Tracking with Buffersize 32?

[audiogamble]

I’m recording at 64 buffer on an m1max MBP fully loaded. Also using an Orion 32, but it’s connected over MADI mind you. It’s worked well and never had any complaints. In theory, there should be more latency going over the madi connection but it’s still been totally workable. I imagine 32 buffer would be fine too but I don’t like living on the edge like that.

Only issue is keep your master bus clear if you’re running in rosetta mode. Even no-latency plugins on the master bus have a tendency to act weird in a tracking scenario. Especially sound id. Haven’t done any critical recording in silicon mode, but I do remember it wasn’t a problem like this when I was on intel.


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[its2loud]

Originally Posted by Darryl Ramm

If you are going to do it, do it well. So as a community service announcement: A friend got me hooked on popcorn popped in Avocado oil with parmesan cheese grated on top. Quality Avocado oil adds a great flavor and has a very high smoke-point.

I pop my popcorn with a 256 buffer setting. 150ms between kernel popping. There’s no combing. Then I add the butter with 512 buffer setting and 15ms between salting. I can hear every kernel.

[LDS]

Originally Posted by RobertDorn

Allright. Perhaps I have to check again now that I'm on an M2 max cpu, actually curious how it works in practice. Last time I compared how it worked in real world practice was when I was on an HD-Native thunderbolt + HD I/O system, tracking at the lowest possible playback buffer in pro tools and working with mix downs in a fresh session. Till I plugged in a guitar into an UAD Arrow (solo nowadays) and thought 'wowww, this feels like just playing through analog gear' . I then swapped my HD native system for some Apollo X racks. But due to the lack of connectivity swapped that later on to HDX with a MTRX , so let's say since 2017 I've only recorded with DSP based systems.

Curious to check out how it'd be nowadays, when tracking just completely natively, especially when overdubbing on big sessions, because I can do that without thinking about the native playback buffer now with DSP, even if the session runs on a native playback buffer of 1024 samples.

The biggest benefit of DSP systems is that they make achieving low latency relatively easy. We can walk into a shop and just ask for it! I suspect a lot of people here have never actually measured the real world latency of their systems. Some would probably be surprised how big the latency is in 'acceptable' HDX sessions.

I ditched HDX2 last year after upgrading my computer. Particularly with the changes to I/O in Pro Tools, I discovered that 400-600 voice sessions at 64 samples were easy-peasy. My native sample buffers never move. But the more surprising aspect of it was that adding a HDX card back into the system sent it backwards in terms of tracking capability. In the 400-600 voice native sessions, I can punch in ~100 channels to record with plugins, routing and sends in place. Putting a HDX card into the system reduced it to 64 inputs, and almost halved the number of plugins and routing on offer.

That is purely the result of hitting the limitations of DSP in HDX. It took me days to comprehend and come to terms with what I was seeing. For decades now we have all been adding DSP to improve the performance of our native systems, but this was totally the other way around. Suddenly I was presented with the reality that I needed to add more DSP to ensure HDX kept up with the native system! $12000AUD in DSP to match a $2500AUD computer, basically.

It takes a little bit of research to execute an effective native system, but it isn't remotely exhaustive. In some people's worlds, DSP will always be 'better'. In others, an all consuming focus on sample buffers will be their thing. But for me and my world, at least, DSP is all but dead. That is the beauty of running your own tests and taking some measurements. You figure out what works best for your own needs, rather than climbing onboard the endless, generalised assumptions. The way people talk about DSP, and native sample buffers, and latency in this day and age has become very reminiscent of other tired digital audio internet statements. Stuff like "sample rate is stair stepped". Or "One bit equals 6dB". Or "resolution decreases with volume".

[Eric Lambert]

I'm with you. having used TDM for ages, then jumping into Native with my home studio I was a little apprehensive because, at the time, TDM still felt like the King in terms of performance and lack of compromises. Then Native hit its stride with CPU upgrades and ProTools making it a more friendly architecture, and today I think I'd feel less happy about DSP. Where it once shined, Native has now (more or less) covered the spread. Latency is never an issue, and the amount of power available to something like a Studio Ultra is ridiculous.

[camshash]

Just wanted to add to the dumpster fire that is this thread.

I have been recording (16 channel full bands/drums), editing, and mixing with my MBP M1max at 32 buffer 48k and have had zero issues. This thing is a beast, Ive left it at 32 buffer for everything the entire time I've had this machine. Plugins that add a siginificant amount of delay compensation I avoid during tracking, but from what I've learned here on the duc is that is a feature of those plugins, and not a limitation of the computer.

Also to add, 128 buffer is the max I'll go, 256 is too much latency. I feel that any difference between 32 and 64 is really the phase interaction of the headphones/IEMs with the direct sound rather than any perceived latency.

[RobertDorn]

Yes, silicon Macs are beasts! But as stated above, tracking without DSP / and a dedicated DSP path, requires to keep an eye on the native playback buffer and on the number of samples that plugins used in the recording session add. For those who don't want to be bothered with that and like to punch in something on a fully loaded session with 3000+ sample delay plugins and a native playback buffer of 2048 samples, luckily also in 2023, we still have DSP options :)

[lesbrunn]

Originally Posted by JFreak

Monitoring does not need to have a sound. It is used to control your voice when you are singing. Feelgood is not necessary, if you know how to sing.

With that said, the really good singers with a classical singing technique almost always require one ear left open for own voice and only one ear for the guide track.

Yes, been there and done that. I can say for certainity -- with cans -- that I am okay with 256 but not okay with 512.

Again, default buffer of those amazingly fast systems is/was 128.

Please do not be unrealistic.

[lesbrunn]

Originally Posted by JFreak

With that said, I know that this threshold is personal and there are some singers that just cannot perform at 256. I however have had this blind tested for me and for me 256@48k is fine, TDM session with 375@48k is fine, but 512@48k makes me hear echoes. So my personal threshold lies somewhere between 375-512 samples at 48k.

Excuse my French, but you do have a weird threshold.

[lesbrunn]

Originally Posted by JFreak

^ yeah.

Whenever I have disagreement I just tell the singer to drink water while I fix this, then continue without doing anything. Usually no complaints afterwards.

If there really is a problem, and I know some are more sensitive, usually best solution is to leave one ear open. Singer with a real singing technique cannot live with two ears closed no matter what.

Still, unless using some super crazy latency introducing plugin, 256 buffer is a good starting point. I have absolutely no symphaties for "64 is too slow" because the problem is somewhere else. MIX/HD had bigger roundtrip latency with 0 buffer (no plugins).

Really bro?

[JFreak]

Originally Posted by lesbrunn

Really bro?

Yes -- the latency problem is somewhere else than playback buffer.

As I already told you, TDM systems have bigger roundtrip latency (than 64 samples) without any plugins (in another words: without playback buffer) and that has been touted so super fast there is absolutely no monitoring problems. And the fact is, TDM latency builds up as you add plugins -- but it is still okay because TDM is so superfast. Everone says so, therefore must believe it is true. This is psychological for the artist and technical to the audio engineer.

Maybe your AD/DA is too slow? Maybe psychology says that if it isn't hardware (TDM) it must be slow because there is native processing involved? Maybe 64 buffer is too fast and the monitor mix is starting to fight against what the singer hears through bones? Who knows.

But it is not the sample buffer that needs to be faster. Think about it: 64 samples at 48kHz means 1.33ms latency for plugin processing.

Converted to distance it means this: 0.00133 seconds * 340 meters/second = 0.45 meters = 45 centimeters. In other words same latency as distance from singer to microphone.

It also means the latency can be fixed by singing just a bit closer to the mic.