Audio Latency: Haven't Found A Solution

Hello,

I'm running DaVinci Resolve Studio (v16 and v17beta) and I can't seem to find a way to overcome the audio latency I hear when recording audio in the Fairlight module.

I'm on a pretty fast and powerful PC running Windows 10 (latest update). I have an Audio Technica condenser microphone and for mic preamps I have two of them. One is the Focusrite 2i2 3rd generation and the other is a Behringer Q802USB mixer. Both devices connect to the PC via USB.

After a lot of research I see that the popular ASIO driver that people recommend using for other applications isn't addressed in Resolve and ASIO4ALL is of no help for the same reason. The only setting in Resolve that pertains to latency is in the General preferences and by default, it's set to Auto, with options for 128 samples and up. I have to leave it on the Auto selection because for some reason, once I select any other setting, I lose the ability to hear audio. Once I switch back to Auto, I get audio back. That seems like something is wrong but either way, if I was able to select 128, it wouldn't help. When I use other audio apps that do address the ASIO driver, I can go as low as 64 without any glitching and 96 seems to be the sweet spot with no glitching and the lowest latency, which is still too much. I see tons of posts and articles/videos about latency in Resolve and not one solution or suggestion helps me.

I have the Fairlight NEAR speakers set as either the Focusrite or Behringer interface and monitor with headphones from either of those devices. I create a track, assign the Audio Input from the interface to the track(s) desired, arm the track and begin my voiceover.

The latency is bad, very distracting and impossible to work with like that.

So, Ive been all over the internet and YouTube and I see all sorts of suggestions and "do this" types or responses, but no one really has a solution and anyone that suggests you change the buffer or sample size seems to forget that those options aren't available in Resolve.

I'd really love to get to the bottom of this but so far, I have no solution.
Do you have a solution or fix? I find it hard to believe that I can't work in Resolve without latency and I don't understand how so many people talk about recording audio and voiceovers into Fairlight but don't make mention of latency.

What is the answer? Is there one?

Thank you,
Robert

Well, I've pretty much given up on resolving the latency issue. After tons of research, trying many DAW's, testing on Mac and PC, and testing with different audio interface's, I see that it's possible to reduce the latency but not eliminate it.

Even if I just accept it, I can't believe that there's no way to have real time metering. Since the meters follow the latency, you can't even see them in real time. So annoying and frustrating.

I can't seem to find a way to overcome the audio latency I hear when recording audio in the Fairlight module.

One reason I prefer recording to a Digital Audio Recorder, like a Zoom.

Hey Jim,

I really like that idea but it's not as flexible as I would like in terms of sync and playback in sync with sounds or effects in Resolve. I know that there is compromise in any of the available solutions, so I'm trying to figure out what makes the most sense for me.

Bummer — it is really terrible.

Not sure why. Using a Yeti Blue sometimes there is no latency, sometimes it is almost a second. Crazy. WTF guys?

Maybe give us a manual setting or a way to calibrate?

Thanks — currently sucking.

Dan

Best I can see is to turn off the Bus on the recording channel in Fairlight.

Hello, guys! To be able to record a voiceover directly in Fairlight with nearly zero latency, follow these step. First, make sure you have an ASIO driver installed on your PC. In my case, the driver is already included in the Scarlett Solo 3rd Gen audio interface driver.

In this way, you can record with nearly zero latency, just like you can in Cubase PRO, FL Studio, Adobe Audition. In addition to that, you can also use real-time effects. For example, you can activate the Dynamics section in the Mixer to have a real-time Expander, Gate, Compressor while recording.

I have also encountered the same problem with high latency that made recording impossible. However, by setting the correct parameters, it can be used just like an authentic DAW.

Step 1. Go to Video and Audio I/O. Click on the I/O Engine and select ASIO from the list.

https://imgur.com/a/vKJw8X9

Step 2. Now click on Device and select your audio interface from the list.

https://imgur.com/a/5dYRTIn

Step 3. Deselect the AUTO option in Playback Processing Buffer Size and select 32.

https://imgur.com/a/YPCtkAM

Step 4. Click on Device Config to open the window where you can set the buffer size according to your preferences.

https://imgur.com/a/6wcnZhd

Hi Robert,

Every DAW suffers from the latency you observe. This latency is a combination of your interface's drivers, the buffer size used and the DAW itself.

To get around this problem you use the DSP mixer in your interface to generate a mix for your headphones. This mix will be a combination of your mic (direct from the interface input, not from the DAW), and the sound from your DAW excluding the track you are recording onto which must be muted while recording.

Some USB microphones can provide this latency free monitoring with built-in headphone outputs. This is the same as mentioned above except the microphone is also an interface (USB).

If your interface does not provide such a latency-free mix function then the next best thing is to mute the track you are recording onto and only listen to the DAW tracks through one over-ear headphone, leaving your other ear free so you can hear yourself more naturally.