Currently Ableton handles Audio Latency differently to all other DAWS when software monitoring is turned on.

Ableton always places recorded Audio late on the timeline when software monitoring. Other DAW's put the audio exactly in time.

There is a big difference between latency error due to bad coding and the philosophical discussion of how latency should be handled. This discussion is one of the latter. To make it easier for people to understand, let us forget about the computer and all the possible delay-based timing errors, Instead let us focus on this....

"Consider you are a guitarist and you are standing right beside a drummer. The two of you are playing together perfectly in time. Your guitar amp is 15 feet across the room. "


This real world scenario is exactly the same as what happens whilst recording with software monitoring turned on.

The drummer is analogous to the click track/pre-recorded track leaving Ableton.
The Guitarist (you) is analogous to you playing along to a click track whilst recording.
The amp 15 feet across the room is analogous to the 15 ms or so of latency you would typically experience whilst software monitoring.


So do you-as Ableton engineers would suggest- play 15 ms early in order to compensate for the 15 ms it takes for the sound to leave the amp and travel across the room?

No- you will play along with the drums- if you are a 'perfect' guitarist playing along with a 'perfect' drummer you will strum the guitar at the exact same moment the drums are hit. You do NOT play 15 ms early.

The Ableton engineers suggest that you do indeed play 15ms early. They also suggest that you can repeatedly measure 15 ms in your head- and if you do not- Ableton will kindly put your audio late!

However, if you are capable of repeatedly playing 15ms early you are ok!

I understand this philosophy alright, there is a kind of logic to it. A logic that states that we humans are capable of our own delay compensation. However It fails to make any musical sense. But more importantly, there is no evidence to suggest we are capable of such precise measurements. There is no scientific evidence to suggest that anyone can play exactly in time, so why would there be any evidence to suggest that we can play exactly 15 ms ahead of time!??!

First, remember that it's above all software for live performance.

If you play a software instrument by pushing on the notes of your keyboard exactly in time with the drum track, then the audience will hear you playing 15 ms behind the drum track. In a sense, when you're playing live, you have no choice but to play "ahead of" the latency.

Think about it this way. Suppose someone built a guitar for you that has a 15ms latency. No computer involved, just a physical guitar that puts out a sound 15ms after you touch the string. You want to use it to play in a band. What do you do? You must play 15ms early, no choice.

When recording rather than playing live, the question is what do you when you press PLAY after recording a MIDI track? Do you want to hear it exactly as it sounded while you were recording it? This is what Live gives you. This is what I always want. I hate hitting play and hearing things placed differently in time than they were while I was recording. My hands have long since gotten to the point where they compensate for latency and I place notes using my ears and not my hands.

Anyway, there is a menu option in Live to give you the 'other' behavior, which gives different playback results from what you hear while recording, like many other DAWs: Options/Reduced Latency While Monitoring".

hth,

-synthoid

I apologize for being confused.

Are we talking about recording audio or tracking a virtual instrument into MIDI?

If we're talking about recording audio, are you saying that audio is being placed on the timeline 15 ms behind where it should be? In other words, if you were to do a so-called ping loopback test by playing an existing test tone from a track out the analog out and record it back into the analog in [being careful not to create a feeback loop with your cue monitor], will the copy show up 15 ms behind the original test tone?*

And, if the latter is the case, is there a difference in that behavior if you have software monitoring turned off? (Like, say you're monitoring your analog inputs from the interface's own direct cue monitor.)


* This can happen when an interface driver misreports conversion latency or a DAW doesn't compensate for the latency correctly. It was a widespread problem some years ago and still seems to crop up. (I have to use a 'tracking alignment' offset in my own DAW to adjust tracks forward on the timeline by 355 samples. Before 5 or 6 years ago it didn't have that and I had to nudge all my new tracks by that distance.

I understand what you are saying and this is what I'm on about regarding philosophy...... you prefer to guess [because it is nothing more than a guess] and play 15ms before the beat. I prefer to play as close as i can relative to the beat. That is the way I learned to play piano. That is the way I learned guitars, drums etc etc. That is the way most musicians play in the real world.......
Even if musicians do encounter latency (consider the example of the guitarist with his amp across the room.) they will still play to the pulse NOT 15 ms before the pulse. Does that make sense?

Consider you got a 4 beat count in to a song? Do you go 1,2,3, 4 MINUS 15ms .....play? Impossible. That is why it is only guesswork. That is why the implementation in Abe is (i believe) wrong.







Thats exactly like what i describe above....... A good musician will strum the strings or hit the drum in time to the beat.......They are not thinking abourt playing 15ms early. They cannot measure that. Their internal musical rythmn is locked to the beat.....not 15ms beforehand.

Its like asking someone to dance to the music but you must dance exactly 1 second before the beat.

The very last thing that dancer wants to do is dance 1 second before the beat. The ONLY thing that dancer wants to do and probably can do is dance on the beat!



Forgive me but I find this really hard to believe that anyone can hit notes 15 ms before a beat.? I mean think about it, no-one can play perfectly and robotically on the beat (otherwise there'd be no need for quantisation!) so how can you expect me to believe that you can somehow play perfectly 15ms before the beat? Your answer will of course be that you are approximating, well if you are approximating why not approximate to something a little more accurate and musically more natural?

Granted a good drummer will be able to sit back in the pocket or push the groove to add axcitement or whatever, but grooves are different to a constant 15ms before a beat otherwise it wouldnt be a groove but just a robot playing 15ms too early!!

[QUOTE=theblue1;8307517]


Yes 15ms late.

Yes 15 ms late BUT ONLY IF YOU HAVE SOFTWARE MONITORING ENABLED

When disabled, audio is placed correctly.......try it for yourself.

Heres another funny thing, do the loopback and put a -15 ms into the driver delay compensation...... I thought this would sort said lateness but no- apparently the driver compensation is only for unmonitored tracks. ie tracks where software monitor is off!?!?


Ahh ur train of thought was so like me today! YES See above


Ha! oh I remember the days! No my interface reports its' correct latency.

First, 15ms is a ridiculous long latency. I have used Live on stage with a laptop and a not especially new audio interface, with a latency as reported by Live of just over 3ms. 5ms is plenty enough buffer depth for any decent audio interface these days, for live use. The copy of Live I have open right now reports a round-trip latency of 4.98ms.

Consider also that a keyboard like the Fantom has 3ms of latency. From MIDI in to audio out: 3ms. For a system like Kyma, which is also designed for Live use, a typical latency figure is 5ms.

You seem to be missing a very basic point. If you use Mainstage; or ProTools; or Cubase; or any host whatsoever; there will be a latency between MIDI in and audio out. You may not like it or believe that a musician can compensate for it, but it will be there anyway. That's the way computers work. They take time (not much, but some) to compute the audio from the MIDI.

So I would turn the question around and ask: when you play a hosted software instrument on the stage with a live band, what do you do to get the sound and your playing to be perfectly aligned? I think you are saying that your sound is always late with respect to your band members. Maybe so, and if it's only a few ms that's fine, no one will notice.

As for me, I have played software instruments with quite large latency and have become able to line up the audio unconsciously -- it just happens. I don't count 1 2 3 4 minus Xms, I just play the instrument and I can line the sound up, if the latency is not ridiculous to the point of being like a slapback. If I try *not* to do that, and play the keyboard exactly in time with the audio coming out audibly late, I can't do it. I compensate whether I want to or not. Maybe it's not the same for other keyboard players, I don't know.

-synthoid

[QUOTE=synthoid;8307690] when you play a hosted software instrument on the stage with a live band, what do you do to get the sound and your playing to be perfectly aligned?

a) play in time
b) low latency

Abletons' philosophy seems to be

a) Guess.......


I have no interest in becoming 'good' at guessing...... that is why all other DAWs don't implement this guessing game, that is why I started this topic, that is why we as musicians spend so much time tweaking our computers to reduce this guesswork.

Ableton too no longer needs this system of guesswork. Think about it how can you accurately play before a beat when you cannot accurately play ON the beat?

How can a dancer dance 15ms, 3ms, 1ms, .00005ms, 2.5 hours 2.5 minutes before the beat???? Why not just let him dance on the beat?

.....very impassioned plea there....really I just want the option to turn it off

I mentioned it above, but here it is from Live's release notes:

8.2.7 Release Notes

Improvements and feature changes:

There is a new entry in the Options menu: "Reduced Latency When Monitoring", which toggles latency compensation on and off for tracks which have input monitoring on. When enabled, input-monitored tracks will have the lowest possible latency but may be out of sync with some other tracks in your Set. When disabled, all tracks will be in sync, but input-monitored tracks may have higher latency. Before, the option was only available via the 'StrictDelayCompensation' options.txt, and was ON per default. It now defaults to OFF.

pretty strange, never had any problems with latency and recording in sync with Live... maybe because i always record live, so use minimal IO buffer size, there's only 7ms of latency, which is not a problem at all.

When i record live bands at studio - there's always a natural delay/latency, just because of nature of sound - musicians located in a studio cannot listen everybody without latency - there's always a distance between them, it was never a problem for live bands to play nice in time.

Hey Reduced Latency while Monitoring when "on" disables the plugin delay compensation for all tracks that are record-ready OR have the "Monitor" switch set to "IN"

ie it only effects PDC which is NOT the issue.


For anyone reading this, I have yet to find a neat solution.......
However there are the obvious workarounds 1-being record whilst direct monitoring through a desk or sound card (abletons' timing is solid [for me at least] this way)
OR You can use the track delay to put a negative delay in ms to the value of ur reported audio round trip latency as displayed in Abe's Preferences.

edit "driver error offset" in preferneces would seem the ideal candidate to fix, but alas, it only applies when tracks are NOT being software monitored.

Abletons' engineers are really hell bent on recording your latency. It is a philosophy which-the more I think about- the less I agree with.

Maybe I'm misunderstanding, but something seems wrong here, not just a question of "philosophy".

I'm wondering if the driver is not reporting latency correctly to Live.

Does the Driver Error Compensation setting help?

There was a very very long thread on the Ableton forum about Live's weird handling of latency:
https://forum.ableton.com/viewtopic.php?t=52985

I eventually sold Live because of this issue. It's a great program if you want to automatically quantize MIDI input, but if you want to play with feel, it does weird things to the timing.

Whether this has or will be addressed since the days of that thread, I'm not sure - but i gave up waiting.

Hi iain I too left ableton around 6 or so but have just come back. Currently testing to see what is broken and what is working!

Apart from the obvious Audio is tight when hardware monitoring.
MIDI on brief initial test seems tight but do not quote me yet!


Really looking forward to trying Bitwig.... looks promising

Ah, now I have a clear picture of what's happening... unfortunately, it mystifies me. heh (I'd tinker with it myself, if I was an Ableton user.)

I'm going to catch up with recent entries (above) and will probably be lurking this thread, though, because it seems like a really interesting issue. If you don't hear from me directly, good luck on this! (And if I have any sudden bright ideas about it all, I'll for sure pop back in.)


PS... Took a look at that thread in the Ableton Forum... if they haven't 'fixed' the issue in 8 years, I'm guessing the philosophy explanation is as good as any (but none I can think of make much sense). Weird.

I have another theory, scanning through the version history, with each new update there are quite a high number of entries concerning bug fixes for things like incorrect latency and generally things pertaining to latency.

Perhaps they are locked in by code bloat and/or poor foresight. Perhaps the audio engine needs a complete re-write to fix this problem. I dunno just thinking out loud. It seems crazy that it is a 'philosophy' although it seems impolite to call the engineers idiots also.

If it is a philosophy, it is wearing pretty thin judging by that thread!

No, driver is definitely reporting correct latency. That was the first thing I checked with a loopback test....

Re: driver error compensation...
for some unexplained reason, (it is literally unexplained in the manual I checked!) Driver error compensation only concerns itself with software monitoring turned off....

Turn on Software monitoring and 'driver error compensation' has no effect.

Perhaps their 'philosophy' is to try to avoid going into the audio engine code base if at all possible...

I thought that the OP was initially talking about MIDI. I gather he is talking about audio. My bad. Forget what I wrote above about software instruments, live play, etc. I don't know why he didn't point out this obvious problem with my response, but whatever.

So, I set Live to have 256 samples of buffer depth, put some audio on a track, play it out of Live and physically loop it back to an audio input. I record it with and without monitoring turned on for the record track. Exactly the same result in both cases: the audio tracks are aligned to within a couple samples.

Can the OP describe a simple test case that can be recreated by others that illustrates this problem?

-synthoid

ok, I went one step farther with my tests.

I created an audio track for playback. Routed it to the master out -- it is the "backing track".

I then looped this back to an audio input; this is the "instrument being played".

I armed an input track to capture it. This is the "recording of the instrument being played".

I turned on input monitoring for this input track. I routed the output to a different physical output on my interface.

I looped back this track, and recorded it on a different track in Live. This is a recording of the live monitoring of the recorded signal.

All of the recording tracks and the original are more or less perfectly in alignment. This doesn't seem to be affected by the latency in my interface.

I can't replicated the problem being described by the OP. I confess that I don't really understand the problem very well. I'm using an RME UFX under Mac OS X for what it's worth.

-synthoid

I thought it was a bit confusing alright. But the point I was making holds true for both MIDI and audio. That is why I answered your line of questioning. IF I hear the click track at precisely 4:30 on a Sunday afternoon..... THEN, recorded audio data AND midi data should be placed on the timeline at precisely 4:30 on a Sunday Afternoon.


Were you using clip view or arrangement view?

I have yet to do tests on arrangement view recording, but I guess/hope that Abe's engineers would be in line with the other DAWs and nudge the audio in time because Arrangement view is 'non real time critical' if that makes sense? ie things can be neatly changed after the fact in arrangement view.

Here is how to do it. I suggest everyone try it now if you can.

In audio preferences set buffer size to 512 or so.

Take a clicky type sample with a sharp transient.
Line 4 of them up in a 1 bar loop exactly on each beat.
Import to a free slot in session view
(if you prefer you can also do this by recording the output of simpler or impulse)

Output this track to a physical output on ur interface
.
Physically wire from this output to a free input on ur interface
Software monitor this input in Abe on a separate track.
Press play

Record for a bar

Look at the newly recorded clip does it match up with the original... are the transients still on the beat? If no, Ableton is putting your audio in the wrong position.

Listen to the two together. (you'll need to change the original clips output to master OR just duplicate it onto another track) Is there a delay? If yes Ableton is putting your audio in the wrong place.

I suspect if everyone follows these instructions, there will be a delay. If there is you should find find out how long that delay is in ms (Ableton cannot measure in samples) .........

Turn on PDC (if not already on)
Show track delay
Enter a minus value to the tune of whatever your round trip latency is (as reported by ableton) And you should be in the ballpark give or take a few ms to account for unreported latencies or undetected latencies.

Give it a go.

Well, ok, but in the case of MIDI there is a real latency due to the time necessary to turn MIDI into audio. Sorry about that, but it has nothing to do with Ableton Live. It is true of digital keyboards, software instruments, Logic, Pro Tools, you name it: welcome to software instruments. If you play the note at 4:30 then the sound will come out at 4:31 in the real world. Again, sorry for the impurity of it all, but that's physics and the real world.


hmm. This doesn't improve the credibility of your argument I'm afraid. There is not difference in the latency handling of arrangement and session view, but it's much easier to see what's going on in arrangement view.

I set buffer depth to 512. Put a drum loop with sharp transients into a clip slot. I record the output via loopback into a new clip slot. All of this in session view. The offset between the original and the recorded track is about 0.5 ms (1/2 millisecond).

-synthoid

I'm sorry synthoid but with all due respect you didn't even know what I was arguing until a post ago......yet you had no problem talking with absolute authority on the subject.

You then had the cheek to blame your ignorance on ME because I politely took the time to answer your somewhat stupid line of questioning because YOU didn't bvother taking the time to understand the problem.

I am not talking in absolutes. I see a problem and I want to talk about it to hopefully fix it.
Now can anyone other than this guy please do the test above. Jeebus H Christ.

Ok so I just tested using the internal soundcard on my mac pro. Same delay s I got with my TC Konnekt 48. Can someone else test as per this methodology.... would really appreciate. Thanks


In audio preferences set buffer size to 512 or so.

Take a clicky type sample with a sharp transient.
Line 4 of them up in a 1 bar loop exactly on each beat.
Import to a free slot in session view
(if you prefer you can also do this by recording the output of simpler or impulse)

Output this track to a physical output on ur interface
.
Physically wire from this output to a free input on ur interface
Software monitor this input in Abe on a separate track.
Press play

Record for a bar

Look at the newly recorded clip does it match up with the original... are the transients still on the beat? If no, Ableton is putting your audio in the wrong position.

Listen to the two together. (you'll need to change the original clips output to master OR just duplicate it onto another track) Is there a delay? If yes Ableton is putting your audio in the wrong place.

I suspect if everyone follows these instructions, there will be a delay. If there is you should find find out how long that delay is in ms (Ableton cannot measure in samples) .........

Turn on PDC (if not already on)
Show track delay
Enter a minus value to the tune of whatever your round trip latency is (as reported by ableton) And you should be in the ballpark give or take a few ms to account for unreported latencies or undetected latencies.

Give it a go.