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Upsample 44.1 24-bit to 48 24-bit for DVD
testyou
post Sep 21 2013, 20:53
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QUOTE (LTD @ Sep 21 2013, 12:10) *
what does the HydrogenAudio community think of SACD, DVD-A, and HD audio tracks?

Sometimes these releases contain different and better (such as larger dynamic range) mastering of the material, but sometimes not. This has been mentioned many times.
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stephan_g
post Sep 21 2013, 22:21
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Concerning the problem of track transitions mentioned earlier, Foobar2000's Converter has a "Don't reset DSP between tracks" option that should take care of that. It also is a teeny tiny bit more comfy to operate than command-line SoX. (The SoX resampler plugin for Foobar should be equivalent in quality, AFAIK.)

I would recommend applying a few dB of negative gain when resampling. This can be done by making use of the ReplayGain preamp setting, or applying album gain if the source material has been RG scanned. Peak levels may increase by up to 2-3 dB or so upon resampling of brickwalled material (which commonly peaks within a fraction of a dB short of fullscale already), and you don't want to needlessly introduce extra clipping.
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Wombat
post Sep 23 2013, 10:30
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QUOTE (stephan_g @ Sep 21 2013, 23:21) *
Concerning the problem of track transitions mentioned earlier, Foobar2000's Converter has a "Don't reset DSP between tracks" option that should take care of that. It also is a teeny tiny bit more comfy to operate than command-line SoX. (The SoX resampler plugin for Foobar should be equivalent in quality, AFAIK.)

This really works. Unfortunately foobar has a different opinion about dither as me and i still prefer SoX itself. Now if there was a way to use SoX in the way foobar does use its dsp.

QUOTE (stephan_g @ Sep 21 2013, 23:21) *
I would recommend applying a few dB of negative gain when resampling. This can be done by making use of the ReplayGain preamp setting, or applying album gain if the source material has been RG scanned. Peak levels may increase by up to 2-3 dB or so upon resampling of brickwalled material (which commonly peaks within a fraction of a dB short of fullscale already), and you don't want to needlessly introduce extra clipping.

With a gentle filter for resampling i often need not more then 0.5dB and sometimes none at all with my type of music that is seldom compressed to death.

This post has been edited by Wombat: Sep 23 2013, 10:30
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phofman
post Sep 23 2013, 11:41
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QUOTE (Wombat @ Sep 23 2013, 11:30) *
Now if there was a way to use SoX in the way foobar does use its dsp.


I do not know if that is a solution but you can feed sox with data from standard input. In this mode sox starts the dsp engine only once. Very listing all the files as input would produce the same result - keep the dsp chain running/"hot" between the processed files. We would have to check the source code or ask at sox mailinglist.
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Wombat
post Sep 23 2013, 14:16
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Maybe i have to fiddle a bit but telling frontah to use stdin with the command line changes nothing, clicks are there.
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phofman
post Sep 23 2013, 14:31
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Do the tracks tested make smooth transition, or are there discontinuities?
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Wombat
post Sep 23 2013, 14:35
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QUOTE (phofman @ Sep 23 2013, 15:31) *
Do the tracks tested make smooth transition, or are there discontinuities?

My files i resample for this test have no smooth transition. I go from 24/48 to 16/44.1 On smooth transitions there is no problem.
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phofman
post Sep 23 2013, 14:42
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Then a big resampled track will have clicks, just like the unresampled (only concatenated) one does.

What I meant by stdin is concatenating the tracks first and resampling one large track at once, to avoid resampling discontinuities at edges of each track (that "Don't reset DSP between tracks" functionality).

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Wombat
post Sep 23 2013, 14:59
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Exactly that is what i want to avoid. foobar works with just setting an option. I think there should be some solution for Sox itself.
btw. maybe we make another thread out of these last posts?
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saratoga
post Sep 23 2013, 15:02
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QUOTE (Wombat @ Sep 23 2013, 09:59) *
Exactly that is what i want to avoid. foobar works with just setting an option. I think there should be some solution for Sox itself.
btw. maybe we make another thread out of these last posts?


I don't see how that would be possible for a command line program unless you're going to list every single file in the album on the command line.
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Wombat
post Sep 23 2013, 15:11
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QUOTE (saratoga @ Sep 23 2013, 16:02) *
I don't see how that would be possible for a command line program unless you're going to list every single file in the album on the command line.

http://www.hydrogenaudio.org/forums/index....st&p=845272
Something like this but i am no expert.

This post has been edited by Wombat: Sep 23 2013, 15:11
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john33
post Sep 23 2013, 15:21
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Slightly off-topic, but perhaps helpful, I have just uploaded to Rarewares, Others page, win32 and 64 bit compiles of SoXRDrop. Essentially, this is SRCdrop recompiled using the SoXR libraries. Much smaller compiles and much faster in execution. smile.gif


--------------------
John
----------------------------------------------------------------
My compiles and utilities are at http://www.rarewares.org/
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saratoga
post Sep 23 2013, 15:23
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QUOTE (Wombat @ Sep 23 2013, 10:11) *
QUOTE (saratoga @ Sep 23 2013, 16:02) *
I don't see how that would be possible for a command line program unless you're going to list every single file in the album on the command line.

http://www.hydrogenaudio.org/forums/index....st&p=845272
Something like this but i am no expert.


That would help, but I think to be completely gapless, you would need to know at least a few samples from both tracks. You can actually make filters that do not have edge ringing (see matlab's filtfilt command for instance), but theres still the problem of computing an interpolated sample that falls between the last sample of the first track and the first sample of the second track.

Its more accurate (and probably much, much faster) to just use a GUI program that can feed in the tracks as one PCM stream and split them afterwards.
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phofman
post Sep 23 2013, 16:53
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QUOTE (saratoga @ Sep 23 2013, 16:02) *
I don't see how that would be possible for a command line program unless you're going to list every single file in the album on the command line.


CODE
H:\Music\audio>( for /R %f in (.\*.flac) do c:\sox\sox.exe "%f" -t raw -r 44100 -b 24 -s -  ) | C:\sox\sox -t raw -b 24 -s -c 2 -r 44100 - -r 48000 -b 24 c:\output.flac


Basically loop through all flac files, convert to raw with common format (rate, bitwidth), output to stdout and feed this whole subshell output to sox doing the final whatever-you-need.

Oh, how much I hate windows scripting, being spoilt by linux bash smile.gif

This post has been edited by phofman: Sep 23 2013, 16:54
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saratoga
post Sep 23 2013, 17:24
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QUOTE (phofman @ Sep 23 2013, 11:53) *
Basically loop through all flac files, convert to raw with common format (rate, bitwidth), output to stdout and feed this whole subshell output to sox doing the final whatever-you-need.


I'm not great with sox, but doesn't that give you one giant flac file? I think you would need to also enter commands to then split the resulting file after resampling.
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phofman
post Sep 23 2013, 17:36
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I assumed you wanted to play the many tracks at once, resampled.

What does that checkbox actually do? Does it affect the resampling module too? Which one? I cannot find any authoritative information https://www.google.com/search?client=ubuntu...%22&start=0
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saratoga
post Sep 23 2013, 17:46
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QUOTE (phofman @ Sep 23 2013, 12:36) *
I assumed you wanted to play the many tracks at once, resampled.


Yeah, but it should output them into the same number of files as you started with.

QUOTE (phofman @ Sep 23 2013, 12:36) *
What does that checkbox actually do? Does it affect the resampling module too? Which one? I cannot find any authoritative information https://www.google.com/search?client=ubuntu...%22&start=0


Resampling is a DSP in foobar, so it should affect it. I don't have the foobar source code, but as far as I know it does exactly what it says: retain the state in the DSP between tracks, so anything with memory will still have samples from the previous track stored in it when it starts a new track.

We do something similar in rockbox, where things like the resampler retain state across track transitions so long as the sampling rate does not change.
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phofman
post Sep 23 2013, 18:52
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Well, then sox is not a right tool for this task indeed. It is certainly doable (though a PITA in windows) but if the solution is just a click away in foobar, no reason to use something less suitable.
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