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1
Opus / Re: Opus 1.3-beta is here
Last post by Nott -
I want to convert my ~2k files with AC3 5.1 and AC3 2.0, should I wait for opus 1.3? What settings do you guys recommend to opus 5.1 and opus 2.0 to achieve transparency ? File size isn't really a issue here, I just want to have a library fully open source

If file size isn't the problem: Keep the your files as-is, in AC3. You've got nothing to gain quality wise, only to lose. AC3 is so old it *may* already be "free" in terms of patents having expired or being close to expiring and there are open-source implementations.

I still care about streaming my stuff, and AC3 isn't supported on most platforms I stream : browsers, phone.  Only my streaming boxes like chromecast/roku can accept AC3 but I think plex still transcode it lol...  and opus is "future proof"

And I thought there is a transparent quality where the quality will be identical?
2
Opus / Re: Opus 1.3-beta is here
Last post by maikmerten -
I want to convert my ~2k files with AC3 5.1 and AC3 2.0, should I wait for opus 1.3? What settings do you guys recommend to opus 5.1 and opus 2.0 to achieve transparency ? File size isn't really a issue here, I just want to have a library fully open source

If file size isn't the problem: Keep the your files as-is, in AC3. You've got nothing to gain quality wise, only to lose. AC3 is so old it *may* already be "free" in terms of patents having expired or being close to expiring and there are open-source implementations.
3
For smooth loudness changes use a compressor for the microdynamics, rather than a "leveller" for the macrodynamics.
4
i will check it, thanks.

and the R128 normalization. it delivers some sort of real radio feeling somehow. it is definetely a great option for parties (in my case).

i am not a programmer, even if interested, but when the lookahead is seeing a louder part coming, the sound gets noticeable reduced in also noticeable steps.

if i could program this, i would add an exponential curve that gets applied to the part where the sound gets loude (or quiet), i mean to deliver a "smooth" loudness change.

it sounds pretty easy to look at this with simple mathmatical terms, but like i said i do not know code (except some html), so i do not know if this is going to be lots of work to implement. but it would be great!
5
Well, that was just posted in totally wrong thread... Currently the preferred way is to drag & drop them - both in order to change their order and in order to enable / disable them (dragging between active and unused lists). And as far as I understand you, this is something that makes impossible to easily use DSP manager with screen readers... Despite I don't use them, I would also welcome restoration of Add / Remove and Up / Down buttons, as an alternative option for managing DSP chains.
6
hey again!

when i load the .tags file that i created for a folder containing gamecube .adx files, m-tags uses the folder name as the title for "all" .adx files.

which is strange because it worked well with my .flac files for example.

any idea how to fix that?

EDIT: i see. it adds |1 to the end of the file name: "@" : "101 - title.adx|1",

it wouldn't be "that" problem to delete this automatically, but also interesting why m-tags is doing that for .adx files.

EDIT2: i will check the dedicated m-tags section soon as well.


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For M-Tag you should check for its dedicated thread here at HA, and maybe there is an answer how to transfer tags between databases (file based and M-Tags based). Thread should be linked on component's download page.
8
Wow, I've been running CUETools 2.1.5 in kvm (Linux VM) with XP for years. Running CUETools 2.1.6 under wine 3.0 on the same system is 7.5x faster (6 seconds vs 45 seconds) for a verify on the same album. Fortunately I did not encounter any crashes using wine, all the config screens worked fine. The font size is microscopically small on my 4K monitor though.
9
The cuesheet format is used when joining multiple files to a single file in foobar2000's converter.

Cuesheets are designed to address audio as 1/75 of second for a single frame.  That's 588 samples for standard CD audio (44.1 KHz).  If we take 96 KHz and divide it by 75, you get 1,280 samples per frame. 

The source in particular as with any self-recorded albums or downloaded music often isn't set up with this in mind, so you get time variations like this.  If this actually happened with a CD rip from a lossless source then I consider it a bug but since this doesn't seem to be the case here, it's simply the limitations of the cuesheet format manifesting itself.

The time format for CDs is MM:SS:FF which is what the cuesheet format itself uses.

The workaround is to use output as individual files option instead when using the converter.  Track lengths will always be the same here, assuming it's just a straight conversion between two different lossless formats (compressed or uncompressed).  This should avoid rounding errors.
10
you could use kode54's normalizer R128 DSP which applies normalization/ReplayGain in a non destructive manner in realtime on files, without them being tagged as such. You can even do so on Internet streams or emulated content.
It uses a bit of lookahead, and tries to keep the volume level from getting too loud, or staying too quiet for too long. It doesn't compress dynamics in a rapid fashion, though, or in a multi-band equalizer sort of way, either.

hey that's just awesome. a real time normalization is just exatly what i need. i will check this out right now. sure, in real time the every audion signal information, whether it is a quiet part or a loud part, will just be normalized without the differences between. but let me check it.

btw i tried m-tags and it is great. i just checked it. i am sure there is a way to read the tags from the files and copy them fast into the m-tag file, since foobar shows only tags from the m-tag file when using m-tag.