Pro Tools 10 Low Latency Monitor Issue

[soundmusic]

When I put Pro Tools in to LLM mode the armed audio channel goes silent, this is obviously extremely frustrating.

Has anyone else experienced this problem? Any help would be much appreciated.

I am running:
PT 10
Mbox Pro 3
OSX Lion 10.7.2
Brand new quad 2.5 GHz Intel Core i5 imac

[CustomDigi]

I think that is meant to happen...meaning, it mutes the armed track so that you can hear your interface's input through its own proprietary software mixer (IE Motu CueMix) without that flangy/phasey sound. However, I'm not sure how that works with the Mbox3. I thought Digi's stuff already had a provision for LLM.

I was under the impression that the newly added LLM function was strictly for people that have an interface that has its own 'mixer' application where you can monitor the input directly.

Originally Posted by soundmusic

When I put Pro Tools in to LLM mode the armed audio channel goes silent, this is obviously extremely frustrating.

Has anyone else experienced this problem? Any help would be much appreciated.

I am running:
PT 10
Mbox Pro 3
OSX Lion 10.7.2
Brand new quad 2.5 GHz Intel Core i5 imac

[Doc Mixwell]

Yes, it is MEANT to mute/un mute the record armed track for you, so that a 3rd party Software "Input Mixer/Matrix" can take over. It shuts off the processing on the track, and shuts of the audio from playing through your buffer size. This is the point, as when you raise the buffer size to accommodate your workflow, and need to overdub a track, you can use this feature with your 3rd party Software "Input Mixer" and monitor with no latency, and punch in, out, moving right along.

The MBOX PRO, DOES IN FACT have a software driven mixer/routing program, that allows you this possible outcome. Finally, we can monitor external SPDIF inputs without latency on an MBOX. I will say though, the MBOX pro rocks out with Low buffers, using Pro Tools, on a strong computer, so I haven't even needed to use the Mixer for latency reasons at all!!! Throw that sucker down to 32 samples, see if you can work effortlessly. If you can't at high DSP volumes, than the MBOX mixer will work great with the Low Latency Mode of PT10.

Good Luck!

[mirrorboy]

Originally Posted by Roc Mixwell

Yes, it is MEANT to mute/un mute the record armed track for you, so that a 3rd party Software "Input Mixer/Matrix" can take over. It shuts off the processing on the track, and shuts of the audio from playing through your buffer size. This is the point, as when you raise the buffer size to accommodate your workflow, and need to overdub a track, you can use this feature with your 3rd party Software "Input Mixer" and monitor with no latency, and punch in, out, moving right along.

The MBOX PRO, DOES IN FACT have a software driven mixer/routing program, that allows you this possible outcome. Finally, we can monitor external SPDIF inputs without latency on an MBOX. I will say though, the MBOX pro rocks out with Low buffers, using Pro Tools, on a strong computer, so I haven't even needed to use the Mixer for latency reasons at all!!! Throw that sucker down to 32 samples, see if you can work effortlessly. If you can't at high DSP volumes, than the MBOX mixer will work great with the Low Latency Mode of PT10.

Good Luck!

Hey Roc Mixwell-

Is this also true of PT 10 HD Native in that it mutes the output and you use your 3rd party software mixer to monitor?

thx man,

Scott

[Doc Mixwell]

Hi Scott,

Yes, that is correct.

[cowrange]

With HD native it's the card's onboard mixer that would "take over" and provide the low latency. Kinda like how it works with 002 and 003 etc. They've just extended this functionality so you can use it with 3rd party mixers in PT10.

[Doc Mixwell]

Yes, the PCie HD Native system [Not HD Native with 3rd Party Interfaces up to 32 I/O] has a FPGA on the card, that allows for zero latency throughput on record armed tracks within the software environment. It does not shut off the track inside the mixer, but does shut off the processing on the track.

[soundmusic]

Thanks a million for the info there people, it sorted me out. I didn't realise that was the case with the new interfaces, i'm only returning to recording after a few years, my brain is still in digi 001 land.

Cheers.

[engmix]

Originally Posted by Roc Mixwell

Yes, the PCie HD Native system [Not HD Native with 3rd Party Interfaces up to 32 I/O] has a FPGA on the card, that allows for zero latency throughput on record armed tracks within the software environment. It does not shut off the track inside the mixer, but does shut off the processing on the track.

When you say processing you mean "no plugins" on the track while the track is armed.

[Doc Mixwell]

Originally Posted by engmix

When you say processing you mean "no plugins" on the track while the track is armed.

Yes, it shuts off the Plug in's and Bussing, with record armed tracks...

[strauss]

Does it only mutes when recording or also when it's just armed?
Punching in gets complicated if it also mutes when it's just armed.

[Doc Mixwell]

It is supposed to make monitoring and punching easier, when using higher amounts of CPU processing, forcing you to run higher buffers. So, to be clear, when this mode is turned on, it shuts off automation, plug in's and bussing, and shuts off the INPUT stream, of record enabled tracks, while recording, or during playback.

It won't prevent track PLAYBACK stream for record enabled tracks, provided there is any previously recorded material on the tracks you are punching onto. It handles everything for you basically. Your 3rd party input stream takes over, and is mixing the Live Input with the DAW return to a pair of your interfaces outputs.

[soundrick]

Originally Posted by Roc Mixwell

Yes, the PCie HD Native system [Not HD Native with 3rd Party Interfaces up to 32 I/O] has a FPGA on the card, that allows for zero latency throughput on record armed tracks within the software environment. It does not shut off the track inside the mixer, but does shut off the processing on the track.

I've tried using this in my HDNATIVE about 15 different ways to Sunday, and I can still hear a little bit of audible latency. It isn't as much as it would normally be with higher buffers, but it does increase with buffer size.

It has to be user error, I've been reading about it and it seems like I don't have it working like it's described.........

I set up my LLM path in I/O settings to A 15/16 and patched that hardware output to a mixer. I've tried setting up an output send on the track. I've tried setting the track's output. I've tried routing the send to an aux master out. I've tried input mode and auto, etc.............etc............etc.............

Still can't get it right.

[floored2112]

Unfortunately to use the aux sends while recording you need to set your buffer to 128. Pro tools 10 was supposed to fix this but I have not seen a change. I intend on asking the guys from Avid about it at this years NAMM show. It seems that It's the only time you can get any real answers from them. If anyone has any questions they want asked let me know.

[dubrichie]

Originally Posted by Roc Mixwell

It is supposed to make monitoring and punching easier, when using higher amounts of CPU processing, forcing you to run higher buffers. So, to be clear, when this mode is turned on, it shuts off automation, plug in's and bussing, and shuts off the INPUT stream, of record enabled tracks, while recording, or during playback.

It won't prevent track PLAYBACK stream for record enabled tracks, provided there is any previously recorded material on the tracks iyou are punching onto. It handles everything for you basically. Your 3rd party input stream takes over, and is mixing the Live IHnput with the DAW return to a pair of your interfaces outputs.

Hang on, if the INPUT stream of the recording track is turned off, why is it necessary to also turn off ALL of the processing and bussing?

This would only be applied to the prerecorded audio on the track, which you might be punching in over, etc. So there would be no problem with latency.

This is a pain in the ass for me, as I use a multichannel monitoring system (Furman HDS-6), to which i route using sends in ProTools. What I would want to do is to combine the live input with this send output from PT in my interface's hardware mixer (RME, Lynx, Motu, etc.) so that the artist has "Me" on ONE knob on their cue mixer, which gives them both the prerecorded track or take AND the live input they are overdubbing.

And what if i/we need some reverb or whathaveyou on the prerecorded track or take? Putting such FX on the live input stream isn't a problem, you just use an aux track with its input set to the live input channel, disable delay compensation on it and send/route it wherever you want.

Do you catch my drift?

[Doc Mixwell]

if you've got a strong enough computer, you'll never even need to flip that on,

[dubrichie]

Originally Posted by Roc Mixwell

if you've got a strong enough computer, you'll never even need to flip that on,

Completely moot i'm afraid.

My point stands. Avid did not think this through fully. Very, very easy for them to put right and make this functionality A LOT more useful. As it is, well, it's a bit of cock-up isn't it?

[Doc Mixwell]

(snip)

[diff caliber]

so because every decent interface is now coming loaded with there own "premix"
software , is latency a big issue anymore? Its my understanding that latency is only a factor when recording. I know AD/DA is still a factor because that deals mostly with quality, in how well converters read signals and data and how it interprets that stream line of info. But again, is latency that big of an issue, meaning overrated, in modern recording? Please someone enlighten me. Keep in mind almost every daw also has internal ways of dealing with latency.

[Shanemacc]

My issue wasn't with LLM but with not being able to hear the existing audio in quick punch mode using pre roll. In previous versions I would set pre roll for 2 bars, hit play, the track would roll and I would hit record. I would hear r recorded track until it reached the punch point. In version 10 the recorded audio would go away and I would hear the live mic. After waiting forever on hold I spoke to avid today and we figured out that in rersion 10 they changed the order. Hit record first and then play and you will hear the recorded track until it goes into record! Yeah :-)

[Joe Porto]

I have a slightly different problem w/ LLM using input track monitoring. I am monitoring 8 tracks of hardware MIDI instruments via input track monitoring, and for some reason, when I turn on LLM, output channels 1 and 2 no longer sound, but 3-8 are fine. This is with an Apogee Symphony system. I thought maybe it was because 1 and 2 were used for audition, but set that (and default mono and stereo channels) to another channel but no luck. Also, I can LLM from inputs 1 and 2 to, say outputs 3 and 4, but can't LLM from any input channels to outputs 1 and 2.

I poked around in Maestro as well, and nothing indicates that outputs 1 and 2 would be muted.

[floored2112]

My problem is setting up a headphone mix using aux sends. Can anyone tell me how to set up a headphone mix without dropping down to 128? I would like to keep it at the highest sample rate but you cant use LLM and use aux sends at the same time.

[mikeyman]

Originally Posted by Roc Mixwell

if you've got a strong enough computer, you'll never even need to flip that on,

Whats a strong computer to you ...? thanks

[cebolao]

Originally Posted by Shanemacc

After waiting forever on hold I spoke to avid today and we figured out that in rersion 10 they changed the order. Hit record first and then play and you will hear the recorded track until it goes into record! Yeah :-)

Really? Which exactly version has this feature?
I don't get it - does it allow "freestyle" punching - whenever i punch in the audio gets muted, when i punch out i immidietally hear the recorded audio?