I've mixed the audio down to -10db for speech, and -20db for background music.Īs you probably could guess, I'm not a professional, and I'm not getting paid for any of my work. I believe the latter is related to Loudness, and so my attention immediately is drawn to the "Loudness" section in Fairlight, however, which of the numbers should I be looking at, and is this still related to my normalization and compression on audio, or is this a different thing to handle? THEN I start googling for Audio Levels and YouTube, and I start coming across the term "LUFS" which is new to me, but YouTube specifies -13 LUFS as some form of limit. I found one article that said for broadcast, -10db is the normal, whilst on youtube it's 0db. Others seem to favor the 0db mark, and mix their audio to be as loud as possible without capping. If you're doing voiceovers, the music underneath, should live at between -20 to -25db as a guide. Some people recommend mixing your audio to between -10 and -15db. Maybe someone else have this issue also.I am looking through some tutorials on how to properly set audio levels, I feel that I'm learning more and more, but a couple of things still confuse: So on linux to work with smartphone video only the AAC sound needs to be converted to PCM which is very fast actually.įor example if the videos (where filename contains mp4) are in the folder called "original" a level bellow, then this will convert the sound and copy to the current (1 level up) folder:įor i in $(ls original/|egrep -i "mp4") do echo $i ffmpeg -i original/$i -acodec pcm_s16le -vcodec copy. I thought this is related to the raw video and the VBR/CBR problem in my case too as it was mentioned above. So the symptom was very similar, however there were some strange sign like when using jog wheel to check the sound delay compared to the frame did not always gave the same result. And I had to fine tune the video I edited earlier as every cut was made for a constantly 1 second delayed audio so I had to move the sound track about a second. In my case the " alsa-plugins-pulseaudio " package had to be installed and all issues gone. I would expect to get the same result what I edited and checked several times and DRS is able to play.Īny idea? Are you sure the source file cause issue after the rendering playing everywhere but DRS?ĭon't say that I understand what problem you are dealing with, but as a matter of interest: Do you say that sound and picture are perfectly in sync when playing back the timeline in Resolve, but they are out of sync by over a second in the rendered file? But I cannot ask everybody to follow this. In that case the 1-1.1 sec delay would not help in the whole video just a part of that, however in my case 1-1.1 delay on audio solved the issue during my test. I read somewhere the VBR could cause this issues. If I change it with constant 1 sec (or maybe 1.1) then there is no slip in the audio. Then I took the VLC player and tried to change the "audio track syncronization" paramter. There the video is behind the audio about 1-1.1 sec.įirst I thought I cannot check the exact delay as if I take the same rendered video (with audio slip) and try to play within DRS there is no such issue, everything is as per my plan. I even attempted to render with "bypass re-encode when possible" option cleared (so I expected to re-encode everything).Ĭhecked on the same linux and uploaded to youtube and watch on both linux and mobile phone. Rendering output QuickTime with MPEG codec, MPEG4 Video type, however I tried several such as DNxHD codec, without success as I have delay everywhere and only DRS is able to play the rendered file correctly. I have several videos together with a music from a different source. This audio/video sync problem looks like a DRS bug and not a video source file issue. Sources from mobile phone, music from a wav file. Convert to CFR.ġ7,4,3 Build 10, DaVinciResolve Studio (DRS), Linux. Uli Plank wrote:It's all about your sources, not the Deliver settings.
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