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24-96 to 24-48/16-48/16-44.1kHz converter sofware

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PhilippeC

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Just curious: will this resampling beat (in quality):

a. Hardware resampling such as done by all MD decks and portables (pls see note below).

b. Hardware resampling done by my receiver which I believe upsamples to 192 (or at least 96). Admittedly I cannot persuade it to output digitally but that's more a matter of SCMS than anything else.

c. Software resampling done by Cool/Audition/SoundForge

d. Hardware resampling done by Behringer 2496

So far I have been the most impressed by d. But it occurs to me that perhaps Sony does all this in both h/w and software. Maybe not on all decks, but I do recall there being circuitry for x4 at the very least, so that would be176.4, in some of the later decks such as the MXD-D400.

Stephen

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Just curious: will this resampling beat (in quality):

a. Hardware resampling such as done by all MD decks and portables (pls see note below).

b. Hardware resampling done by my receiver which I believe upsamples to 192 (or at least 96). Admittedly I cannot persuade it to output digitally but that's more a matter of SCMS than anything else.

c. Software resampling done by Cool/Audition/SoundForge

d. Hardware resampling done by Behringer 2496

Stephen

a. yes

b. yes

c. yes

d. yes

The r8brain resampler is one of the best available resamplers anywhere. Equally good: SoX (foobar plugin). Both are free, yet their quality leaves nothing to desire.

The best resampler is the awfully expensive Weiss Saracon. Pristine results... at great costs.

r8brain & SoX are almost as good but free. What more can one want?

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Wow.

May I ask how u can be so certain? Are there new techniques bring used which date more recently than the above hardware and software that I mention?

The samplerate converters in all Sony MD recorders were released more than 10 years ago. Their major goal was to be portable, meaning they weren´t allowed to consume much energy/power. Conserving energy always goes hand in hand with limited processing capabilities. In turn that means that the CPU for resampling is small, slow and doesn´t take too much energy while working. This limits the complexity of the embedded algorithms used for resampling (and resampling from 48 kHz to 44.1 is an extremely taxing process involving complex algorithms).

Therefore, the resampler used in those machines is OK, but nothing more. It works and gets the job done with - from today's standpoint - mediocre to bad quality.

Conclusion:

1. they´re old

2. not sophisticated

Compare that to r8brain or SoX. Would you use them with computer hardware from 10 years ago they´d take forever due to their modern and complex architecture. But on a modern PC they´re fast AND they resample with a quality that would have been considered impossible 10-15 years ago. I should know, I started to work with Sound Forge in 1995. The resampler built into this DAW around that time was slow, extremely slow. And the quality wasn´t even that good. It has been improved somewhat over the years, yet it still is an OK-resampler only.

This is a site where resamplers are compared:

http://src.infinitewave.ca/

On those tests they resampled from 96 to 44.1 kHz (equally taxing as going from 48 to 44.1) and the results were measured. The resamplers I recommended (SoX and r8brain) are one of the best. I confirmed their results for myself with a bunch of my own measurements.

Please note that foobar2000 is included in the list. They measured the resampler foobar comes with in this case. SoX is listed seperately.

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SoX

I have it in my foobar2000 components and in my active components but NOT in my active audio file right clik menu (like Replaygain).

I see also in the Ctrl+P settings DSP manager window that you can use it for upsampling and downsampling but only for "frequencies" (kHz), no 24 to 16bit or 16 to 24bit setting. It look like a mess to set up for beginners. Does I miss something ?

What are the steps to convert a list of 24-96 files to 16-48 ? What are the settings for a high quality conversion ?

d6eu.png

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If I understand well, SoX is also for the upsampling

And as such, be used as a plugin in foobar2000 to upsample all audio files during the listening BEFORE the sound is sent to the 24-96 (or more) DAC. Some guys in forums tell that this the only way to control HOW / BY WHAT the upsampling is done.

A french guy's setting (24-96 DAC)

upsampling%20sox.jpg

MDietrich, what is your experience with foobar2000 + SoX + Creative USB HD soundcard ?

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SoX

I have it in my foobar2000 components and in my active components but NOT in my active audio file right clik menu (like Replaygain).

No, it only appears through the right-click 'Convert' dialogue. Just like every DSP within foobar (except for playback).

I see also in the Ctrl+P settings DSP manager window that you can use it for upsampling and downsampling but only for "frequencies" (kHz), no 24 to 16bit or 16 to 24bit setting. It look like a mess to set up for beginners. Does I miss something ?

What are the steps to convert a list of 24-96 files to 16-48 ? What are the settings for a high quality conversion ?

nk09.jpg

The passband setting configures the steepness of the aliasing filter. In my opinion it should always be at 90% - that way you lose a bit of frequency content but you´ll gain perfect impulses that way.

For downsampling: NEVER tick 'allow Aliasing/imaging'. Otherwise you´ll introduce artifacts into your audio.

For upsampling: now you can tick 'allow aliasing/imaging', for upsampling it doesn´t matter, it´ll also relax the aliasing filter further.

u06i.jpg

You configure bit-depth within foobar itself. SoX will always resample internally with 32 bit floating point precision (every DSP embedded within foobar will). I´ve encircled the important parts.

Please note: the settings I pictured above are only for 16/44.1, 16/48, 16/96, etc.

If you want to store as mp3, aac, wavpack or ogg the output bit-depth should be set to 32. Those codecs can handle floating point input very well. If you want to save to FLAC, I recommend 24... or 16, depending on what you encode your audio for.

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If I understand well, SoX is also for the upsampling

And as such, be used as a plugin in foobar2000 to upsample all audio files during the listening BEFORE the sound is sent to the 24-96 (or more) DAC. Some guys in forums tell that this the only way to control HOW / BY WHAT the upsampling is done.

Yes, SoX can be used very well for upsampling.

Should you want to use it for playback, place it in the 'DSP' menu of foobar (this won´t apply to the converter; that one uses a different engine for conversion). And those guys in forums are right: if you want to have full control over resampling you´d have to do it yourself (depending on the hardware and the driver you use).

MDietrich, what is your experience with foobar2000 + SoX + Creative USB HD soundcard ?

I don´t have any experience with foobar2000 + SoX + Creative USB HD. For one, I don´t use this card for playback (except for when recording to MD - and in that case I´ve already resampled the material to 44.1 with another resampler).

I upsample my material before playback, I also store it this way.

BUT: if I would do my upsampling on-the-fly, I´d do it with SoX. The Creative USB HD can only handle samplerates based on 48 kHz for analogue playback (meaning: 48 & 96 kHz are fully supported - if all DSPs of the card are inactive). The card itself doesn´t have a resampler, it uses the one built into Windows instead. And that one is crap compared to SoX. If you use the Creative USB HD through its analogue output I´d always resample any material to 96 kHz.

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