FIX latency in line with other DAWs?

[aviavi]

I want to revive this thread, because I agree with the original poster: Live should behave like other DAWs: Audio recorded on a track with Monitoring set to In or Auto should be time aligned. As it stands, it is only time-aligned (presuming you have your buffer offsets in the Preferences set correctly) when Monitoring of a record-enabled track is Off.

I think I understand why Ableton implemented this: The theory is that if you're monitoring through software (In or Auto), then you will end up playing ahead of the beat by the amount of the latency in your system, for example, 20 ms. Live then places the audio 20 ms behind where you actually played it, thinking that it will all line up musically.

(By the way, you can verify the behavior by doing a loopback test: record a click, put it on a playback track, and output that track to your audio interface. Plug in a cable connecting this output to your input, and re-record the track (WITH VOLUME DOWN ON THE INPUT TRACK to avoid hellacious feedback) Try it with Monitoring In/Auto and Off. Zoom in and you will see that the recorded tracks align differently depending on which Monitoring option is selected).

I would prefer it if Live behaved like other DAWs and time aligned tracks that are input monitored.

Live should have a setting so that one can toggle Live's current behavior on or off.

[alltomorrowsparties]

I want to revive this thread, because I agree with the original poster: Live should behave like other DAWs: Audio recorded on a track with Monitoring set to In or Auto should be time aligned. As it stands, it is only time-aligned (presuming you have your buffer offsets in the Preferences set correctly) when Monitoring of a record-enabled track is Off.

I think I understand why Ableton implemented this: The theory is that if you're monitoring through software (In or Auto), then you will end up playing ahead of the beat by the amount of the latency in your system, for example, 20 ms. Live then places the audio 20 ms behind where you actually played it, thinking that it will all line up musically.

(By the way, you can verify the behavior by doing a loopback test: record a click, put it on a playback track, and output that track to your audio interface. Plug in a cable connecting this output to your input, and re-record the track (WITH VOLUME DOWN ON THE INPUT TRACK to avoid hellacious feedback) Try it with Monitoring In/Auto and Off. Zoom in and you will see that the recorded tracks align differently depending on which Monitoring option is selected).

I would prefer it if Live behaved like other DAWs and time aligned tracks that are input monitored.

Live should have a setting so that one can toggle Live's current behavior on or off.

Yeah not been here in a while but in response to you YES!

In response to dragonbreath, you misinterpreted or misunderstood what I was saying. The session view behaves differently because everything has to happen in real time (just to note: audio is put EXACTLY in time on the arrange page) BUT there is no reason why audio cannot be placed back in time in session view once it has been recorded.... ie after the fact. Hence, it is not physics that is preventing the Abe engineers from placing audio in time after the fact. My hunch is that they designed the session view audio engine to always place things late to better serve people on slower machines who would probably find it really annoying to have a large delay when tracking and suddenly they hear it back in time...... but this shouldn't mean that those of us with low latencies should suffer.


In response to the dude who reckons he can play 15 ms early; (consistently and repeatedly) you're wrong. No-one can do this. But even if you could, why would you play 15ms ahead of a drummer who is BESIDE you? What if your amp is 500 feet away? Are you going to play 500ms early to compensate or are you just going to play along with the drummer who's beside you?
Think about the analogy again and come back to me cos it's kinda important.

[H20nly]

In response to the dude who reckons he can play 15 ms early; (consistently and repeatedly) you're wrong. No-one can do this. But even if you could, why would you play 15ms ahead of a drummer who is BESIDE you? What if your amp is 500 feet away? Are you going to play 500ms early to compensate or are you just going to play along with the drummer who's beside you?
Think about the analogy again and come back to me cos it's kinda important.

pretty sure i'm the dude you're talking about, i was getting a read of 14.7. along the way we chalked it up to the latency not being reported correctly... it's either that or you're wrong and i can play that way. regardless of what the numbers said, i could not hear it and i believe my ears... especially since there's no logical way i could not hear it and it be there at the same time.

then there's this:

ok fwiw... i came home and decided to play with my mixer settings. i'm currently at 44.1 24 bit... 96 samples... 5.17 in and out and 10.3 ms overall. i've been recording bass for a while... after playing on my keys - i decided to use a VST to take it further, AAS String Studio. no latency, no pops, no crackles, no RME.

i only mention it because i believe this is within the range you call acceptable.

anyway...
bump

[alltomorrowsparties]

i was getting a read of 14.7. along the way we chalked it up to the latency not being reported correctly... it's either that or you're wrong and i can play that way. regardless of what the numbers said, i could not hear it and i believe my ears... especially since there's no logical way i could not hear it and it be there at the same time.

I've no idea what you're saying here.
If you're saying you can play 14.7 ms ahead of the beat, here's a really simple way of testing it.....

just record a midi clip whilst playing along with a click. Zoom in and observe. Depending on how good your timing is, you'll notice that your recorded midi file will deviate between +/- 30 to 50 ms from the grid. That'll give a total range of human error between 60 to 100ms. Wayyyyy more than your (quite specific) 14.7ms!
So if you cannot play on the beat consistently and repeatedly, how are you gonna play 14.7 ms before the beat consistently and repeatedly.
Impossible!

This is why all DAW's including Ableton's arrange view, don't expect you to do this either and will put audio where it's supposed to be.

Hence this thread.

ok fwiw... i came home and decided to play with my mixer settings. i'm currently at 44.1 24 bit... 96 samples... 5.17 in and out and 10.3 ms overall. i've been recording bass for a while... after playing on my keys - i decided to use a VST to take it further, AAS String Studio. no latency, no pops, no crackles, no RME.

i only mention it because i believe this is within the range you call acceptable.

My problem is NOT latency. I don't mind latency. My problem is audio being late on the timeline.
No range is acceptable for this.

[alltomorrowsparties]

I think I understand why Ableton implemented this: The theory is that if you're monitoring through software (In or Auto), then you will end up playing ahead of the beat by the amount of the latency in your system, for example, 20 ms. Live then places the audio 20 ms behind where you actually played it, thinking that it will all line up musically.

I used to think that too, but then I thought why wouldn't they just do that in the arrange view as well and why do all other DAW's not behave in the same manner. I think it's more for people on slower machines with higher latencies. Imagine the DJ with a 512 buffer whose re-sampling in a club...suddenly the loop plays back 512 samples "early". It'd be even worse if he was sampling decks or whatever cos that'd be 512 in and 512 out latency.....worse still if he's monitoring with latency inducing plugins and/or latency inducing plugins on other tracks with PDC turned on. Again worse still if the previously re-sampled stuff is further re-sampled as the effect would be cumulative.



Abe decided to just circumvent all that. Which is fair enough i guess, but it'd be nice to have the option to turn it off without resorting to hardware monitoring.

[Dragonbreath]

Ableton will only reallign your audio if your record with monitoring set to off

[H20nly]

i was getting a read of 14.7. along the way we chalked it up to the latency not being reported correctly... it's either that or you're wrong and i can play that way. regardless of what the numbers said, i could not hear it and i believe my ears... especially since there's no logical way i could not hear it and it be there at the same time.

I've no idea what you're saying here.
If you're saying you can play 14.7 ms ahead of the beat, here's a really simple way of testing it.....

just record a midi clip whilst playing along with a click. Zoom in and observe. Depending on how good your timing is, you'll notice that your recorded midi file will deviate between +/- 30 to 50 ms from the grid. That'll give a total range of human error between 60 to 100ms. Wayyyyy more than your (quite specific) 14.7ms!
So if you cannot play on the beat consistently and repeatedly, how are you gonna play 14.7 ms before the beat consistently and repeatedly.
Impossible!

This is why all DAW's including Ableton's arrange view, don't expect you to do this either and will put audio where it's supposed to be.

Hence this thread.

ok fwiw... i came home and decided to play with my mixer settings. i'm currently at 44.1 24 bit... 96 samples... 5.17 in and out and 10.3 ms overall. i've been recording bass for a while... after playing on my keys - i decided to use a VST to take it further, AAS String Studio. no latency, no pops, no crackles, no RME.

i only mention it because i believe this is within the range you call acceptable.

My problem is NOT latency. I don't mind latency. My problem is audio being late on the timeline.
No range is acceptable for this.

ah. i see what you're saying. sorry if i missed it before. no, i agree with you... there's no way i could play like that.

[Stromkraft]

I think I understand why Ableton implemented this

Why don't you just ask Ableton, why this is?

[Lokua Chicago]

This is very serious issue IMO. I have spent many, many hours trying to get this right.

For those struggling with this or even not understanding, please read this post:
https://www.muffwiggler.com/forum/viewt ... 314cce1dc2

^ that is the only way to record what you hear in session view and have your recording be on grid as far as I've found. It is not intuitive or documented or fool proof. Now, that solution works all fine and dandy if you're just recording external audio, but add MIDI keyboards and controllers that have to receive clocks and MIDI and send audio back to Live and it's a whole different set of problems. For the MIDI scenario I have to add negative track delays to the monitoring tracks as well as negative MIDI clock outs. Likewise not all external gear is created equal - some have their own latency - which means more negative track delays, and once you start adding negative track delays to one track, it messes up previously set delays on other tracks and you play this game of whac-a-mole. And god forbid you need to change your buffer - because all of these millisecond delays all over the place are completely derived from the overall latency, so you'll have to do it all over again (and you can only adjust one track delay at time - which is another complaint).

Anyway, the fact that I could apply that workaround gives me faith that a built in solution is possible. I'd love to not have to split monitoring and recording into two tracks as it doubles my track count (my Recording template contains 47 tracks for only 18 external channels + CV tools clock outs).

Edit: sorry that was kind of a tantrum. That was just my way of venting and saying hell yes, I agree with the OP.

Honestly, I'm having a hard time understanding why it works this way? I mean I read some theories above but I don't follow. It's not like the audio is placed late on the grid but sounds on time when you play it back. It's just late period. Why would anyone want this?