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Recent Posts
FLAC / Re: FLAC unicode patch: some help wanted
Last post by ktf -
The point of this discussion is to test the DLL interface for compatibility, so providing an exe would be off-topic and probably derail the discussion.

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.
FLAC / Re: FLAC unicode patch: some help wanted
Last post by doccolinni -
- 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?
General Audio / Re: Directory structure for organizing FLAC files.
Last post by SimBun -
You say you're not having a problem yet you've had to introduce an extra folder layer to alphabetize your tracks to keep the file counts manageable, and no doubt as your collection grows you may need to manually tweak that further depending on how many artists/albums you have fall under each letter (at the moment you have 2 folders for A and 4 for B). What decides where to put the files i.e. do you do it manually, or do you have to maintain some code to handle it for you?
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.
Polls / Re: 2013-21-Lossy-Format-Poll Graph
Last post by doccolinni -
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.
General A/V / Looking for a free tool to sync second audio track to video
Last post by psycho -
I am looking for a free tool to sync second audio track to video.
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.
General Audio / Re: Getting and running fdkaac and opusenc encoders for dbpoweramp and foobar.
Last post by rupeshforu3 -
Hi at present I am using fdkaac encoder latest which is obtained from media auto build suite.

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.

3rd Party Plugins - (fb2k) / Re: foo_skip: skip tracks that match a specified search query
Last post by ubboo -
I have to limit duration to 30 second when "genre" tag is set to "interlude".

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)
Lossless / Other Codecs / Re: "Tested": codecs at same-signal-different-sample-rate (boring result!)
Last post by ktf -
Perhaps testing ffmpeg's flac encoder will give more interesting results: that changes the blocksize depending on the samplerate by trying to keep the blocksize a certain time, sticking to the default blocksizes. That time is 105ms for most compression levels, which translates to 4608 @ 44.1kHz, 8192 @ 96kHz, 16384 @ 192kHz and 32768 @ 384kHz

With your test method I assume results are going to be "larger files for larger blocksizes"