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Recent Posts
1
FLAC / Re: FLAC v1.4.x Performance Tests
Last post by jaro1 -
Quote from: Wombat on
Compression -4 may be pretty good with a clang compile but high compression and multithreaded stress should be clearly faster with GCC builds.


Ok, i didn't try that.. but gladly, all others do it here. I choose compression "-4" based on the charts and own tests years ago, all resting compression steps ceased to exist for me, i forgot them completely.. and that continues to this day.
But I understand that the code here is primarily tested under stress conditions of higher compression, where the benefits are more pronounced.
4
3rd Party Plugins - (fb2k) / Re: Dynamic Range plugin
Last post by Henry66 -
I love this so much! Thanks for all your work on this (and other components) @Case.

After running DR Meter on an album, I see this:


Would it be possible to get all this data written to tags or to a log file (or both)?
6
FLAC / Re: FLAC v1.4.x Performance Tests
Last post by ktf -
Quote from: Porcus on
Ah. So you have just improved a criterion of the following kind:
"This predictor vector cannot possibly create any too big residual from a history of N-bit samples, so we can save time by bypassing the size checks that we are mathematically sure it would anyway pass"?
Yes, that is correct.
7
3rd Party Plugins - (fb2k) / Re: foo_vis_milk2
Last post by darkflame23 -
Couple of issues trying this here today.

Firstly, right clicking on the Milkdrop window instantly crashes Foobar (bug report sent).

Secondly, changing the default preset folder in the Advanced Preferences doesn't stick. Change it, hit Apply, OK, then go back to the same setting and it changes itself back to the default: C:\Users\...\AppData\Roaming\foobar2000-v2\milkdrop2\presets\
8
MP3 - General / Re: Resurrecting/Preserving the Helix MP3 encoder
Last post by maikmerten -
I'm glad to help, but clearly Case by far put in most of the work!

The downsampling issues were introduced with https://github.com/maikmerten/hmp3/commit/6d57f3ff9affbe4b5a1b4135dd4b0b1a8ca35ba5 - so at least it's known where to start looking into things.

(The encoder has two resamplers: srccf.cpp took 16 bit input and produced resampled 16 bit output, srccfb.cpp took 8 bit input and produced 16 bit output. Now one takes 32 bit floats, the other still takes 8 bit chars, I guess at one point one could drop the 8 bit thing)

---

edit: The resampling-code has various special cases, between which is chosen here: https://github.com/maikmerten/hmp3/blob/28a129a96b7bbc7a2cb68e76ff2381e1c070cc06/hmp3/src/srcc.cpp#L832

On the current dev branch:

44100 -> 48000: case 7, works
44100 -> 44100: case 5, works
44100 -> 32000: case 9, no audio
44100 -> 24000: case 9, no audio
44100 -> 22050: case 9, no audio
44100 -> 16000: segfault

32000 -> 48000: case 7, works
32000 -> 44100: case 7, works
32000 -> 32000: case 5, works
32000 -> 24000: case 8, no audio
32000 -> 22050: case 9, corrupted, stuttery audio
32000 -> 16000: case 8, corrupted, stuttery audio
9
Scientific Discussion / Re: Smart Cross(fade)
Last post by tasamagos -
I'm posting this because there seems to be a discussion on the exact same component I was hoping to get.
I agree with the first explanation of this discussion.
crossmixDSP is great, but it does not seem to calculate the mix-in point!
It is necessary to trigger playback ahead of time by the FADE IN INTERVAL in the explanatory diagram.
If someone could implement this, it would be greatly appreciated!
10
MP3 - General / Re: Resurrecting/Preserving the Helix MP3 encoder
Last post by jaro1 -
Quote from: maikmerten on
As far as I'm concerned, those builds should be plenty fine and can be called "stable".

There are further patches on the dev branch (Case's reworked progress printing, improved help messages, some safeguards against negative array indices in the encoder math), but nothing that invalidates the current builds.

edit: For those wondering why the dev branch isn't yet merged to the main/stable branch: I'm still trying to figure out why *downsampling* doesn't appear to work (in either main or dev). For instance, when requesting downsampling a 44.1 kHz input file to 32 kHz (hmp3 -A32000 test.wav test.mp3), the encoder goes haywire and crashes/segfaults. That's because when computing some array indices in l3math.c, the indices can get negative (the dev-branch guards against those negative array indices). However, that's most likely because the encoder is fed invalid audio data (with the negative-indices guards in place, I get silent mp3 files) and I haven't properly investigated out what's going on there.

Thanks for clarifying and for putting your and Case's effort into this project!