44.1>48 resample. How to avoid it?

Slash_irk · Mar 27, 2005, 09:58 AM

Can anyone tell me if it is possible to playback mp3 44.1 files without 48K resample? If so, how can I do it?

Today I tested my Audigy 2 ZS with an oscilloscope and realized how 48K resampling kills the music. First, I generated 48K WAV file with 10hz-20Khz sin waveform. The signal at 48 Khz was perfect up to 18.5 Khz. Above 18 Khz the signal got worse and at 20 Khz it didn't look like a sin wave. It was rather like a noise. Next, I generated 44.1 WAV file with the same frequencies. Up to 10 Khz the signal is perfect. Above it's getting worse and already at 18 Khz it doesn't show any similarities with the sin wave.
Now I clearly realize where from the dirt in my 44.1K records....

*my current version of drivers is KX 3537
*found similar topic in russian forum, but due to the problems with russian fonts I was not able to read the message.
*sorry for my english :-)

Rogacz · Mar 27, 2005, 12:18 PM

Hm that's interresting so far. I'm using Foobar2k for music playback, and I've got SSRC Resampler in it's DSP set to 48KHz, and I wonder how does it look this way. Also, output is set on ASIO 16/48.

TravelRec. · Mar 27, 2005, 12:21 PM

I don´t think this is a resampling problem, I think the D/A conversion introduces some distortion too. Anyway - oscilloscope says nothing about the sound, the resampling noise should be inaudible in music stuff, compared to mp3 filter and compression noise. A clean sine wave is not the right instrument for testing the sound of an audio system. For example: give your amp a clean sine of say 8kHz at -5db and push the volume a bit up - the harmonics will kill the clean sound of it, dependent of your speakers. Then repeat the same with 0db input. BTW: I can proof your experiences with an oscilloscope too, stay tuned.

Slash_irk · Mar 28, 2005, 01:27 AM

SSRC plugin doesn't change the signal. I tried to configure it, but there was no change at all! So I can assume that SSRC plugin will not improve the quality in any way. If you can hear the difference in sound with SSRC plugin, it is probably of insignificant changes in waveforms or even increased distortion, but not in improved waveform.

Concerning the music... What I could see, is that 44.1K files can't be played on my Audigy 2 without increasing distortion above 10Khz. 48K files are played perfectly up to 18 Khz! So what is the problem? Also I'm sure that if an oscilloscope can see the distortion, it can also be heard by human ear...

At maximum volume the output is the best (no matter if it is 44.1K or 48K). When we lower the volume at the soundcard, the distortion is increasing. This is probably due to the D/A conversions.

And as the person who built several power amps, I can defenitely say that the first thing any person should do, is to test it with an oscilloscope and other equipment and only after that connect the speakers and evaluate the sound.

I can actually enclose the pictures of my oscilloscope, but I don't know how to do it in this forum.

The question is still open: Is it possible to avoid resample in audigy 2 and if so, how to do it?

Rogacz · Mar 28, 2005, 09:09 AM

As far as I know, Audigy 2 is internaly resampling everything to 16/48, because DSP works on this level, and I heard that software resampling to 48KHz is better than internal A2 platform one. That's why I'm using SSRC resampler.
Of course I can be wrong.

Chester01 · Mar 28, 2005, 09:42 AM

if ya wanna send a picture just host it on a different website host set up a free account with www.1asphost.com and host it there (i dont work for them and am not advertisting for them, that is just what I do so I suggested their name)

EDIT: So basically through your research we now know when ripping MP3's we should use 48khz

Also, do you compress the audio file you play through the card or is it uncompressed?

Nappylady · Mar 28, 2005, 11:04 AM

Slash,

When you play a 44.1 file, does it sound good, or does it sound muddy and yucky?

Go into Control Panel -> Sounds and Multimedia, click on the "Audio" tab, and explore the "Advanced..." buttons for both input and output devices. Make sure your hardware acceleration is turned all the way up (to the right) and the sampling rate is set to "best"--when I first made this change, Annie Lennox and David Bowie stopped lisping!

Slash_irk · Mar 28, 2005, 11:54 AM

Chester01, you are wrong. If you are ripping mp3 files from ordinary Audio CDs into 48K files, it will also be resampling, because Audio CDs are all in 44.1K.
Nappylady, I set bypass everywhere I could set it, I turned off all the effects etc. Only after that I made the measurements.

One new thing surprised my guys! When I set the PCM level and analog outpout level at the medium position such a way reducing the output signal, the 48K files now are played much better! The signal of 48K files is perfect up to 22khz!!! Unfortunately it didn't help too much 44K files. Of course the wave form became better, but it is still bad above 10 khz... You can actually check it yourself without an oscilloscope. Download any generator capable of creating both 44.1k and 48k streams. Set the level at the output of your soundcard at maximum and volume of your amp at minimum. Warning: pure 18Khz sine wave might burn your tweeters, so if you are not sure about your speakers, don't do this operation (6 watts of pure sine is enough to burn my 150-watt tweeters) . Next, increase a little the volume on your amp and play one by one 44.1K and 48K 18khz sound. You will hear that there is much less garbage at 48K. Then you can reduce the volume at your soundcard (for 50%) and you will hear, that the garbage will disappear. However, if you have speakers and amp capable of producing high resolution, you will still be able to distinguish between two streams.

Slash_irk · Mar 28, 2005, 11:58 AM

I'm still trying to keep the question open:

Is it possible to avoid the resampling on Audigy ZS?

Chester01 · Mar 28, 2005, 12:43 PM

"Then you can reduce the volume at your soundcard (for 50%) and you will hear, that the garbage will disappear. "

so cut the output by 6 db and it will make the audio clearer at 48khz and help a small amount at 44.1khz?

Nappylady · Mar 28, 2005, 02:40 PM

Slash,

When I resample 44.1khz files up to 48k for playback, they sound fine to me. I suspect something is amiss in your configuration.

Are you using the default DSP setting? And, did you check the resampling slider in the control panel? If that is set wrong, it might be causing what you're hearing.

Slash_irk · Mar 29, 2005, 12:28 AM

Chester, it is probably connected with the operational amplifier on the board of our soundcard.
Nappylady, I suppose there will no be difference in sound for ripped mp3s both 44.1 k and 48 k because they are resampled in any case. When you convert them into 48k, you get software resamppling; when you convert them into 44.1 k, you will finally get hardware resampling in your souncard.

popej · Mar 29, 2005, 07:04 AM

I'm using FXBusX plugin to get clean input to DSP at 48kHz and 16 or 24 bit.
In foobar I'm using ssrc plugin and output set to 48/24. Ssrc plugin takes about 6-9% of my CPU power, but gives better results then hardware resampling.

I made some measurements to compare playback quality, you can see this here:
http://www.anpo.republika.pl/sb0400/index.html

Slash_irk · Mar 29, 2005, 07:29 AM

I made some pictures and I can sent them by e-mail if anyone would like to post them :-)
It is probable senseless to require 44.1 k support from Audigy for its price.

Nappylady · Mar 29, 2005, 10:00 AM

slash,

my email is david@naptastic.com and anything you send to me, I will host & post as soon as I get it--as long as it's not too big :-)

On that link popej posted: make sure you look at the IMD numbers for 44.1 playback--that sounds a lot like the phenomenon you're describing.

Slash_irk · Mar 30, 2005, 08:45 AM

Nappylady, I hope you received the pictures I sent you.

Slash_irk · Mar 30, 2005, 10:11 AM

Guys, I made some more tests. I tested the output with SSRC plugin and without it at different frequencies.

I should say I was wrong about SSRC plugin. It really helps. But something is still wrong with the signal. Up to 12 Khz the signal is relatively good both with SSRC and without it (44k). From 12khz to 18 the signal is bad both with SSRC and without it. From 18 Khz to 22khz the signal is good with SSRC and really bad without it. Strange phenomena. I tried different configurations of SSRC plugin, but it doesn't help the frequency range from 12 to 18 khz.

Here is the link of tests - http://fota.mota.ru/view.php?id=30547

The web site is russian, but all comments are in english.

Slash_irk · Mar 30, 2005, 10:13 AM

Nappylady - I sent you the photos, but I decided to post them myself. Thanx for your help anyway!

Nappylady · Mar 30, 2005, 10:45 AM

Hmm.... well... no offense, but I'd rather not click that link, with those banners on the bottom of the page and my dad just in earshot http://www.driverheaven.net/images/s...big%20grin.gif

http://www.naptastic.com/44_10000.jpg
http://www.naptastic.com/44_16000.jpg
http://www.naptastic.com/44_20000.jpg
http://www.naptastic.com/48_22500.jpg
http://www.naptastic.com/48_23400.jpg

So there they are, one by one, as you sent them to me... (and no naked ladies peering over my shoulder)

(I guess I should check my email more often, eh?)

Slash_irk · Mar 30, 2005, 12:18 PM

Nappylady, Should I send you the rest photos I made today (include tests with SSRC plugin on different frequencies)?

Chester01 · Mar 30, 2005, 07:23 PM

have you tried the pphs resampler? is it inferior to the ssrc one? also: noise distribution gaussian - how can i enable this in foobar2k? Thank you for all your hard work on this topic! We need more perfectionists like you

(not putting anyone down there)

Chester01 · Mar 30, 2005, 07:32 PM

slash, can you please post a picture of your DSP? I am curious which plugins you are using. I currently use the p16v plugin which I believe accepts a 24 bit input. To avoid re-resampling; I should use 24 bit output on foobar correct?

TravelRec. · Mar 31, 2005, 01:10 AM

Hi there!

Made also some tests to check whether the resampling is audible or not. My oscilloscope shows the distortion also, but I don´t think that my ears can feel any difference, so for you all to compare:

Analog recorded wave with SB0400 UDA input and 44.1k, rebuild as *.mp3 44.1kHz 160kbps:

www.electricstart.de/sound/psyche441.mp3

Analog recorded wave with SB0400 UDA input and 48k, rebuild as *.mp3 48kHz 160kbps:

www.electricstart.de/sound/psyche48.mp3

Can you feel any difference?!?

igor_levicki · Mar 31, 2005, 02:40 PM

To answer the question, the only way to avoid resampling in hardware is to play everything at 48 khz. Has anyone tried r8brain src converter from voxengo? You can rip cds and then batch convert them using HQ conversion to 48 khz and then pack to .ape or .flac format. It will surely sound a lot better.

Slash_irk · Apr 2, 2005, 02:04 AM

Chester, can you tell me how to make the picture of dsp (should I just post the photo of "4X DSP" or something else)? And I didn't try any other SSRC's, only SSRC 2.2.3. of winamp. The different configuration didn't change the signal too much. No matter what it is, 24 but, guassian or whatever. Just set 48k resampling and 16 bit. It will be enough.

Trevelrec, I tried to hear the difference, but I couldn't both using SSRC and without it.
Can anyone post a short fragment from DVD audio good-sounding CD? As far as I understand the records on DVD Audio are all in 48K. We could resample it into 44K and compare two records.

Chester01 · Apr 2, 2005, 08:44 AM

the easiest way to post a picture of the DSP is to take a screen shot of your computer screen. All you have to do is press the 'Print Screen' key at the top of the keyboard (by F12) and a screen shot of your current screen will be copied into the clipboard. Open up MSPaint or whatever program you use to create/modify pictures with and select Edit then Paste. You may want to crop so we only see the dsp (so we dont see personal things on your computer.) Good luck!

igor_levicki · Apr 2, 2005, 01:22 PM

Check this out. They claim it is the best SRC software around.

refraction · Apr 2, 2005, 01:24 PM

Quote:

Originally Posted by Slash_irk

One new thing surprised my guys! When I set the PCM level and analog outpout level at the medium position such a way reducing the output signal, the 48K files now are played much better! The signal of 48K files is perfect up to 22khz!!! Unfortunately it didn't help too much 44K files. Of course the wave form became better, but it is still bad above 10 khz...

thats very strange, if both results in 48khz and 44khz were the same the that would be right, obviously as 48khz is the resamples one, the results should be the same as 44khz as you cant gain data from taking a lower sample quality and raising it. 22khz would be right as long as you are in stereo as you have to remember thats 44khz for BOTH channels, so 22khz for each, obvioulsy this only applies to stereo.

i think i would agree its an A-to-D problem, signal loss is common especially where there is a lot of electrics, only thing you can try is moving your sound card as far away from any other components as you can.

nyquility · Apr 3, 2005, 04:56 PM

BTW, just to let everyone who used a resampler in foobar2k know, there is a 3rd resampling option I have fallen in love with personally

http://pelit.koillismaa.fi/plugins/dsp.php#114

It is based on the Secret Rabbit Code resampler from here

http://www.mega-nerd.com/SRC/

Quality seems to be good on my Audigy 2 ZS Platinum.