They said that Live has a buffer for plug-ins that is separate from the regular audio buffer that your audio interface uses. I saw it and it was set for 256. I seemed to think I tried to change it and nothing happened, but opening it again I saw it was still set at 256. So for fun, I set it for it's lowest value: 32. Latency issue: gone. Popping issue: there. The track I was running was a CPU hog, but with some moderate changes I got it working glitch free. Anyway, it's under CPU in your prefs. I had limited time tonight so I didn't see if I could get it acceptable using a value of 128, but I bet the answer is I could.

zerocrossing wrote:

dinaiz wrote:

leedsquietman wrote:Most latency issues are down to computer performance and audio card settings anyway, rather than choice of DAW.

Very true. Actually I think he means that live 8 is more CPU hungry than Live 7, so he needs a bigger buffer, which induces more lattency. At least, that would make sense

Actually, you're both right... but wrong as well. I think I may have it. Someone else said it somewhere (maybe earlier here) which led me to what I think is a fix. They said that Live has a buffer for plug-ins that is separate from the regular audio buffer that your audio interface uses. I saw it and it was set for 256. I seemed to think I tried to change it and nothing happened, but opening it again I saw it was still set at 256. So for fun, I set it for it's lowest value: 32. Latency issue: gone. Popping issue: there. The track I was running was a CPU hog, but with some moderate changes I got it working glitch free. Anyway, it's under CPU in your prefs. I had limited time tonight so I didn't see if I could get it acceptable using a value of 128, but I bet the answer is I could.

With a capable CPU I was able to go as down as 64 with both Line 6 TX1 and FW card. However at 64 some popping occurs on slower systems.
I definitively use 256 most of the times with occasional jumping to 128 in less complex situations. Moreover I do not perform live but I compose
Live. Owning an old Macbook and a 1 year old iMac I can say that Macbook needs 256 but can go to 64 (all track freezed otherwise hiccups and pops) when I record guitar while iMac is still comfortable at 128, but occasionally pops at 64 in complex situations. I have an hardware mixer before the audio card so latency is not an issue but sometimes when I want to use Live as an effect processor for Guitar and Bass Guitar or recording Vocals while having reverb it can be done at 128 or 256, no problems. I do not like anything lower than that because CPU is stressed a lot and we should be careful on stressing hardware nowadays. Quality is sure lower than in the past.

- Best
- Pasha

One thing with latency minimisation that gets overlooked....

The buffer samples are linked to the sample rate. The time passing for each 256 samples is
more at 48000 samples per second then it is at 96000 samples per second. This means less
latency at 256 samples running at 96kHz then 256 running at 48kHz. However much more CPU
is used as the "tape" is moving so much faster.

If you have massive CPU and want the fastest latency possible, experiment with 96kHz sample rate.

-Ben

MrYellow wrote:One thing with latency minimisation that gets overlooked....

The buffer samples are linked to the sample rate. The time passing for each 256 samples is
more at 48000 samples per second then it is at 96000 samples per second. This means less
latency at 256 samples running at 96kHz then 256 running at 48kHz. However much more CPU
is used as the "tape" is moving so much faster.

If you have massive CPU and want the fastest latency possible, experiment with 96kHz sample rate.

-Ben

Isn't it the same than having 128 samples at 48khz ?

hmmm I guess likely uses similar amount of CPU in the end.....

Still it's worth playing around with, depending on setup can likely squeeze a lil more outta your hardware.

edit: Guess it only really comes into play if you're already able to get buffers down below 128.
Bigger buffers at higher sample rate might have some efficiency gain or loss in some software

-Ben

I'd say it uses less as the sample frequency is smaller but not sure ...

Tried yesterday night again. Same big set, in Live 7 with buffer at 256 samples, works like a charm. In live 8 I have to set the buffer to at least 512 to avoid crackles when switching scenes. All other settings are exactly the same (sample rate etc.). Bummer.

What I am going to do on the show tomorrow is to use the big liveset in Live7 and during one part of the show open live8 additionally for using looper on a certain track. I did this in rehearsal now two or three times and it worked fine.

zerocrossing wrote:

dinaiz wrote:

leedsquietman wrote:Most latency issues are down to computer performance and audio card settings anyway, rather than choice of DAW.

Very true. Actually I think he means that live 8 is more CPU hungry than Live 7, so he needs a bigger buffer, which induces more lattency. At least, that would make sense

Actually, you're both right... but wrong as well. I think I may have it. Someone else said it somewhere (maybe earlier here) which led me to what I think is a fix. They said that Live has a buffer for plug-ins that is separate from the regular audio buffer that your audio interface uses. I saw it and it was set for 256. I seemed to think I tried to change it and nothing happened, but opening it again I saw it was still set at 256. So for fun, I set it for it's lowest value: 32. Latency issue: gone. Popping issue: there. The track I was running was a CPU hog, but with some moderate changes I got it working glitch free. Anyway, it's under CPU in your prefs. I had limited time tonight so I didn't see if I could get it acceptable using a value of 128, but I bet the answer is I could.

So, keeping my audio interface at 192 and Live's plug in buffer at 64 gives me a good compromise between performance and latency. No worse than Live 7, that's for sure. Note, I'm not triggering any audio clips or doing stretching/audio warp. I'm satisfied.

In live 8..i noticed more latency on my midi controller controlling some effects.
live 7 no problems

i've had a lot of latency issues as well. using live 7- everything naturally was on time, now for some reason i have to warp all of my quantized midi sequences sent to my synthesizers. total bummer as i dont usually have to use warping but now i feel like i have to warp every track to get the groove right

greenscreens wrote:i've had a lot of latency issues as well. using live 7- everything naturally was on time, now for some reason i have to warp all of my quantized midi sequences sent to my synthesizers. total bummer as i dont usually have to use warping but now i feel like i have to warp every track to get the groove right

I had the very same problem, and it diapeared when I switched my audio interface's ASIO drivers, for ASIO4ALL... No comment

i am having problems as well, also, i've noticed that if i record something with a huge buffer (lets say 1028). the recording is not being compensated for and is offset late (meaning it is tracked "behind" the music and i have to slide the waveform over). this didnt happen to me in 7.

???

so it is an actual problem and not something that someone just made up for 'fun'. apologies are in order from certain board members.

Tone Deft wrote:

to say he gets ZERO latency in Logic is just a dumb thing to write.

Perhaps he meant "ASIO direct monitoring" in which the input signal bypasses the audio buffers completely giving zero latency even when monitoring via the software?

I use it regularly when recording into Cubase.