Latency When Recording Audio Into Davinci Resolve

Hello,

I am trying to record a voiceover into DaVinci Resolve, but when I do, there is terrible latency. Even when I speak on time, the audio is delayed in my timeline, and I constantly have to move the audio file back to line it up with where it is supposed to be.

I can't find any settings to deal with the latency. Is there something I'm missing? How can I just have the audio I record show up without any lag?

What audio interface are you using? There are no latency adjustments in Resolve. The latency is coming from your system and whatever audio device you are using to record and replay.
That is unless you have a monitor offset set which compensates for video replay.
From the manual.

I can't find any settings to deal with the latency

Application Preferences > System > General > General Preferences > Audio Processing Block Size.

I see that setting but 128 is way too much latency to monitor while doing voiceovers. Is there another way to get the latency lower? Why can't we set it lower than 128? My audio interface goes down to 16, although I've found that 64 is about the max I can go without the latency interfering with my voiceover.

Also, I don't see any setting for video monitoring offset, not in the section shown above or anywhere else.

I don't see any setting for video monitoring offset

Preferences > User > Fairlight > Video I/O Offset > Video Monitor Offset.

A custom delay of between 0 and 7 frames may be specified.

I don't see any setting for video monitoring offset

Preferences > User > Fairlight > Video I/O Offset > Video Monitor Offset.

A custom delay of between 0 and 7 frames may be specified.

No settings like that are there...

Ensure that the most recent version of Resolve is installed - currently 16.2.7 (010).

This function is outlined in the August 2020 Resolve manual:
Page 135 | Chapter 3 | System and User Preferences

System Preferences.png (135.34 KiB) Viewed 4058 times

Ok, thanks. I updated and made sure it was set to 0, but I'm still getting a lot of latency.

I was also getting an error saying my speakers could not be found and after messing around a while I wound up going into my speaker device settings in Windows and changing them from 24 bit to 16 bit, which is strange because they worked on 44.1k on 24 bit, even thought he timeline is 48k. It was a weird place to have to go to get my speakers to work and no other program has demanded that I do that. So if I want to use Resolve I'll have to remember to go into my speaker settings and change that before I open the program, because I use 24 bit to record music in Reaper. I don't know if I can set the bit depth before I create the timeline, like the sample rate.

I really like Resolve's interface and it is definitely smoother at editing than Premiere, but these audio issues might be a show-stopper. Either way, I need to get away from Premiere. That program is horribly choppy and I haven't been able to fix it in like 2 years.

Bit depth is not as important as sample rate in Resolve. Resolve runs at 48k which is normal for video. If you import a 44.1k file, Resolve will automatically resample it to 48k.
You do need to make sure that your interface is set to 48k when recording directly into Resolve, and for replay.

These are my settings in Windows using the in built Realtek card. On the recording side the max bit rate is 16 which is either a limitation of the card or Windows.

Resolve has no problem importing and using 24bit audio files.

Remember, Resolve does not recognize ASIO drivers.

With respect to Charles, I'm not sure the answer you have given (although very good) is actually the answer to the question that was asked. I think this is the case because I have the same issue as described by the originator.

Perhaps I could add to the question to give what I think might be some helpful context. If I have a standalone audio track that provides a regular click for instance, like a metronome, and I want, in Fairlight, to record another audio track where on each click I say "now", I want the new audio track and the click track to line up perfectly so that all the clicks and all the "nows" happen at the same time, irrespective of what is happening visually, which may be completely separate. I can't do this with your solution.

I have found a way to workaround the latency in my headphones when I am recording my voice - I simply mute the track and slide one can of my headphones off an ear so I can hear both the already recorded audio and my own voice.

However when the recording is done, I need to nudge the resultant audio track to the left so that both audio tracks match. This is nothing to do with audio matching video - this is about audio matching audio and I have exactly the same issue.

I realise that Resolve is not designed as a DAW, but there are times when it needs to be used to record audio that exactly matches existing audio timing.

My interface is a Steinberg UR22 and even at the lowest possible latency setting, there is zero impact - I have tested various interface latency settings but the final result is exactly the same in Resolve. I have used the UR22 and the UR44 in many settings with Cubase and Reaper among others so I know my way around the kit.

The conclusion is that the issue can only be with Resolve.

I hope this helps and that there is a good answer out there.

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

Does macOS have something similar?

Does macOS have something similar?

I really don't know. As far as I know, Mac uses a driver called Core Audio. However, I don't understand how there could be latency on a Mac. Apple claims that their driver is the most efficient.

Thanks Mihai. Cheers.