I usually have 2-3 video tracks (.mp4/60FPS) and a single audio track (.wav), but I made a simple test problem with a single audio file (.wav or. I have gone through many other similar posts, but could not find a solution. But if the formats aren’t the same and each format is designed to eliminate different parts of the spectrum to achieve bitrate reductions, then we get a cumulative spectrum loss. Hello, I am having problems with crackling, popping, distorted, crunched paper kind of sounds in my exported videos. The second pass of Opus will look at a first pass of Opus and say “all the removable stuff has already been removed” and pass it on relatively unchanged. This test informally confirms the theory that multiple conversions in the same format lose less data than multiple conversions across different formats. The louder the playback, the more difference there was compared to the original PCM.Īlthough Opus and AC3 had the same average difference, AC3 had a wider variance around -51 dBFS, which means greater overall distortion than Opus.ĪAC, of course, had insanely loud playback to signify that it can’t encode a sine wave to save its life. Measure the dBFS of playback, which is the difference between the original sine wave and a double-converted sine wave.
Using track mutes, play the original PCM sine wave plus one of the inverted lossy sine waves.Align and invert the double-converted files from Step 3.Bring files from Steps 1 and 3 into Audacity.Convert files from Step 2 to Opus at 192k (YouTube’s audio bitrate for 4K uploads).Convert sine wave to AAC at 576k and AC3 at 640k and Opus at 512k (max allowed for each).Generate a 1 kHz -12 dBFS sine wave as pcm_s24le 48kHz.The test is a bit limited because it’s based on sine waves rather than real music, but it’s a start. I had a theory that Opus -> Opus would have less conversion loss than AC3 -> Opus, but never tried to prove it until now. I did another round of tests to measure distortion due to double format conversion.
Here is a Dropbox link for the test problem: I am always using aac audio codec and, for rate control, I’ve tried average, constant, and quality-based VBR. I tried the default YouTube settings, activating and de-activating both parallel processing and hardware encoder. The audio peak meter is always below -6, mostly below -10. The original file is clean (no noise) and the playback on editing mode is also normal except from lag-generated noises. I am running the latest version of Shotcut and already tried re-installing it. mp4 video and the problem still persists. I am having problems with crackling, popping, distorted, crunched paper kind of sounds in my exported videos.
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