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Recent Posts
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##### Movie/Multichannel audio / Re: mapping n channel to m channel
Last post by polemon -
Provide what you've done already, since you said "I always get stuck at one point".
If it's down to programming as such, I suggest stackoverflow.com, when it's about specific programming questions.

I'd probably map them out in a a graph and then use positional informaion to map them onto the new graph. Note that using pure trapezoidal information is often not desirable, you'd be downmixing onto all channels. I'd probably use the resulting adjacency matrix to figure out what channels are mixed by which source channels.

If say you're adding a new channel in between an existing channel, you can use its positional information to figure out the fractional downmix of the the two source channels, but this might be not what you get as a result. In the end, it'd be a lot down to checking.

The "easiest" approach would be to simply disregard the additional channels and map eight of the new 13 channels to their closest original counterpart, and leave the five new channels just silent.

Another Idea to consider is something like Dolby Atmos, which uses a large number of speakers but also a variable number of speakers between users. The audio source is provided in terms of "sound objects" and the mixing happens on the user's devices. The exact math to get a decent result is kept somewhat secret, it's not just a simple trapezoidal fractions based on a graph, with the distances of the speakers as their weight to the object location as an additional node. Although there are articles and tests, which suggest it isn't that much more
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##### 3rd Party Plugins - (fb2k) / Re: Georgia-ReBORN - A Clean foobar2000 Theme
Last post by CelticDog -
Any thoughts on adding custom colour themes to match Windows colours?
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##### General - (fb2k) / Re: Group by different bits per sample as one group
Last post by kode54 -
Set the bit depth manually instead of letting it do it automatically. It will be promoted to 24 bits on output, with the extra bits filled with padding.
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##### Development - (fb2k) / Re: Wishlist for new foobar functions
Last post by wojak -
Bitstream parsing does not allow for any modifications of the data, such as volume scaling, ReplayGain included. Or DSPs. Or allowing the OS to generate audio at the same time.
That is exactly what is needed. No modifications. Just passing unaltered signal to AVR in order to decode it in there. Volume management is done by the knob on the appliance. No other activities on the PC/foobar side - it serves only as a player ("transport" for files just like "old times" optical CD players or DVD players or BR players).
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##### 3rd Party Plugins - (fb2k) / Re: My components
Last post by arch21 -
Its been long time since I played Crash Bandicoot 3, I realize now latest psf decoder failed to play them. Thankfully there is an old version to check and play fine with it. Could you take a look at it? thanks. the psf got from joshw site.
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##### FLAC / Re: Why FLAC level 5 was chosen as the default and recommended setting?
Last post by krafty -
Thanks for the replies.

It took some time for me to figure out these graphs. Indeed, 4, 5, 6 are virtually the same decoding speed and use same CPU cycles.

Actually, I am using Monkey's Audio for quite a few albums, and I am testing them through network playback. FLAC is virtually indiscernible whatever setting I use, so yes, it's pretty fast. APE also decodes very well, but I notice that the apps that supports it, the "Insane" setting chokes a little in VLC (ATV4K) and the time progress halts and continues, then halts again - it is unable to map exactly where the timing is. However, it does decode. "Extra High" is better, does gapless in Poweramp (Android) and saves usually 12 to 22 MB for each album compared to FLAC -8. What annoys me is that MAC stores its own MD5 in the tag, which is not like FLAC (stores the MD5 PCM raw data).

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##### Lossless / Other Codecs / Re: Use of non 8/16/24-bit audio, like 12-bit audio
Last post by bennetng -
20-bit Cool Edit/Audition wav file attached. Basically just 20-bit padded to 24-bit with a slightly modified header.

Some programs cannot open these 20-bit files. A simple tool I made (oldsCool) can convert them to normal 24-bit files.
https://hydrogenaud.io/index.php?topic=97746.msg1004399#msg1004399
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##### General - (fb2k) / Re: Group by different bits per sample as one group
Last post by phunkydizco -
That is just how grouping works - when items have different tags / info, it creates a new group. I guess you could convert the one different track to make it the same bit depth as the rest (?)

That is a good idea. So I have to convert a 16bit FLAC / WAV 44.1kHz file to 24bit 44.1kHz. How can I do that with foobar2000?
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##### CUETools / Re: 2.19 - Object reference not set to an instance of an object.. - EAC 1.6 related?
Last post by c72578 -
Hi dpr,
thanks for checking and confirming, where the issue is coming from.
Wolfgang
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##### FLAC / Re: Why FLAC level 5 was chosen as the default and recommended setting?
Last post by Porcus -
FLAC -4 / -5 / -6 decode alike: http://www.audiograaf.nl/losslesstest/Lossless%20audio%20codec%20comparison%20-%20revision%204.pdf , figure 1.2. -7 and -8 use the -l 12 switch (higher order for the prediction), which in that study costs about 10 percent decoding speed. Shouldn't bother a Monkey's user at all ... *cough* I just learned something about that codec.

And saratoga could very well be right about the diminishing returns as well, it makes sense - though I don't know what were the actual considerations made.

But also take note that the lowest order modes have a special purpose. Explained at https://hydrogenaud.io/index.php?topic=120158.msg999755#msg999755 .  Maybe to facilitate decoding on special low-end hardware.
(-3 is also quite special, in that it does not do stereo decorrelation at all. Well you could in the early years call -8 "quite special" in that it included -e, and was for the particularly patient user.)

So FLAC actually has opportunities for ultra-light decoding, that no other codec has (well maybe if TBeck wanted, TAK could come close) - and so other codecs don't have switches in that end of the scale, to do what they are unable to. FLAC can, so FLAC has.