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How to convert 192 kHz FLAC to lower sampling rate with fb2k or other?, [was “192kHz FLAC”/TOS #6]
NuKleos
post Aug 13 2013, 13:15
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Hello audio nerds! Long time site lurker, first time posting (you guys do a good job initiating people to arcane audio-tech stuff smile.gif ). I have recently been meddling with some flac files @192khz, 24 bit. I am using foobar for playback and I would like to convert them to a lower bitrate form.

a) foobar's flac conversion panel gives no choice for bitrate (only bit depth), FLAC frontend doesn't run in my Windows 8 OS (some coder please make a new frontend for 1.3.0!). What do I need to get this thing done?
b) Is 96kHz still an overkill?
c) Any other tips for the process?
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hlloyge
post Aug 13 2013, 13:31
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- FLAC, by definition, is VBR, and you can't choose bitrate; you can choose larger compression level, but you won't gain much.
- Well, if you can hear frequencies up to 48 kHz, it's not.

Tips? You can use lossywav processing, for example smile.gif
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db1989
post Aug 13 2013, 13:34
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QUOTE (NuKleos @ Aug 13 2013, 13:15) *
a) foobar's flac conversion panel gives no choice for bitrate (only bit depth), FLAC frontend doesn't run in my Windows 8 OS (some coder please make a new frontend for 1.3.0!). What do I need to get this thing done?
Search for info on the various resampling DSPs that are available and put one in the signal chain of the Converter.
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Rollin
post Aug 13 2013, 14:26
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QUOTE (NuKleos @ Aug 13 2013, 16:15) *
FLAC frontend doesn't run in my Windows 8 OS (some coder please make a new frontend for 1.3.0!).

Have you tried this one - http://flacfrontend.sourceforge.net/ ?
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db1989
post Aug 13 2013, 14:38
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QUOTE (NuKleos @ Aug 13 2013, 13:15) *
FLAC frontend doesn't run in my Windows 8 OS (some coder please make a new frontend for 1.3.0!).
You were probably using the old version. It has recently been rebooted.. It was updated to 1.3.0 on page 3.
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birdie
post Aug 13 2013, 15:23
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Downsample them to 44.1KHz/16bit and be happy.

If they are true 192Khz/24bit files (not upsampled 44.1KHz) files then you'll be able to compress them to ~66% of their original size.
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pdq
post Aug 13 2013, 15:46
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QUOTE (birdie @ Aug 13 2013, 10:23) *
Downsample them to 44.1KHz/16bit and be happy.

If they are true 192Khz/24bit files (not upsampled 44.1KHz) files then you'll be able to compress them to ~66% of their original size.

Much smaller than that, I think.
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Propheticus
post Aug 13 2013, 15:52
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192kHz * 24 bit = 4608 kbps
44.1kHz * 16 bit = 705.6 kbps

705.6/4608 *100% = 15.31%

So yes, a lot smaller.

--- these are uncompressed bitrates per channel (so example is mono)

the ~65% figure fits PCM (wav) to Flac compression. But since the source is already compressed (Flac) and the target file will be compressed by Flac as well, we can assume similar compression and disregard this in comparing relative size.

This post has been edited by Propheticus: Aug 13 2013, 16:09
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NuKleos
post Aug 13 2013, 16:08
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Apologies for n00b mixing up of bit- with sample-rate. I grabed the newest FLAC frontend, it runs okay. Now, do I encode with Advanced>Extra command line options>--sample-rate=?
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saratoga
post Aug 13 2013, 16:18
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I think db1989 understood what you meant and suggested how to change the sample rate in the third post to this thread.
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db1989
post Aug 13 2013, 17:33
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QUOTE (NuKleos @ Aug 13 2013, 16:08) *
Now, do I encode with Advanced>Extra command line options>--sample-rate=?

No, because, as the official documentation states, such options are only for raw input files whose parameters FLAC cannot detect automatically: it cannot resample and will only be instructed on how to interpret the incoming data as existing samples.

As saratoga said, I already indicated how to resample using foobar2000 in my first reply here. FLAC Frontend is, by its own admission, just a simple facade for flac.exe, which therefore lacks any resampler of its own, so you would have to either resample the WAV before converting to FLAC via the Frontend, which would be laborious and probably lose tags… or you can just use another program as a frontend for FLAC, ensuring it provides resampling. foobar2000 has several plugins available that can resample for you. A safe bet is the SoX-based one by lvqcl.
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NuKleos
post Aug 13 2013, 23:26
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I tried SoX 14.4.1 but I couldn't get it to run after installation, so I used the foobar component version instead. I created a new conversion preset with a FLAC@defaults output and a Processing DSP with SoX@defaults. (db1989, I didn't use the DSP chain presets as you suggested because this field is blank in foobar 1.2.9) I got a 300MB@192kHz/24bit/2channel FLAC down to 93MB@48kHz/24bit/2channel. I understand this process is lossy in the mathematical sense but is the output equal to a FLAC that would have originally been generated @48kHz or at least @44.1kHz? I would ask you to bare with my ignorance a little more and help elucidate those fine points of audio engineering, as I am personally interested in music creation and I attempt to determine a point of balance, between being paranoid and a total hack.

This post has been edited by NuKleos: Aug 13 2013, 23:32
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saratoga
post Aug 14 2013, 00:57
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QUOTE (NuKleos @ Aug 13 2013, 18:26) *
I didn't use the DSP chain presets as you suggested because this field is blank in foobar 1.2.9)


Looking at foobar2000, I think you are referring to the file name box, which is used if you want to save DSP preset to a file. Its blank because if you're saving a preset file to disk you have to choose a file name. I don't think you actually want to do that though.

QUOTE (NuKleos @ Aug 13 2013, 18:26) *
I understand this process is lossy in the mathematical sense but is the output equal to a FLAC that would have originally been generated @48kHz or at least @44.1kHz?


No, because its lossy. If you care depends on what you're trying to do.

QUOTE (NuKleos @ Aug 13 2013, 18:26) *
I would ask you to bare with my ignorance a little more and help elucidate those fine points of audio engineering, as I am personally interested in music creation and I attempt to determine a point of balance, between being paranoid and a total hack.


What are you actually trying to do?
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db1989
post Aug 14 2013, 01:20
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QUOTE (NuKleos @ Aug 13 2013, 18:26) *
I understand this process is lossy in the mathematical sense but is the output equal to a FLAC that would have originally been generated @48kHz or at least @44.1kHz?

Equal in a literal sense? Almost certainly not. I mean, it could be. But more likely, a plurality of minor differences will emerge from the processes of analogue-to-digital conversion for recording vs. resampling purely in the digital realm.

The more relevant point is that, assuming a good ADC and a good resampler, their results would be perceptually equivalent.

It’s your choice whether you care more about not changing the file than about saving a lot of space. The former is not invalid from a perspective of archival, but don’t expect to reap any audible rewards from it. I suspect most people would probably advise keeping both: the original copy for archival and a downsampled copy for portable use and so forth.
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phofman
post Aug 14 2013, 06:29
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QUOTE (NuKleos @ Aug 14 2013, 00:26) *
I tried SoX 14.4.1 but I couldn't get it to run after installation,


What specifically did you do with sox? Your requirement is a matter of seconds with sox

CODE
sox input-file.flac -r 44100 output-file.flac
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NuKleos
post Aug 14 2013, 20:33
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I want to establish a threshold of sensible samplerate for lossless archival and for my own recordings (transparency aside, working with lossless avoids conversion rot). I expected that performing such a drastic downsampling of a recording from 192kHz to 48kHz would at least be 44.1kHz-equal (implying quality-wise, not bit-by-bit identical).


QUOTE (phofman @ Aug 14 2013, 08:29) *
What specifically did you do with sox? Your requirement is a matter of seconds with sox

CODE
sox input-file.flac -r 44100 output-file.flac


I clicked on the shortcuts created by installation but nothing happened. I suspected there was some command-line stuff I needed to do, but since there was a foobar component alternative I didn't look deeper into it. Is the standalone version better? Also is the foobar solution I used going to have a permanent effect with the DSP in my other conversion presets (the GUI is a little confusing with those options as it is)?

This post has been edited by NuKleos: Aug 14 2013, 20:36
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saratoga
post Aug 14 2013, 20:40
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QUOTE (NuKleos @ Aug 14 2013, 15:33) *
I expected that performing such a drastic downsampling of a recording from 192kHz to 48kHz would at least be 44.1kHz-equal (implying quality-wise, not bit-by-bit identical).


You'd have to go out of your way to find a resampler bad enough to have any effect on quality at all when making any sample rate change to an final rate >= 44.1khz.

edit:

QUOTE (NuKleos @ Aug 14 2013, 15:33) *
I clicked on the shortcuts created by installation but nothing happened.


Yes, sox is a command line program.



This post has been edited by saratoga: Aug 14 2013, 20:42
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jkauff
post Aug 15 2013, 13:51
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The excellent program TAudioConverter (see Validated News) uses latest SoX (as an option) and will very quickly resample your FLAC input files. It's got a nice GUI so you don't have to deal with command lines.
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NuKleos
post Aug 15 2013, 17:53
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Thanks for the pointers, jkauff. I already had TAC installed, using exclusively as a tool for extracting audio from video files without reencoding I completely overlooked its other nifty potentials.
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