Resampler plugin, uses SoX 14.2.0 resampling routines Options

[jamps]

Could I set 24-bit or 32-bit somewhere, or is it pointless?

File -> Preferences -> Playback -> Output -> Output format

But it is quite pointless.

It is chosen automatically by foobar2000, I can't change it.

[Dario]

I don't see why it shouldn't let you modify the output's bit depth.

Make sure you're seeing this: http://i.imgur.com/ujavx.png?1

[jamps]

Ah-ha, I had to change the setting to use explicitly my soundcard(with Kernel Streaming):
Before:
http://i.imgbox.com/acwAb5r6.png
After:
http://i.imgbox.com/adjXTRh7.png
This option doesn't exist with DS(DirectSound). Can it be added? Because using KS results that I only hear foobar2000,
nothing else makes a sound

So, this resampler should give the best audio quality? I am resampling up to 192000Hz.
Oh and btw, I didn't understand what the aliasing does, seems like one picture is missing from the explanation...

This post has been edited by jamps: Dec 20 2012, 18:46

[xnor]

All of this is off-topic, but ...

DS will always use 32-bit on Vista/7/8,
Resampling to 192 kHz will not improve audio quality,
Aliasing.

If you have questions I recommend opening a new thread in the appropriate forum or doing a search.

This post has been edited by xnor: Dec 20 2012, 21:21

[lvqcl]

Pictures from http://src.infinitewave.ca/ , resampling of a sine sweep signal.

without aliasing:



with aliasing:



This post has been edited by lvqcl: Dec 20 2012, 21:52

[Dario]

Pictures from http://src.infinitewave.ca/ , resampling of a sine sweep signal.

without aliasing:



with aliasing:

Something doesn't seem right about those images. I thought SoX allowed aliasing to occur only above the passband?

This post has been edited by Dario: Dec 20 2012, 22:00

[lvqcl]

"Aliasing" option actually means aliasing for downsampling ang imaging for upsampling.

[bandpass]

True, but the InfiniteWave graphs are all for downsampling, so what we are seeing here is aliasing, and since the (high-level) aliasing is at >20kHz, it is above the passband.

The bottom graph also show some low-level aliasing below the passband (the purple line extending from the high-level aliasing). So a more accurate description of the aliasing option (for downsampling) might be that it allows aliasing, at levels above the selected artefact rejection level, above the passband; below the passband there may also be aliasing (and even without the aliasing option) but only at levels less than or equal to the selected artefact rejection level.

(These graphs are not from SoX BTW, but the same principles apply.)

[Avi]

I just wanted to add my thanks for your work on this resampler. I use the mod1 version to resample the few tracks I have at 22050 to conform with my DACport LX (16/24 - 44.1/48/88.2/96). It is fantastic!

[xsmid4]

Hello,
would you be so kind and add the possibility to select the target samplerate depending on source samplerate? It would be very useful for some USB dacs with 2 masterclocks 22mhz and 24mhz which select the masterclock speed depending on incoming usb stream. For example Casea Orion Lite. Because resampling just to one fixed samplerate cancels the advantage of such 2 masterclock dacs. Many thanks.

[lvqcl]

Probably you can do it using _mod and _mod2 versions.

[xnor]

Because resampling just to one fixed samplerate cancels the advantage of such 2 masterclock dacs.

May I ask you why you resample then?

[xsmid4]

Because resampling just to one fixed samplerate cancels the advantage of such 2 masterclock dacs.

May I ask you why you resample then?

every dac chip resamples internally, but then integrated resampler is not as good as Sox, so pre-resampling the 44khz (and other) material 4x to 176khz will skip 2 levels of internal resampler in dac chip. These first two levels are most important for quality. Addtional benefit is that you can choose your own filter (for example 90% passband, which results in slow roll off filter), because AD1955 chip doesn't have selectable filters and the default one is quite sharp = more artefacts.
For same reason Mark Levinson highend dacs use DSP in front of DAC chip, with custom filters/resamplers

[SoNic67]

Correct, most of the simple on-chip oversamplers in the DAC integrated circuits are just... OK. The filtering is simplified compared with a PC or DSP version and audio quality is not at maximum possible.

[Soundgeezer]

hey guys, first post here:

i'd like to try sox in foobar and dragged and dropped the dll into the components folder.
but when right clicking on an audio file in the playlist and go to convert i only get "quick convert" and default as an option.

Could somebody help me ? i'd like to convert files from 44.1 24 bit to 96 24 bit and vice versa.

thank you

[lvqcl]

i only get "quick convert" and default as an option.

And also "..."

[Soundgeezer]

Oh yes, the small print

thank you

[Night Rain]

Just wondering if it would be possible to get a mod or mod2 with passband down to 85? Thanks for considering this.

[j7n]

Thank you for publishing the old version as an attachment. Much appreciated. It also works under WinXP SP1.

[Zippy7]

@lvqcl: Thanks for suggesting how to do an AB comparison between different settings of the resampler. For other forum readers, lvqcl suggested using the SoX resampler in the configuration of the convert utility within foobar2000 to create WAV files. Then use the official ABX component of foobar2000 to ABX compare the WAV files. It even supports comparing (the same original file converted to) two different sample rates. The ABX component supports pressing the A and B keys on the keyboard to select between the choices, so testing can be performed with the eyes closed. A pause, but no clicks are heard, when switching sample rates. Slick!

I found that minimizing the buffer size in foobar2000 and my ASIO driver's controls minimized the pause duration when switching between A and B with different sample rates.

[juantendo8]

Hi, I have one quick question. My DAC supports only 44.1k, 48k, and 96k sample rate. I have some 32k sample rate music that I need to upsample in order to play. Which sample rate will give me the best quality?

[Sandrine]

From a mathematical point of view, that would be a samplerate which is an even divider or multiplier of the source samplerate. That would rank it 96,48,44. However, 96 might be overkill, so I'd go with 48. Then again, in the end it's up to your ears.

[lvqcl]

From a mathematical point of view, that would be a samplerate which is an even divider or multiplier of the source samplerate.

This means simpler calculations, but not necessarily better quality.

Hi, I have one quick question. My DAC supports only 44.1k, 48k, and 96k sample rate. I have some 32k sample rate music that I need to upsample in order to play. Which sample rate will give me the best quality?

Any.

[Dario]

Hi lvqcl,

Would it be possible to expose the resampler's "Quality" setting (Normal / Best) via its default settings (found in Advanced -> Playback -> SoX Resampler default settings)? It is the only thing that is not exposed there.

Thank you for all the hard work!

This post has been edited by Dario: Apr 21 2013, 12:46

[Robertina]

Hello, lvqcl,

SoX Resampler v0.8.3 mod and mod2 both don't create messages in foobar2000's console, although I can hear that they are resampling. In the changelog I didn't find a hint that this feature, that I appreciated very much when using v0.5.4.1 mod has been removed.

Do I overlook something?