PRO TOOLS HD LATENCY VS. LOGIC 7 PRO

[theother]

I still would like to know what the 'real world' round trip with PT HD @ 44.1kHz is.

Maybe someone finds the time to check it out including converters.

I did a quick test with my RME MADI (winXP) and the CraneSong HEDD 192 @44.1kHz.

At buffer size 32 (0.7ms) I get 200 samples (4.5ms?)
and at 64 (1.5ms) I get 264 samples (5.98ms) for a total roundtrip incl. converters.

I use 64 most of the time, which 32 I can only use very basic setups with not a lot of plugins/softsynths.

the way I measured it was:

Analog signal on Input Left recorded to Audio Track 1.
Monitoring Audio Track 1 on analog output R.
Analog output R directly patched into Analog Input R.
Analog Input R recorded to Audio Track 2 at the same time as Audio Track1.

--->afterwards measure offset Audio Track 2 from Audio Track 1.

nativeraudio, are the 2.34ms you've measured including analog converters or digital only?

And it's very important to measure a real roundtrip and not the analog inserts.

If I measure the analog inserts with Cubase 4/RME I get 15 samples for a digital and 99samples for an analog roundtrip.
The test would be:

Send an audio track to an output and re-record it to another track.

Result: 15samp digital/99samp analog

[nativeaudio]

Originally Posted by theother

nativeraudio, are the 2.38ms you've measured including analog converters or digital only?

It was total roundtrip, including A/D and D/A conversion, on a PT HD system (not Accel). The Mix systems had a slightly lower latency. But some of the plug-ins I used a lot back in the days of the Mix systems added latencies: LoFi (405 samples, or 9 ms @ 44.1), Waves Ren Comp (65 samples, or 1.46 ms @ 44.1) and C1 (341 samples, or 7.7 ms @ 44.1) - and people were (and are?) recording with these plugins on the recording track. Well, not the LoFi, I guess. The Ren EQ had only 1 sample latency.

When comparing two sessions, one with a Ren Comp and Ren EQ on a TDM system @ 44.1, and one with Logic and two native plugins (not adding latency, since the processing happens within the buffer cycle), here's the difference we get, based on the numbers from this thread:

TDM system: 105 samples + 66 samples for the Comp and EQ = 3,87 ms.
Native system using a Mac Pro: 3 ms (@ 32 buffer setting, I guess). (Last time I checked at Apogees site, the recording roundtrip @ 44.1 was 3.3 ms, but Max mentioned 3 ms in this thread - maybe he was rounding off...)

In both cases (rounding off or not), a native recording track has lower latency than a TDM recording track if an EQ and a compressor is used during record - if the plugins I happen to know the values for are used. (Disclaimer: I haven't checked if the Ren Comp still use 65 samples).

While I think that PT will be strong and alive for many years, and that it will take a little time before TDM users will trust that they can get what they need from a native solution, the situation reminds a lot of 1991-1994 where the 'oldies' were very skeptical about Pro Tools (and had a good reason to be worried!). It took two years from Sound Tools developed into the first 4 voice Pro Tools system, and it took another three years before PTIII came with a 'normal' amount of voices. When I suggested to colleges back in the early days of PT that one could skip using tape and possibly even mixers, and use a TDM system instead, the arguments and criticism against PT and TDM were pretty similar to what TDM users say about native systems today - "it won't be reliable enough, it's gonna crash all the time (PT was very unstable in the early days), we'll loose clients..." etc., which is.... interesting.

[TC Custom Audio]

The reason I am bringing this up in a Logic vs. PT HD thread is that Logic, even with the Apogee Symphony cards, still monitors through software. This means you're a slave to your buffer size and that sucks.

I run my Lavry Blue and Aurora 8 into a Lynx AES16 installed in a PC. I just moved from Nuendo 3.x to Cubase 4.52 and the Control Room FINALLY works with direct monitoring (i.e. through the DSP mixer on the AES16). If I recall correctly, Steinberg referrs to this functionality as "ADM." So, I get a 5 mixes (monitors + four stereo headphone feeds) and this works the same for live inputs and playback and is accessible entirely through Cubase. No more messing around with a secondary montitor mixer.

I assume this also works with Nuendo 4.x and higher. I don't know, however, if this functionality is supported on MAC. Also, I'm not sure what other interfaces besides the AES16 work with this. I've heard that the RME cards don't do this, but I'm not sure about that either.

Just to be clear, I achieve acceptable (i.e. comparable to PT HD or better) monitoring latency ALL the time, regardless of my ASIO buffer size or how loaded up my projects are.

Here's what I don't get that PT HD does provide: no plugins on live inputs and I am limited to 4 cue mixes. However, I only use analog outboard while tracking and I don't have need for more than 4 cue mixes.

My point here is that Cubase finally is a viable alternative to PT HD for some professional tracking situations. Of course there is the fixed AD-DSP-DA latency, and the only way to get around this is by making analog cue mixes on the front end with an analog mixer. This is also a limitation of PT HD.

My 2 cents.

[nativeaudio]

Originally Posted by Sannytown

The reason I am bringing this up in a Logic vs. PT HD thread is that Logic, even with the Apogee Symphony cards, still monitors through software.

That's true, if you need to use live plugins on the tracks you are recording, which you don't seem to need. Some people do, some don't...


If you don't, you can just monitor directly from the inputs, which will give you lower latency than PTHD (see below). If you need to increase you buffer setting to 64 or higher, your monitoring signal wouldn't care, because you won't be monitoring it through the software. You can still keep Software Monitoring mode ON if you want - this has some benefits.

Here's the setup (make sure "Independent Monitoring Level for Record Enabled Channel Strips" is enabled in Preferences>Devices>Core Audio):

• Keep software monitoring on. This will let you use sends to internal/external reverbs and delays for the recorded track.

• Turn the fader on the recording track all the way down, and make sure you use pre fader mode if you use reverb sends during record.

• Monitor directly (you'll set this up on you hardware mixer or the software mixer that comes with your interface).


That's it. Now you're ready to record.

Your whole mix will be heard in the headphones if set up the normal way, the recorded signal will be heard with a latency that's 70-80% lower than PT HD's latency with the right hardware, and the reverb will come with a tiny, insignificant pre-delay (which it always does anyway).

it it will take a little time before TDM users will trust that they can get what they need from a native solution

Disclaimer: my previous post was written back in 2006. Time is running fast!