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Best Free-software or no charge proprietary tool to downsample to 44.1
Morality124
post Apr 28 2012, 23:16
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I have some audio files I've ripped that I want to down-sample to 44.1khz/16-bit format so they can burned to CDs. The audio files in questions are 24-bit (obv) and range from 96-192 kHz. I don't have any commercial proprietary audio software like Adobe Audition, but I do have the Free software programs Audacity and SoX at the moment, as well as EAC and ImgBurn. Would down-sampling using either of these programs be advisable, and if so, what is best settings and/or method(s) to do so? If not, what would be the recommended program and settings/methods? Thanks!
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skamp
post Apr 28 2012, 23:47
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I use this:
CODE
sox -G "in.wav" -b 16 "out.wav" rate -v 44100 dither -a -s


See the manual for an explanation of the parameters and if you want to tweak them.


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Apesbrain
post Apr 29 2012, 01:43
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If you have a number of files to convert you may find it easier to use foobar2000 with SoX resampler plugin.
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m45t3r
post Apr 29 2012, 04:56
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The "Secret Rabbit Code" is a very good quality resampler and is completely Free Software (licensed as GPL). On the webpage you can found a plugin to mass convert files using foobar2000.
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Morality124
post Apr 29 2012, 06:09
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Hmmm. I actually do have Foobar 2000 also, though I never thought of using it for this work (I also have the SoX plugin mentioned, but I was actually trying to use it for a different purpose). What exactly differs in quality between the methods mentioned?
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Morality124
post Apr 29 2012, 07:52
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To clarify, what is the consequence of using a lower quality down-sampler? Abnormalities? A particular distortion?

This post has been edited by Morality124: Apr 29 2012, 07:53
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Juha
post Apr 29 2012, 09:00
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If your files are type of WAV already then try Voxengo r8brain free.

Check some SRC Comparisons - http://src.infinitewave.ca/

Juha
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AndyH-ha
post Apr 29 2012, 09:10
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There are some varying opinions on what is the "best" result for both resampling and for bit depth reduction. For bit depth this has mainly to do with the dither used. Unless something really gross is done, however, (and maybe even then) it is rather unlikely you could hear the difference.


For sample rate you can see some visual representations a
http://src.infinitewave.ca/
Look at the sweep tone conversion using Adobe Audition with filters and say Ableton Live 7. The Audition result is more ideal. There are other interesting displays. As with bit depth reduction, being able to hear a difference does not seem to be an established fact.
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stephan_g
post Apr 29 2012, 22:44
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QUOTE (Morality124 @ Apr 29 2012, 06:09) *
Hmmm. I actually do have Foobar 2000 also, though I never thought of using it for this work (I also have the SoX plugin mentioned, but I was actually trying to use it for a different purpose). What exactly differs in quality between the methods mentioned?

Anything of decent quality (I don't think people would bother with anything less here) should mainly differ in user experience. The only thing that may be potentially critical is clipping during downsampling - sometimes even hi-rez material is brickwalled and likely to require a level reduction.

In order to safeguard against this problem, I'd suggest the old SSRC "twopass" approach - downsample to 32-bit float first, determine peak levels, then dither to 16 bit.

Foobar should do all of that quite nicely with two converter runs and one Replaygain scan (multiple albums) in between, for just about as many files per pass as you like. I have verified that "32-bit" output does in fact produce float samples (v1.1.11). It's obviously helpful if Fb2k is set up to plainly show RG information in selection info, which makes it easy to determine total peak level.

If you find that downsampling produces levels in excess of 0.999969, the album(s) in question will require a small level reduction. That again can be taken care of with a converter run... enabling Replaygain in album mode with "prevent clipping" (only) should do the trick. Actually one could integrate that with converter run #2 (dithering to 16 bit), now that I think about it - it would only affect material that would clip otherwise, and no manual inspection of peak levels would be required.
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Arnold B. Kruege...
post Apr 30 2012, 14:40
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QUOTE (AndyH-ha @ Apr 29 2012, 04:10) *
There are some varying opinions on what is the "best" result for both resampling and for bit depth reduction. For bit depth this has mainly to do with the dither used. Unless something really gross is done, however, (and maybe even then) it is rather unlikely you could hear the difference.


For sample rate you can see some visual representations a
http://src.infinitewave.ca/
Look at the sweep tone conversion using Adobe Audition with filters and say Ableton Live 7. The Audition result is more ideal. There are other interesting displays. As with bit depth reduction, being able to hear a difference does not seem to be an established fact.


It's interesting to see how CEP/Audition which used to get no respect among pros and semi-pros always did a lot of things right, even when more "pro", prestigious, and expensive software like some old releases of Pro Tools didn't even do the basics right.

This post has been edited by Arnold B. Krueger: Apr 30 2012, 14:40
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stephan_g
post May 2 2012, 09:55
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QUOTE (Arnold B. Krueger @ Apr 30 2012, 14:40) *
It's interesting to see how CEP/Audition which used to get no respect among pros and semi-pros always did a lot of things right, even when more "pro", prestigious, and expensive software like some old releases of Pro Tools didn't even do the basics right.

Some of those look quite scary indeed. In some cases you have to wonder how much of an understanding of SRC the programmers had.

I suppose it's a case of "black box syndrome". Musicians are used to tools that do something which has to be determined empirically. Verification of function against a spec is not usually an issue. Now obviously, an SRC's function is pretty well-defined and verifiable, as with a speaker amplifier. I guess that has yet to sink in. Hence, people are basically "flying blind" and questioning of performance (negative feedback) rarely takes place.

This post has been edited by stephan_g: May 2 2012, 09:59
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phofman
post May 2 2012, 10:19
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SoX results on http://src.infinitewave.ca are very decent, considering the price and availability :-)
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rednyrg721
post May 2 2012, 11:30
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Audacity 2.0 results are much worse than Audacity 1.39 results for some reason.
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stephan_g
post May 2 2012, 17:05
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That one puzzled me, too. I think I'll ask the devs...

EDIT: Ah, so it seems the 1.3.9 results were obtained using libsamplerate, while 2.0 apparently uses libresample (whose "high quality" setting isn't a match for libsamplerate's "best" by a long shot). It looks like there were some issues with libsamplerate, like this one and this. Seems like a Linux thing... I'd have to check how the Windows build is configured. ... The same, it seems. Hmm. I don't think I like that.

EDIT^2: Ah, I think that's the problem:
Resample.h
CODE
    libsamplerate, written by Erik de Castro Lopo.  GPL.  The author
    of libsamplerate requests that you not distribute a binary version
    of Audacity that links to libsamplerate and also has plug-in support.

And as we all know, Nyquist and VST plugins are supported by default (VST was added after 1.3.9 then I guess, can't remember). So yet another "private build only" feature (next to ASIO with its proprietary license)... *sigh*

Guess I'll reconfigure that and update my build.

This post has been edited by stephan_g: May 2 2012, 17:40
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john33
post May 2 2012, 17:39
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If you're Windows based and like libsamplerate, you could try srcdrop on the 'Others' page at Rarewares. smile.gif


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lvqcl
post May 2 2012, 17:55
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QUOTE (stephan_g @ May 2 2012, 20:05) *
EDIT^2: Ah, I think that's the problem:

This issue is also mentioned at http://wiki.audacityteam.org/wiki/Libresample
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stephan_g
post May 2 2012, 21:59
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QUOTE (john33 @ May 2 2012, 17:39) *
If you're Windows based and like libsamplerate, you could try srcdrop on the 'Others' page at Rarewares. smile.gif

'fraid I'm thoroughly spoiled by the Foobar2000 / SoX resampler combo (best quality AND ease of use)...

(Before that, I'd typically use SSRC. The ringing around filter cutoff makes it a less-than-ideal choice if sample rates significantly below 44.1 are involved, but other than that it still performs well.)
QUOTE (lvqcl @ May 2 2012, 17:55) *
This issue is also mentioned at http://wiki.audacityteam.org/wiki/Libresample

I think I even read that one back when I first compiled Audacity, but - not being aware of the quality implications at the time - had all but forgotten about it.
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Morality124
post May 3 2012, 01:28
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QUOTE (stephan_g @ May 2 2012, 22:59) *
QUOTE (john33 @ May 2 2012, 17:39) *
If you're Windows based and like libsamplerate, you could try srcdrop on the 'Others' page at Rarewares. smile.gif

'fraid I'm thoroughly spoiled by the Foobar2000 / SoX resampler combo (best quality AND ease of use)...


If I use SoX from the command line (which I have for de-emphasis, stereo channel flipping, and inverted absolute phase), will this produce the same results as the method you are using?
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bandpass
post May 3 2012, 05:45
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They're essentially the same. Architectural differences between the standalone version and the Foobar port might result in some variation in the least-significant bit of the converted audio, but such differences are negligible and can and do occur for other reasons too.

Skamp's sox command line suggestion above should produce good results. (Though you could probably save a little CPU time by dropping the -v option—it's really intended for cases when both input and output audio is at least 24-bit.)
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phofman
post May 3 2012, 12:44
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QUOTE (Morality124 @ May 3 2012, 02:28) *
If I use SoX from the command line (which I have for de-emphasis, stereo channel flipping, and inverted absolute phase), will this produce the same results as the method you are using?


The foobar sox plugin is code taken from the sox rate effect and modified to fit foobar plugin API. I think I saw in the foobar plugin source code there was support for modern CPU directives added too (SSE, etc.) I do not think I saw that in the original sox sourcecode (it is kept portable to other CPU architectures). I did not check if the results are identical.

If sox rate.c code gets modified in the future (for whatever reason), the foobar plugin functionality can start deviating. That does not mean any problem at all, just an information.

This post has been edited by phofman: May 3 2012, 12:48
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ArtistofMind
post Jan 11 2014, 02:26
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Hi there,

I am keenly interested in using the Resampler (SoX) DSP component for the function specified by the OP (batch conversion of a folder full of "HD" files to CD or DVD spec, where appropriate). This function was implied by Apesbrain in post #3, who seems to suggest that Foobar can convert a large number of files for me; yet when I look at the Settings menu of the component, it has options for playback but not source file conversion.

To clarify: the plugin works great for downsampling the currently-playing file in realtime; but that is a CPU-intensive process, and my source files are still huge. I want to actually convert the files, themselves, prior to listening. I know it's possible to do this using e.g. Audacity, but only one-file-at-a-time; I stumbled upon this thread while searching for a program that can do batch downsampling of files. I am confused since at least a couple of the posts here suggest this is possible, yet I can't figure out how to accomplish it.

Can the above-mentioned plugin perform such an operation? If so, how can I set it up to do this for me? Alternatively, if I am mistaken and it can't, does anyone know of some other plugin (or even standalone program) that can employ SoX to do batch conversions of a large number of files? In the best case scenario (where I've read everything correctly), I guess I'm just looking for a tutorial. happy.gif

Thanks!
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Apesbrain
post Jan 11 2014, 05:01
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Converting Hi-res (HR) to 16/44 (CD)
These instructions are for Windows; all software is free

1. Do a "Full" installation of foobar2000 and its Encoder Pack

2. Open foobar2000 "File" > "Preferences" - "Components" and drop the SoX Resampler installation zip file on the open window; click "Apply" and restart when prompted

3. Drag your hi-res files to foobar2000 main window

4. Configure your file conversion:

- select all music files in foobar2000
- right-click and select "Convert" > "..."
- "Output Format": Format: FLAC, level 5. Dither (always), output 16-bit
- "Destination": Output folder: ask later. Output type: tracks into individual files. File name pattern: /%artist%/%album%/%filename%
- "Processing": Resampler (SoX): configure Resampler for 44100 "Target Sample Rate"
- "Other": When finished: ReplayGain-scan output files as albums. Optionally, select "Transfer attached pictures" if your files have embedded artwork. If you have a cover art JPG in the folder, put *.JPG in the box labelled "Copy other files..."
* IMPORTANT - Save this configuration as "HR To CD" preset so you can easily use it again
- Press "Convert" -- BE SURE TO SPECIFY AN OUTPUT FOLDER DIFFERENT FROM WHERE THE ORIGINAL FILES ARE LOCATED

5. [OPTIONAL] Good hygiene with 16/44 files is to check for Sector Boundary Errors (SBEs) which can cause problems if you ever burn these files to CD. You can use Trader's Little Helper to do this.

This post has been edited by Apesbrain: Jan 11 2014, 05:14
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ArtistofMind
post Jan 11 2014, 05:22
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Thanks a million for the thorough walk-thru, Apesbrain!

(I am new to Fb2k; I guess that's why it didn't occur to me I could simply right-click the files in my playlist.) laugh.gif

<3
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ArtistofMind
post Jan 11 2014, 09:10
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QUOTE (Apesbrain @ Jan 10 2014, 20:01) *
1. Do a "Full" installation [. . .]

FYI, it works just fine with only the necessary components installed, so I'm not sure why you specified a "full" install? (Unless it was just so noobs like me wouldn't screw something else up.) tongue.gif

QUOTE (Apesbrain @ Jan 10 2014, 20:01) *
- "Other": When finished: ReplayGain-scan output files as albums. [. . .]

I've never used ReplayGain before, and I notice you have it specified here during the post-processing stage. If I'm converting from 24-bit / some insane number of KHz to still 24-bit / normal KHz (only changing the samplerate, not the bitrate), could I safely skip this step? Also, should RG ever be applied during the Processing phase to "Prevent clipping according to peak"? (It gives a warning message when I select this option, understandably I'm sure, but can you think of a situation in which it would be desirable?)

In addition to your instructions, I have ticked the box that says "Don't reset DSP between tracks," since it seemed logical for album-level operations (even though tracks will be separate), and I've enabled aliasing, since (as I understand it) it is an acoustically transparent operation, yet theoretical improvement (like dithering), so why not? That said, when I'm merely changing the samplerate and not the bitrate, will dither even do anything?

Thanks again for your walk-thru, it told me everything I need to know; I'm just curious about these details.

This post has been edited by ArtistofMind: Jan 11 2014, 09:19
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saratoga
post Jan 11 2014, 09:14
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If you are just resampling you don't need dither, but if you are also decreasing the dit depth it is a good idea. I would probably not enable alaising if down sampling from a much higher sampling rate.
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