Huge Latency diff. between 7 & 8

[DaffyDub]

In live 7 the midi track sending out to my Mopho needs -15 ms delay compensation to play in sync...
In Live 8 it needs -41ms..... that's quite a difference isn't it.... haven't tested my other synths yet but am guessing the same will be the case... anyone else notice this??


Peace Daffydub

[poshook]

Live 8 is very lazy. Much more latency everywhere... for serious music production with virtual instruments and plugins almost unusable in the middle of the project It is a pity for Live 8 and for me and my money too. I need to do everything in Logic.

I think the Abes should decided to make two versions - one for live and one for studio work. Live engine specialized to live/stage music handles with plugins and instruments much worse in therms of latency and CPU load.
When I insert brickwall limiter with internal latency to Live in the middle of the project and then I need to record any MIDI via my keyboard, bypassing that brickwall limiter to avoid its latency is not enough, I need to kick it off and save its settings as preset before!

Using Logic with the same project with the same plugins/instruments I will get no latency and less than half of CPU load...

[Tone Deft]

Live 8 is very lazy. Much more latency everywhere... for serious music production with virtual instruments and plugins almost unusable in the middle of the project It is a pity for Live 8 and for me and my money too. I need to do everything in Logic.

I think the Abes should decided to make two versions - one for live and one for studio work. Live engine specialized to live/stage music handles with plugins and instruments much worse in therms of latency and CPU load.
When I insert brickwall limiter with internal latency to Live in the middle of the project and then I need to record any MIDI via my keyboard, bypassing that brickwall limiter to avoid its latency is not enough, I need to kick it off and save its settings as preset before!

Using Logic with the same project with the same plugins/instruments I will get no latency and less than half of CPU load...

omfg shut the hell up with this idiotic rambling. wow.

DaffyDub - review your buffer settings, they may have been reset when you installed Live 8. that's what you should do anyway before starting a thread, right? give it a once over.

[nowtime]

@ Tone

http://www.metamucil.com/our-products.p ... awodKBNJbg

may help...

[Tone Deft]

@ Tone

http://www.metamucil.com/our-products.p ... awodKBNJbg

may help...

another great post n00btime.

actually I read
"should decided "

as

"decided"

bad writing, bad reading.


still
Using Logic with the same project with the same plugins/instruments I will get no latency and less than half of CPU load...
is bullshit because there's ALWAYS latency. this is a lie as is his description of bypassing a limiter to avoid its latency?? what a crock of shit.

[zerocrossing]

First of all, I'd like to get right out here that I'm an idiot. However, lately I've been noticing latency in my audio signal (guitar running though Guitar Rig or Amplitube). I thought it was my imagination. The even odder thing is I don't notice it all the time. I'm on a Toshiba laptop running at 2 gighz (core duo). I've got a MOTU 828 that normally runs fine at a 192 buffer with no noticeable latency. Perhaps we're actually experiencing something real?

[Tone Deft]

it's a big issue.

I spent time over the weekend playing with a latency checker (thanks pants_!)
http://www.thesycon.de/deu/latency_check.shtml

turns out both Live and my sound card are dirty lying WHORES, my latency is much worse than it read. so, all my rants about feeling 10mS of latency were BULLSHIT. if I had to say at this point, anything under 35mS is OK, IMO but I'm still working on understanding it all.

don't even bother reading the values in Live or from your card, use this checker.

[MrYellow]

The input/output buffers reported to ASIO are only half the picture.

The ADC-DAC conversion process also takes time which isn't reported
via the buffer size. This needs to be tested by loop-back and entered
into software to compensate for the converters as well as buffers.

Don't believe this is the problem though......

As for when it's "noticeable". Even a few ms is noticeable depending on
what instrument you're playing. I get distracted by anything over 12ms.
Each ms is basically equal to standing 1 foot away from your amp on stage,
add a few feet of real-world distance in there and the latency gets annoying.

Best bet is to use compensation and monitor your input before it goes through
the computer. This way you're hearing your live sound mixed with the compensated
output in real-time, once recorded the latency that would have existed is removed.

If you use compensation and monitor through the soundcard then you'll be playing
along and compensating manually in your head and with your playing, then once the
latency is removed by software you're left recorded as playing before you heard
yourself.


You can use Live as an effect processor in real-time with low-enough latency.
However from bitter experience I've decided that for me at least external analog
effects are a much better idea.

-Ben

[Tone Deft]

The input/output buffers reported to ASIO are only half the picture.

The ADC-DAC conversion process also takes time which isn't reported
via the buffer size. This needs to be tested by loop-back and entered
into software to compensate for the converters as well as buffers.

Don't believe this is the problem though......

it's not, that's on the order of microseconds, it's negligible. it's the bus speed on the computer, USB/firewire conversion and data rates, things like that. the low level hardware is quite fast.

As for when it's "noticeable". Even a few ms is noticeable depending on
what instrument you're playing. I get distracted by anything over 12ms.
Each ms is basically equal to standing 1 foot away from your amp on stage,
add a few feet of real-world distance in there and the latency gets annoying.

yeah, know about 1ms = 1foot. problem is, I haven't had a good readout of what I was actually using!! I put off getting a proper tool to learn what the REAL latency is. all this time I thought I was playing with 2-15mS of latency, no it was 12-48mS I was playing with!!! at least I'm learning.

Best bet is to use compensation and monitor your input before it goes through
the computer. This way you're hearing your live sound mixed with the compensated
output in real-time, once recorded the latency that would have existed is removed.

yeah, I do all my guitar processing outside Live and monitor as a blend of signals, which is pretty sloppy but it's been easy to use that way. I'm changing this, the best move IMO is to get your latency low and play through Live. that way you get what you hear. at times in the past I would monitor direct then go back and change the start marker, pretty lame but it works.

If you use compensation and monitor through the soundcard then you'll be playing
along and compensating manually in your head and with your playing, then once the
latency is removed by software you're left recorded as playing before you heard
yourself.

I can compensate in my head OK, but I'd rather not tweak my playing to appease Live, I'd rather relax and play to my fullest (half assed) ability.

You can use Live as an effect processor in real-time with low-enough latency.
However from bitter experience I've decided that for me at least external analog
effects are a much better idea.

agreed. it really really blows when you want to play guitar and you end up working on computer settings which is about as opposite to guitar playing as it gets.


in time latency won't be an issue. new standards to replace USB and firewire, faster computers, we'll look back on this like we look back on betamax video tapes.

[Michael Hatsis]

Also,
I remember hearing that Live adds your plugin buffer size per vst/au plugin to your latency. So If your using lots of third parties you'll have more latency. Maybe someone else can confirm or deny this???

[leedsquietman]

I'm sure this is correct because Live is the only DAW I know of that has a seperate plugin buffer, certainly Cubase, Protools and Logic don't have it. Protools LE/M-Powered of course is extra annoying here in that it has no automatic plugin delay compensation, so you have to manually enter these values.

TD is correct about latency checking and that utility, I've used it for ages since seeing it referred to in Sound On Sound, they use it when reviewing interfaces.
yes, the Analog to digital converters are not included in this process either, but are usually a low ms figure.

I agree with TD that for most intents and purposes, a real time combined latency of about 30 ms is actually fine for most uses (and will probably be reported more like 14ms in Live or the audio card control panel) - exceptions are tracking Live drums, and if you are some insane 10 million notes per second shredder, then you will need latencies under 10ms or it gets annoying and noticeable.

One thing to be aware of before comparing CPU overhead and latency issues with Live vs Logic or Cubase etc. Live is the only DAW to have a gapless audio engine, drop plugins into a realtime mix with other DAWS and there is either a total pause of anything from half a second to about 3 seconds, and to achieve this, it does require more buffer space and CPU overhead. This is one reason that I mix quite often in Cubase, because in the studio, realtime playback isn't essential and I can run lower latencies and less CPU - but would I take Cubase out on the road with me as a Live performance tool - no way in hell !

[Tone Deft]

One thing to be aware of before comparing CPU overhead and latency issues with Live vs Logic or Cubase etc. Live is the only DAW to have a gapless audio engine, drop plugins into a realtime mix with other DAWS and there is either a total pause of anything from half a second to about 3 seconds, and to achieve this, it does require more buffer space and CPU overhead.

nice post. I wonder if it would be worthwhile to make this feature midi assignable. in the spirit of how some people say Live should have a live mode and a studio mode. live mode being real time buffered, studio mode meaning you have to stop to add effects, or the audio drops out.

[pepezabala]

tone, did you notice differences in latency between live8 and live7?

My BIG liveset needs a bigger buffer in live8, with the same settings as in 7 I get droputs when triggering scenes. But this was with 8.01, need to test again with 8.02.

This is only audio-related, I don't use 3rd party softsynths playing live. Actually I only use some simplers, drumracks, operator and electric. Got all them softsynths bounced to audio.

[Dalibor Loncar]

in time latency won't be an issue. new standards to replace USB and firewire, faster computers, we'll look back on this like we look back on betamax video tapes.

in a real world scenario latency will always be an issue, but the technical solutions will hopefully get better soon.

the culprits of "bad" latency compensation are the actual specifications. in the asio driver model for example there is only one global value for the input and output latency for a piece of hardware, but a soundcard mostly has more than one analogue in & out and digital ins & outs at that. the latency of a digital interface (spdif, aes/ebu, madi) is smaller than the one of an analogue port where conversion (AD DA) comes into play (in fact analogue connections also differ/vary in delay times). it would be indispensable to have a separate latency indication for each interface/port (analogue or digital) of a soundcard. to come clear: we need separate latency settings for each physical port of an audio interface and the host software has to support that. currently there is only one overall latency compensation setting available. then there is only one global value for the input and output latency for all sample rates, but the hardware delay @ higher sample rates also differs with the one @ lower sample rates. this problem fortunately could be solved by resetting the asio engine (maybe there also are interfaces on the market with a bad implementation of that feature). another problem is the simultaneously use of more than one connection on a soundcard. for example: you have connected your (bass-) guitar/synth/drummachine etc. via an analogue port of your soundcard simultaneously with another device which is connected via your digital interface of your soundcard -> now you want to record all the connected devices in realtime: the issue is obvious: which latency value should the driver now pass to the recording software? it will be ok only for one port ...

for the mac fanboys: the osx core audio specification isn´t better in this field (in fact the osx audio development team copied the specs of the ASIO specification)

[Tone Deft]

tone, did you notice differences in latency between live8 and live7?

My BIG liveset needs a bigger buffer in live8, with the same settings as in 7 I get droputs when triggering scenes. But this was with 8.01, need to test again with 8.02.

This is only audio-related, I don't use 3rd party softsynths playing live. Actually I only use some simplers, drumracks, operator and electric. Got all them softsynths bounced to audio.

dunno if I'd even bother to read Live's latency readout anymore!! I don't know if I can use a tool like the one I pointed out to get a genuine latency check between 7 and 8. I guess the sure fire way would be to do a 'loopback recording' out of and back into Live 7/8.

maybe with some releases Live 8 will be as 'light' on CPU as Live 7 but software always bloats.