Long post: skip to the bottom if you want to read it all.
I am setting up a project that involves using a DAW as a live mixer while also streaming the event via OBS. I know there are about a half dozen different ways to do this, but considering it is live, latency is a major consideration. Also, I am designing something others may need to operate, so I don't want to get to far into finicky solutions that require a lot of nerdy tinkering.

Background, I am on Windows and will likely be using Ardour as the DAW, but will be buying a ProSonus interface, so the included version of Studio One is also an option. I have a live audio background including using digital mixers, but DAWs are new to me.

The obvious option is using something like ASIO4All to wrap the interface and a virtual audio cable together as a single ASIO device for the DAW to use. My concern with that is latency. ASIO4All will always use the MME driver for devices it talks to, even if they natively support ASIO, so it seems like that would add latency there as well as latency inherent to the virtual audio cable.

I know ASIO routers / patch bays exist, but options there seems limited / finicky and OBS doesn't natively support ASIO anyway, so I would still need some kind of bridge.

OBS does support VST plugins, so to me that seems like the best option. Have the DAW directly control the hardware interface via ASIO and use VST plugins to send sound to OBS.

I have found a number of VST plugins for sending/receiving audio, but most are geared towards network audio, which clearly adds latency. Are there any VST plugins with the same idea (acting as a virtual patch cable / insert) that use some kind of local routing like ASIO to keep the latency super low?

 

The PreSonus audio interface also comes with drivers that are much better than ASIO4ALL.

Podcasting and Live Streaming - PreSonus Technical Articles
www.presonus.com/learn/technical-articles/podcasting-and-live-streaming

 

Agreed, that is why I am trying to avoid using it.

 

What is latency in music production?

Audio latency is one of the things that you will pay more and more attention to, once you're getting more skilled at music production.
Latency is the time delay it takes for your hardware and software equipment to read the sound signal that's being played, process it, and play it into your speakers.

To be more in-depth, when an audio signal is being played on any device, let's say microphone or guitar, for example, that sound is in analog form and being delivered into your system. Your software then converts it into a digital signal in order to be processed, and then changes it back into an analog, for it to be played into your speakers/monitors.

This whole process, having low processing power in your machine, not having the correct settings and tools optimized for lowest latency output, and a couple of other factors, are all potential reasons that might cause a high latency.

Having a better setup will of course help in reducing latency, but there are other things you can do to improve your latency.
One of the most important things you need to do if you want to start producing music is to get an audio interface. Buying an audio interface is going to be a lot more reliable than using the cheap sound card your pc or laptop has built-in. It also allows zero-latency recording, with its built-in "direct monitor" option. Just remember to update all of the drivers for it first.

Buffer size is the amount of time spent for processing the audio, measured in ms. Installing an audio driver like Asio will give you the option to increase or decrease your sample rate. Having a lower sample rate is going to improve your buffer size, but it's a lot more taxing on your system.

Use the lowest sample rate when you are recording, and set it to a higher rate when mixing, this will spend less of your computer's resources, and allow you to use more plugins without stutters and errors.

https://transverseaudio.com/tip/what-is-latency-in-music-production

 

video encoding takes around 200ms in OBS (depending on encoder settings) so you have lots of headroom when it comes to audio,
best pro-grade hardware encoders can go as low as 1 frame (so about 34ms with 30fps video),
OBS is not an audio-friendly piece of software, so you'll need some virtual Loopback audio device, or some local streaming plugin, like ReaStream by Cockos,
if you want heavier audio processing, then better go for 2 computers (1 for DAW, 2nd for OBS)....

 

Yeah, I was looking at ReaStream, seems like a decent option. There is also a VoxengoRecorder plug-in that can output to a MME device such as a loopback driver. Between the two I would guess the VoxengoRecorder has lower latency considering it doesn't interact with the IP stack, but as you pointed out, ReaStream opens up the option to run the DAW and OBS on seperate systems.

Yes, I understand that OBS will inherently have a delay, which I plan to accommodate for by measuring and delaying the streamed audio by the same amount. What I don't want is a solution that requires me to delay the live audio just to interact with OBS (such as using ASIO4ALL as a wrapper).

 

I'm not a network expert to tell, but I think local IP stack (sending/receiving) within one computer should definitely offer lower latency than any MME based loopback driver,
ASIO is an audio mode and its properties are highly dependant on audio interface driver implementation, only few interfaces offer nearly zero-latency loopback like RME, most interfaces allow direct monitoring without computer which isn't what you look for,
ASIO4ALL is software emulation of ASIO hardware with use of computer CPU, so nothing good for reliable low-latency task like livestreaming...

 

Thanks for all the feedback, I know what I am going to setup for the test/demo. I probably should have mentioned this, but latency between the DAW and OBS isn't a huge issue because stream viewers will not be in the same room. All I care about is latency between the Mics and loud speakers, so Interface-DAW-Interface. The main thing I care about on OBS is consistency. My video feed has a 340ms delay, so I am already delaying audio by that amount. If the audio chain ends up adding 50ms, all I have to do is subtract that from the delay and use 290ms. The important thing there is consistency. I can account for consistent latency, jitter is what would kill my lip sync.

I am a network expert (not meant to be a brag, but that is my profession), I could see arguing either way if MME or IP adds more latency. I do think the IP method will be more predictable. Also, VST settings can be saved, whereas Windows Audio devices settings seem to change themselves randomly. So my plan is to use a VST plug-in on both the DAW and OBS to transport audio.

I am already using the NDI protocol for video distribution from the camera to OBS over the network. That software toolkit includes a VST plug-in to send/receive audio. Given that NDI is primary a video solution, I would probably choose ReaStreamer instead if I was only worried about audio, but given that we are already using NDI, I will try that first in order to limit the number of different tools involved in the solution.

 

Forgot virtual cables. Use reastream plugin