aaronleese wrote:here you go.... not an exotic problem at all, you can replicate it with a single plugin: https://www.youtube.com/watch?v=Tstw68U-24w
leisuremuffin wrote:No one is arguing that automation isn't compensated
aaronleese wrote:here you go.... not an exotic problem at all, you can replicate it with a single plugin: https://www.youtube.com/watch?v=Tstw68U-24w
leisuremuffin wrote:No one is arguing that automation isn't compensated
this is the other issue... time based fx that have reference to the bar will be out of sync... i.e. if it's just using the BPM to set the delay time - fine. if you're trying to do trancegate effects, though, e.g. where the 1st beat is suppressed, that will be out of sync. this is arguably worse than the automation issue, because automation you can move manually; there really isn't any workaround for this one, though a lot of third party tools like LFOTool and VolumeShaper/FilterShaper will let you offset it there.ze2be wrote:Also all time based fx in generel, 3rd party or native are not very well compensated if I remember corectly.
Exactly. It's all cool and all that people are dedicated to live, but eventually you just have to face unmistakable issues.garyboozy wrote:aaronleese wrote:here you go.... not an exotic problem at all, you can replicate it with a single plugin: https://www.youtube.com/watch?v=Tstw68U-24wleisuremuffin wrote:No one is arguing that automation isn't compensated
umm... it *better* add up buffer delay every time you insert a VST... otherwise, the *audio* will be out of sync, not just the automation. however, from what i have heard, some other hosts don't adjust while playing - they don't recalculate PDC until you stop. live may be unique in that it attempts to keep things sample-synced while it's playing... either way makes sense, there's no way to insert a latent effect and not get at least a hiccup somewhere, so stopping is not a bad requirement if you're doing studio work.pencilrocket wrote: 2. Live has odd behavior that adds up buffer delay every time you insert vst. No other host does this.
All hosts add plugin latency to compensate.theophilus wrote:umm... it *better* add up buffer delay every time you insert a VST... otherwise, the *audio* will be out of sync,
But it's even worse..: If you manually offset the waveforms, it can get out of time as soon as you add other plugins or channels, our change routings.theophilus wrote:this is the other issue... time based fx that have reference to the bar will be out of sync... i.e. if it's just using the BPM to set the delay time - fine. if you're trying to do trancegate effects, though, e.g. where the 1st beat is suppressed, that will be out of sync. this is arguably worse than the automation issue, because automation you can move manually; there really isn't any workaround for this one, though a lot of third party tools like LFOTool and VolumeShaper/FilterShaper will let you offset it there.ze2be wrote:Also all time based fx in generel, 3rd party or native are not very well compensated if I remember corectly.
Q: FilterShaper or VolumeShaper work fine in Ableton Live when being the only plugin on a track. But they can show noticeable latency in other cases — this can (but does not have to) happen when there are several plugins on the track, or when used on a group or on an audio track with other tracks routed to it.
A: This is a PDC (Plugin Delay Compensation) issue with Ableton Live. FilterShaper and VolumeShaper are synced to the timing information that is provided by the host. Sadly, Live does not take PDC into account here and thus the timing information provided by Live can be wrong. We've already talked about this with Ableton in 2011 and are glad about anyone who bugs them about this issue.
Hmm, these may be good practice in live for you. But its not good practice for Music Production generally. I don't want to open another this DAW that DAW thing here, but in S1 changing buffer has no side-effect to the audio being produced. Also -for me- setting higher samplerates in live requires much higher buffers to avoid artifacts, i guess because it consumes more CPU - ending up in more delay over all. Because of that it also doesn't scale easily to bigger projects.leisuremuffin wrote:I don't know about that but I think working with consistent settings start to finish is the way to go. And I also think it's best to use the highest sample rate that you can handle. Not for fidelity but for lower latency. 512 samples is obviously faster at a higher rate.
1. All PDC Algorithms need to add delay on other tracks to compensate for a new plugin that has a plugin delay.pencilrocket wrote:Live is more likely to get muddy phasey mixes. Why? Because there are two obvious problems. 1. Automation isn't compensated. 2. Live has odd behavior that adds up buffer delay every time you insert vst. No other host does this. 2nd issue aggravates 1st issue more than the other hosts.
That's what people are complaining.
kayhel wrote:Hmm, these may be good practice in live for you. But its not good practice for Music Production generally. I don't want to open another this DAW that DAW thing here, but in S1 changing buffer has no side-effect to the audio being produced. Also -for me- setting higher samplerates in live requires much higher buffers to avoid artifacts, i guess because it consumes more CPU - ending up in more delay over all. Because of that it also doesn't scale easily to bigger projects.leisuremuffin wrote:I don't know about that but I think working with consistent settings start to finish is the way to go. And I also think it's best to use the highest sample rate that you can handle. Not for fidelity but for lower latency. 512 samples is obviously faster at a higher rate.
A word to bigwig: I know it is not perfect, there are still many features i miss - i.e. rewire to be able to integrate it into S1 for easy submix (and ARA usage...). But the PDC discussion between users and devs in KVR-forum was exemplary and ended up in a very fast solution after days, not years. Still, PDC is not fully done, but if it would be implemented in live as it actually is in Bitwig, i would be the happiest live user. What i miss ist that live is simply ignoring this - there's not even a discussion possible with the live guys regarding PDC.
And come-on... Timing is _the_ crucial thing in music production these days. If we are talking about best practices, my suggestion is to talk about improving the root cause of these never ending threads instead of talking about workarounds. Because: none of these so called yet-another-best-practice-workarounds did work out for me, they all ended up in having other side-effects. And it looks that I am not alone here...
Thanks leisuremuffin, I'll give it a try and report back soon.leisuremuffin wrote: I've found that for me, 88.2 24bit 1024 size buffer works on my largest projects