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Seems the FLAC maintainer hasn't got time/motivation to work on FLAC. He merged a bunch of PRs on March 15 2021, and his last commits before that are from May 14 2020. Someone else involved with Xiph/Mozilla has merged a few PRs, but the last activity (merging, commenting etc.) by anyone with write access to the FLAC repository has been well over 6 months ago.
... your tagged album has "2011, 2011" in %date%. Just remove it from the %date% tag.
The Date field only has one "2011" in it. Could it be showing the Original Date as well?
Weird: I changed just the Original Date to "2029" and now I have "2029, 2029"
- UTF-8 on Windows change (which is what this topic is about)
- Compression improvement patch, discussed on HA here
- Bug fix, discussed on HA here
- Fixed subframe speed improvement
- some build system improvement (which made it possible for me to build this DLL with MinGW)
What's taking so long for these to be merged, anyway?
Also, could you provide flac.exe as well instead of just a .dll?
What do you do for Various Artists?
You're also having to use a lot of metadata (including catalog number) in the filenames just to keep them unique.
Whilst I do use albumartist and album as folder names I only use discnumber.tracknumber in the track names because given all the characters that can't be used in filenames they'll never perfectly reflect what I've got in the tags, so what's the point. For albumartist and album I translate (via foobar) the characters that can't be used into the closest unicode equivalent so it at least looks correct, but I have often thought about just hashing them just to make it even simpler.
Would be interested to hear about your experience of using that structure with various software packages.
AAC userbase had a large jump.
I don't see this, tbh. The AAC curve looks rather flat, and the fact that it gently wiggles as much upward as it does downward is apparent from the fact that the last datapoint is almost at the exact same height as the first one (at around 25-ish %). The only real "large jump" in these graphs I see is for Opus from 2014 to 2016.
I think it's also worth noting the increase in the last three years of the number of folks saying they don't use lossy as their main audio format. Do we have a trend here, perhaps?
If I had to guess I would say that it's an effect of continued increase of available storage space, but I honestly can't imagine this going very far. The fact of the matter is that no matter how much storage space you have available, a ten-fold (or even twelve-fold) reduction in the amount of storage taken up by your music library will always be a ten(twelve)-fold reduction in the amount of storage space taken up by your music library. Sure, having less storage space available creates an additional pressure to prefer lossy compression over lossless, but something taking up ten or more times more storage space for absolutely no perceptible benefit is equally wasteful regardless of that factor.
I have a set of videos that have audio and video and I'm adding a second audio stream to that. The original videos have excess stuff at the beginning of various lengths. If this length was always the same, I'd have no trouble to batch program this.
What I'm doing now is I playback the original video in MPC-HC with milliseconds visible, find the spot where the second sound source should begin, note the time and then use that time as a parameter for ffmpeg to cut the excess. Sometimes I need to repeate the process a few times, because I don't always get it right the first time. Cumersome, especially since I have ~100 videos to process.
I have tried Shotcut and the likes, but since I'm not into video editing, I can't figure out how to do this. So, I need either a guide for dummies for Shotcut or some other software suggestion that can help me.
I'd like to open the existing video, add the second audio and be able to just drag the second audio to the place where it's supposed to be and then export that as a new video.
Shotcut, DaVinci Resolve, etc. all promise this is what you can do, but I'm just not smart enough, apparently.
It has downloaded msys2 and ming64, gcc etc.,. After that it has compiled source code of many audio tools and ffmpeg.
First I tried to create ffmpeg tool but after compiling fdkaac and opusenc the process has been stopped due to errors.
Many of you may be struggling to compile source code of fdkaac and opusenc, ffmpeg etc.,. Including me as I tried to compile fdkaac source code in visual studio but not succeeded.
I am suggesting to all of you to try this tool called media auto build suite.
I have tested m4a files generated by various versions of fdkaac ie., One downloaded from media fire, compiled by msys2, old fdkaac etc.,. My conclusion is all versions generated little differences in file size but quality is same of all output audio files.
My doubt is all source mp3 files are of 16 bit depth. What happens when I convert these mp3 files to m4a files by specifying bit depth as 24 bit.
I think that encoding to 24 bit may improve the quality of output m4a files.
For the time being, I add SKIP tag with "30-" to all track of this genre.
I would like to automatically obtain this duration when genre is interlude (without adding "SKIP" tag).
Anyway to get this result ?
If not today, could it possible to add this feature ?
(for instance with a "virtual" SKIP tag set up in global preference for skip tracks)
With your test method I assume results are going to be "larger files for larger blocksizes"