To upsample or not to upsample? Any experiences?

[acalex]

[/quote]

In Windows Vista or 7, there's a setting in "sounds" (control panel or somewhere), if you check that it says 16/44, then Windows will leave it alone and you'll get the "native" digital track going directly to the DAC as you say.

Or you can use the same setting to set your output to 24/96 and let Windows do the resampling and see if it's better or worse.

When I tried it, it was hard to tell the difference. The upsampling maybe helps, but the Windows upsampler itself is not so great, so I think the two effects cancelled each other out.

[/quote]

I see, now I understand! My Win 7 set up is on 24/96 meaning that when I use media monkey to play my tracks Windows will automaticalle upsample my 16/44 songs to 24/96. So this won't be done at DAC level (have an rDAC).

So if I want to upsample by not letting Windows do that, I just set 16/44 in the sound control panel. This way the digital track will go directly into the DAC without resampling. Than 2 question (hopefully last ) arise...

- If I play a 24/96 track on my pc and windows is set to 16/44...what happens? Windows will be "downsampling"the track to match the 16/44 set-up or nathing will happen and the signal will still go "untouched" to the DAC?

- If I play a 16/44 FLAC on my pc (with Windows set up on 16/44), the signal will go native through the rDAC which will do the upsampling itself via hardware, right?

EDIT: just verified in the control panel sound there is not the 16/44 choice. There is 24/44, 24,48 and 24/96!

Thanks a lot for taking time to reply and explaining this!

Alessandro

 

[paradiziac]

EDIT: just verified in the control panel sound there is not the 16/44 choice. There is 24/44, 24,48 and 24/96!

You can't choose 16/44 because the rDAC has "aSync" USB--the DAC "tells" the software running on the PC that it "wants" 24 bits, rather than just accepting whatever the PC sends (as in non-aSync USB).

If you choose 24/44 I think the PC just adds a bunch of zeros, so it's effectively the same as sending the data at it's native rate.

 

[paradiziac]

- If I play a 24/96 track on my pc and windows is set to 16/44...what happens? Windows will be "downsampling"the track to match the 16/44 set-up or nathing will happen and the signal will still go "untouched" to the DAC?

Not too sure, try it and see

Either the rDac will override the Windows setting to send the 24/96 file directly (I think it will do this)

OR

Windows will downsample to 24/44, and then the rDac will upsample again (obviously not a good idea).

P.S. 2 extra tips:

- Media monkey is maybe OK, but have you compared it with Foobar with output mode set to "WASAPI"?

- How does your rDac sound if you use co-ax from a reasonable CD player playing a clean disc compared to USB from the PC? If it sounds worse on the PC, you should a) look at your PC software set up b) consider using a USB isolator which stops noise from the USB port going into your rDac. This is what the designer of the M-Dac recommends. There's a company called Olimex doing them for around 30 pounds--cheaper that one of those (probably unneccessary) "audiophile" USB cables.

 

[acalex]

EDIT: just verified in the control panel sound there is not the 16/44 choice. There is 24/44, 24,48 and 24/96!

You can't choose 16/44 because the rDAC has "aSync" USB--the DAC "tells" the software running on the PC that it "wants" 24 bits, rather than just accepting whatever the PC sends (as in non-aSync USB).

If you choose 24/44 I think the PC just adds a bunch of zeros, so it's effectively the same as sending the data at it's native rate.

I think so as well...

 

[acalex]

- If I play a 24/96 track on my pc and windows is set to 16/44...what happens? Windows will be "downsampling"the track to match the 16/44 set-up or nathing will happen and the signal will still go "untouched" to the DAC?

Not too sure, try it and see --> How? The rDAC does not show the resolution being inputed...there is no display on it.

Either the rDac will override the Windows setting to send the 24/96 file directly (I think it will do this)

OR

Windows will downsample to 24/44, and then the rDac will upsample again (obviously not a good idea). --> Indeed, that why I was worried!

P.S. 2 extra tips:

- Media monkey is maybe OK, but have you compared it with Foobar with output mode set to "WASAPI"? --> Actually I have tried WASAPI and it does not work with my system. Even if I put buffer very low I keep getting the "pops" and clicks during the sond. What I have tried today is Kernel Streamer which apparently on Foobar 2000 and Windows 7 is working for now...had to but buffer size at 900ms. I might say that in this way (exclusive mode) looks like the sound improved a little bit, it seems clearer and the basses already are more "real". So using the KS and foobar 2000 I am bypassing windows mixer and playing bit perfect "unsampled" music directly through the rDAC, right?

- How does your rDac sound if you use co-ax from a reasonable CD player playing a clean disc compared to USB from the PC? If it sounds worse on the PC, you should a) look at your PC software set up b) consider using a USB isolator which stops noise from the USB port going into your rDac. This is what the designer of the M-Dac recommends. There's a company called Olimex doing them for around 30 pounds--cheaper that one of those (probably unneccessary) "audiophile" USB cables. --> Do not have a CD Player (and won't buy it) since my source is 100% digital so I will do whatever I can to improve the quality via software. I have a normal gold-plated double-shielded USB cable which I bought for 15 eur...for now I will keep using foobar 2000 with KS. I will check this company, thanks a lot!

Thanks a lot for your answer, please see my comments!

 

[acalex]

I actually found a Danish company doing it as well...including case and with no PSU needed...more or less for same price (32 eur)

 

[paradiziac]

@acalex: sounds like you are on the right track.

It's good to have a basic understanding. But after that, you can go crazy thinking too much about the "why"--in my experience, every computer, OS and playback software sounds slightly different even if you go to great efforts to configure the output to be "bit perfect". Best just to experiment and to listen.

I think the best sounding solution on Windows is to use WASAPI/Asio or KS with Foobar or JRiver Media centre with an aSync DAC. These solutions should avoid the kind of double sampling issues you are worried about. If you're playing something through your browser, just make sure you already turned off system sounds and keep the computer volume at max.

I wouldn't worry for now about software upsampling (as in the title of my thread)--I did try Sox on Foobar (which is supposedly one of the better ones) but didn't find any improvement.

Maybe you have seen this, it's a good primer on computer audio:

http://thewelltemperedcomputer.com/SW/Windows/Win7/Win7Audio.htm

Explains it better and in more detail than I could! Good luck!

 

[acalex]

Thanks a lot for all your answer, I have now a much better knowledge!

So basically an upsampling software like the one in Foobar does the resampling offline (you basically convert 16/44 into 24/96)? And then you play it through a player bypassing the Windows mixer, right?

Would be nice to have a DAC which shows the frequency in input...maybe soon :bounce:

 

[paradiziac]

Thanks a lot for all your answer, I have now a much better knowledge!

So basically an upsampling software like the one in Foobar does the resampling offline (you basically convert 16/44 into 24/96)? And then you play it through a player bypassing the Windows mixer, right?

Would be nice to have a DAC which shows the frequency in input...maybe soon :bounce:

Basically right, but I think I mixed up 2 different subjects.

1. Using WASAPI/ASIO/KS in exclusive mode in players that support it--e.g. Foobar or J River (or others--see the link) will improve sound quality by letting the (aSync) DAC control the audio and thus bypassing the normal Windows mixer. It sounds better in this mode because the audio has a more direct path with less processing by the OS and better syncing with the DAC's clock. This point is separate from the original subject of this thread about upsampling. Foobar, for example, does not resample unless you use an additional plugin that you enable in the options.

2. Software upsampling on the computer may help if you have a really good software resampler and (maybe) a not-so-good DAC (like me). Or it may not make any difference. Give it a go and post here!

 

[acalex]

Basically right, but I think I mixed up 2 different subjects.

1. Using WASAPI/ASIO/KS in exclusive mode in players that support it--e.g. Foobar or J River (or others--see the link) will improve sound quality by letting the (aSync) DAC control the audio and thus bypassing the normal Windows mixer. It sounds better in this mode because the audio has a more direct path with less processing by the OS and better syncing with the DAC's clock. This point is separate from the original subject of this thread about upsampling. Foobar, for example, does not resample unless you use an additional plugin that you enable in the options.

2. Software upsampling on the computer may help if you have a really good software resampler and (maybe) a not-so-good DAC (like me). Or it may not make any difference. Give it a go and post here!

Ok so let me recap to see if I got everything right.

1. Using Windows mixer set to 24/96 and a normal player (like Media Monkey) Windows will upsample everything to 24/96 before sending to the DAC. So the DAC will just get a 24/96 as an input and won't make any change to it (this was my set-up so far)

2. Using Windows mixer set to 24/44 and a normal player, Windows will send native source to the DAC which will make the upsample to 24/96 through its chip. Tried as well this one...couldn't spot any difference

3. Using Foobar2000 and KS will exclude Windows mixer (the system volume won't work) and this will sent a "bit perfect" output (the original bit/resolution of the track is kept) to the DAC which will upsample to 24/96. This is my current set-up and it seems better quality than 1st one.

4. Using foobar2000 wit SoX we still bypass windows mixer but this time the software will upsample everything BEFORE sending it to the DAC which won't do anything at this point. This has the advantage over the solution 1 to use a better re-sampler than windows mixer. Didn't try this one yet as I am now downloading SoX component for Foobar.

5. Using an off-line resempler software (like r8 brain) which will resample a WAV file to a wanted bit depth/resolution. But what will happen if we play this format through a normal player? A further resample coud be performed by windows mixer, correct? If we, on the other hand, use Foobar with KS we could send the resampled track straight into the DAC which won't do anything at this point (for example we resample all the 16/44 files to 24/96 and send to the rDAC using Foobar with KS). This last one (if correct and possible) should be the best solution since there is no real-time upsample with the potential advantage of using the most performing resampling software.

Are 4 and 5 correct? Or am I missing something yet? :wall:

Thanks!

 

[paradiziac]

The only other thing that's not quite correct in your summary, the DAC will still do 4 times oversampling even if you feed it upsampled 24/96. With a 24/44 file, it will do 8 times oversampling.

But it's not something to really care about. It seems a bit funny/illogical to up/oversample twice I know.

But it's an oversampling DAC, that's just how it works.

 

[paradiziac]

I'm not surprized you're :wall:, but yes I think you got it

Now you understand the basics, you can forget the theory and just try a few tweaks and see if they sound better.

Regarding 5. -- Windows should not do additional resampling if your source file (whatever it is) matches the sampling rate that you set in the mixer. In theory it would be "bit perfect". But maybe it will still sound different/worse. For example, selecting DS in Foobar sounds worse than WASAPI even though in both cases the output is "bit perfect". KS never worked for me.

I've read the rDac has problems with WASAPI which seems to be the best method, so maybe worth trying to fix. Google "WASAPI rDac" for some info on how to get WASAPI working with the rDac. Some tweaks like that might give greater improvements than software upsampling.

 

[acalex]

I'm not surprized you're :wall:, but yes I think you got it --> I am like this...when I start doing something I want to enter into small details to deeply understand how it works. It is only by doing that you can really can improve because you know how to manipulate (instead of doing many trials and errors) and know in advance what is is to be expected (more or less)

Now you understand the basics, you can forget the theory and just try a few tweaks and see if they sound better.

Regarding 5. -- Windows should not do additional resampling if your source file (whatever it is) matches the sampling rate that you set in the mixer. In theory it would be "bit perfect". But maybe it will still sound different/worse. For example, selecting DS in Foobar sounds worse than WASAPI even though in both cases the output is "bit perfect". KS never worked for me. --> Only way it works on my system.

I've read the rDac has problems with WASAPI which seems to be the best method, so maybe worth trying to fix. Google "WASAPI rDac" for some info on how to get WASAPI working with the rDac. Some tweaks like that might give greater improvements than software upsampling. --> Thanks for suggestion, I will try for sure, because it is an easy free improvement and I like "tweaking"

If something good gets out of this thread, shall we write something to sum up all the different trials and outputs we got? This should be a nice starting point for everybody wanting to improve easily a PC --> DAC through USB.

On the subject, at this point I think that the solution would make most of the sense is to get the best "bit perfect" output from the PC (making WASAPI works) and let the DAC do the upasampling. In this way everybody does what it has been made for (the software will give a native input to the DAC which will resample the native signal, don't like duplications).

If we chose to use a NOS DAC, than the solution is different of course. My next steps will be to:

- Make WASAPI work with the rDAC through USB (if somebody did already, please advice)

- Buy an USB isolator (30eur) to separate eletrically the PC from the USB DAC (which makes sense).

Stay tuned! :dance:

 

[paradiziac]

Good summary...look forward to the next results.

 

[Anonymous]

I tried this out last night from my Mac into a Bel Canto DAC2.5. The Bel Canto shows the incoming bit rate and playing lossless through iTunes shows 48hz (same as DVDs/Bluray). However if I put a cd in the DVD player it shows 44.1 as expected. What's going on in the iMac is 48hz standard?

 

[krazy_olie]

Lot of factors involved but it does make sense that upsampling can improve the sound.

Nice thing about foobar is that you can resample on the fly if your processor is up to the task. Essentially giving you the option of bypassing the Dac's internal upsampler and potentially using a better one.

Generally a good idea to bypass the windows mixer/upsampler but evn that could be beneficial for some DACs.
You can just set it to the correct output of what you are playing, it should resample it but it may mix it with the rest of your pc sound ina not particularly desirable way.

If you use ASIO in foobar it will just send the correct output automatically so you can switch between sample rates without havng to do anything. WASAPI tends to work the same but seems to be a bit iffy when switching between sample rates, I reckon that's more a foobar issue than WASAPI itself though.

On my sound card when I use asio I can see the sample rate change as i turn SOX off and on.

The main issue with the Windows resampler is that it default to 48khz, which from 44khz is not the best idea, it might actually be decent at resampling 96khz

 

[Anonymous]

I tried this out last night from my Mac into a Bel Canto DAC2.5. The Bel Canto shows the incoming bit rate and playing lossless through iTunes shows 48hz (same as DVDs/Bluray). However if I put a cd in the DVD player it shows 44.1 as expected. What's going on in the iMac is 48hz standard?

Your Mac is set to output 48Khz, if all you have is CD source material it is being resampled. You can change your Mac settings to 44.1KHz, this should improve your situation.

If you are also playing other sourced material such as 24/96 downloads, then this setting needs to change.

ITunes won't change the setting automatically, therefore you have to go into the settings to change this each time! However there are 3rd party solutions from Amarra (pricey), Pure Music (130 quid or so), and Decilbel (cheap) that will change this setting on the fly as you play tunes - go to their sites and elsewhere to check these options out for bit-perfect music.

-N-

 

[acalex]

Lot of factors involved but it does make sense that upsampling can improve the sound.

Nice thing about foobar is that you can resample on the fly if your processor is up to the task. Essentially giving you the option of bypassing the Dac's internal upsampler and potentially using a better one.

Generally a good idea to bypass the windows mixer/upsampler but evn that could be beneficial for some DACs.
You can just set it to the correct output of what you are playing, it should resample it but it may mix it with the rest of your pc sound ina not particularly desirable way.

If you use ASIO in foobar it will just send the correct output automatically so you can switch between sample rates without havng to do anything. WASAPI tends to work the same but seems to be a bit iffy when switching between sample rates, I reckon that's more a foobar issue than WASAPI itself though.

On my sound card when I use asio I can see the sample rate change as i turn SOX off and on.

The main issue with the Windows resampler is that it default to 48khz, which from 44khz is not the best idea, it might actually be decent at resampling 96khz

I am not sure of two things

1. Resampling via software is better than letting the DAC doing it. This can be tested...

2. Resampling via software will "bypass" the DAC resampling...as paradiziac said already I think that resampling is done anyway since an OS DAC is made for that as well

 

[Anonymous]

I tried this out last night from my Mac into a Bel Canto DAC2.5. The Bel Canto shows the incoming bit rate and playing lossless through iTunes shows 48hz (same as DVDs/Bluray). However if I put a cd in the DVD player it shows 44.1 as expected. What's going on in the iMac is 48hz standard?

Your Mac is set to output 48Khz, if all you have is CD source material it is being resampled. You can change your Mac settings to 44.1KHz, this should improve your situation.

If you are also playing other sourced material such as 24/96 downloads, then this setting needs to change.

ITunes won't change the setting automatically, therefore you have to go into the settings to change this each time! However there are 3rd party solutions from Amarra (pricey), Pure Music (130 quid or so), and Decilbel (cheap) that will change this setting on the fly as you play tunes - go to their sites and elsewhere to check these options out for bit-perfect music.

-N-

Much appreciated, will check this out as soon as I get home!

I guess locating these settings will be intuitive enough?

 

[acalex]

I made it work WASAPI with foobar2000 setting the buffer to a very low level (120ms). As a first impact I don't seem to hear any sensible difference using KS and WASAPI...need to listen some more before saying it is the same quality...

 

[paradiziac]

I guess locating these settings will be intuitive enough?

http://www.benchmarkmedia.com/wiki/index.php/Audio_MIDI_Setup

It's on a plate for you, mate

Though it remains to be seen if it will sound any different!

16/44 v 24/96?

 

[paradiziac]

I made it work WASAPI with foobar2000 setting the buffer to a very low level (120ms). As a first impact I don't seem to hear any sensible difference using KS and WASAPI...need to listen some more before saying it is the same quality...

Wouldn't be surprized if it was the same. I think WASAPI just tends to crash less, but it depends on your PC.

 

[krazy_olie]

Lot of factors involved but it does make sense that upsampling can improve the sound.

Nice thing about foobar is that you can resample on the fly if your processor is up to the task. Essentially giving you the option of bypassing the Dac's internal upsampler and potentially using a better one.

Generally a good idea to bypass the windows mixer/upsampler but evn that could be beneficial for some DACs.
You can just set it to the correct output of what you are playing, it should resample it but it may mix it with the rest of your pc sound ina not particularly desirable way.

If you use ASIO in foobar it will just send the correct output automatically so you can switch between sample rates without havng to do anything. WASAPI tends to work the same but seems to be a bit iffy when switching between sample rates, I reckon that's more a foobar issue than WASAPI itself though.

On my sound card when I use asio I can see the sample rate change as i turn SOX off and on.

The main issue with the Windows resampler is that it default to 48khz, which from 44khz is not the best idea, it might actually be decent at resampling 96khz

I am not sure of two things

1. Resampling via software is better than letting the DAC doing it. This can be tested...

2. Resampling via software will "bypass" the DAC resampling...as paradiziac said already I think that resampling is done anyway since an OS DAC is made for that as well

Yes and no, as there are no hard and fast rules! at a guess I would expect a software upsampler to be superior to a cheap hardware one, but really depends on the dac. Some DACs will take all your input and resample it to 96khz or 192khz for example, so if we resample in software then we avoid using the DAC's internal upsampler. In that type of "upsampling" design then using software upsampling could be beneficial.

Even if the DAC isn't upsampling it could still be beneficial, it may have support for various sample rates without upsampling but it still might perform better at higher sampling rates, so if we upsample then we tap in to the better performance. Of course it may perform worse at higher rates in which case upsampling would be a bad idea!

AS mentioned some DACs will use a completely different type of upsampling in which case upsampling in software is neither here nor there, it might help it might not.

Truthfully I suspect that the best playback of 16/44.1 material would be done with a good native 16/44.1 DAC rather than playing around with upsampling and whatnot. However having a 16/44.1 only DAC that isn't a cd player probably wouldn't sell very well.

 

[acalex]

Yes and no, as there are no hard and fast rules! at a guess I would expect a software upsampler to be superior to a cheap hardware one, but really depends on the dac. Some DACs will take all your input and resample it to 96khz or 192khz for example, so if we resample in software then we avoid using the DAC's internal upsampler. In that type of "upsampling" design then using software upsampling could be beneficial.

Even if the DAC isn't upsampling it could still be beneficial, it may have support for various sample rates without upsampling but it still might perform better at higher sampling rates, so if we upsample then we tap in to the better performance. Of course it may perform worse at higher rates in which case upsampling would be a bad idea!

AS mentioned some DACs will use a completely different type of upsampling in which case upsampling in software is neither here nor there, it might help it might not.

Truthfully I suspect that the best playback of 16/44.1 material would be done with a good native 16/44.1 DAC rather than playing around with upsampling and whatnot. However having a 16/44.1 only DAC that isn't a cd player probably wouldn't sell very well.

The problem is also the connection. rDAC best connection by far is USB which will accept "only" up to 96KHz. Meaning that the DAC will still probably upsample to 192KHz...so anyway there will be some upsampling done at DAC's level. Would be very interesting to have two DACs in the same price range, one OS and the other NOS and comparing the sound quality.

16/44 only dac is pretty useless considering that some of the people here (most I would say) buy music on Linn records or HDTracks in Studio Master quality (24/96 or 24/192).

 

[krazy_olie]

16/44 only dac is pretty useless considering that some of the people here (most I would say) buy music on Linn records or HDTracks in Studio Master quality (24/96 or 24/192).

The majority of music is still on cd though

.

I think pro-ject have a DAC which is true 16/44.1 . http://box-designs.com/main.php?prod=dacboxsfl&cat=digital&lang=en looks like it can take the higher res stuff but I imagine it will probably downsample.
Also regarding the hi-res stuff it's hard to say how much benefit is really from the high resolution of the music when it most likely comes from a completely different/probably newer and better remaster.