![]() ![]() Apart from the audio features described, the new version is focused mostly on the new cut page, which continues to evolve around Blackmagic's declared goal of making it "the world’s fastest editor." Finally, one the audio front, the company introduced support for displaying and persisting individual audio effect windows on top of other windows.Īs the large film studios that use DaVinci Resolve for most of the world’s feature films need to get their systems updated before the next programming season starts, this new public beta lets Blackmagic Design continue to develop new ideas without causing delays for the large studios who need a finished and well tested release of DaVinci Resolve. There's also an improvement in performance when loading and switching timelines, when generating waveform profiles for nested timelines, and peak hold and decay behavior for meters in FairlightFX plugins. ![]() The audio user interface now also benefits of many improvements, such as support for clipping indicator for main buses on the Fairlight mixer, support for enabling extended bounce tail durations from the option menu for audio clips with reverb and other tail effects, support for customizing the tail duration threshold for reverb and other tail effects, and support for automatically enabling Insert In for an audio track when manually patching track inputs. The new public beta allows the Australian company to continue development of the software, announcing among many other improvements the support for 96 KHz and 192 KHz sample rates at a project level for DaVinci Resolve Studio. But it ensures I don't accidentally select 16 when my audio is actually 24, or select 24 if my audio is 32bit.įor Windows users I would recommend always using the Quicktime output, then remuxing to MP4 using ffmpeg or Shutter Encoder.Bringing multiple improvements in editing, color, Fusion and the Fairlight audio pages, DaVinci Resolve 16 is now shipping, while DaVinci Resolve 16.1 is now available in public beta. If my audio is lower bit depth (24 or 16), it will make no difference to the output. Therefore if in future I use MP4 with AAC I will always be choosing 32bit as the source option. I assume choosing 24bit when source audio is 32bit would do likewise. This implies that choosing 16bit when the source audio is 24 will involve a conversion down to 16bit which can lose data. The second render was a 100% null there was no difference between 24bit and 32bit renders.īut the 16bit and 24bit test did show some minor differences between those two outputs. I then did two test renders: 16 + 24bit (inverted phase), and 24bit (inverted phase) + 32bit. I then imported these exports into Reaper and inverted the phase of the 24-bit version. My source audio in Resolve for these exports was 24bit PCM. I tested on macOS, doing three test renders where I rendered out a video with WAV dialogue track to MP4 H264 AAC with the AAC export bit depth set to: ![]() I couldn't quickly find docs for the macOS equivalent With ffmpeg you'd certainly do that, and it would use whatever source audio you had to create the AAC.īut as we see from the Windows AAC docs, it seems these OS encoders require PCM input of a fixed bit depth, so there's an unfortunate but necessary conversion step first. So logically you would think one should just pass in the source audio - at its native bit depth. I think it's usually regarded as being 32-bit float? AAC, like MP3, has a variable bit depth that changes from sample to sample. In general, I was rather confused why Windows was limited to 16bit, and why on macOS you have to choose between 16 vs 24 vs 32. Perhaps there's some technical or legal reason for this use of two different encoders, but on the surface it does seem a bit odd. That does raise the question of: why are different encoders used, and couldn't the encoder that's used for Quicktime also be used for MP4? Especially as in both cases it's ffmpeg's LibAV that writes the container. I guess in the Quicktime case a different encoder must be used. The Microsoft Media Foundation AAC encoder is a Media Foundation Transform that encodes Advanced Audio Coding (AAC) Low Complexity (LC) profile, as defined by ISO/IEC 13818-7 (MPEG-2 Audio Part 7). ![]()
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