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Dunno is it better or worse, but USB-ASIO seems to widen the soundstage a tad and brings out more details. The bass of my DT880s have more body.
But definately a good thing to pass through windows mixer. The price is however really high, compared to the "improvements" and free ASIO4ALL...
EDIT:
Listened to Matt Mays & El Torpedo - "Cocaine Cowgirl". Result: OMG the highs! The price came a bit closer to acceptable...
Well, the advantage is definitely there. USB-ASIO actually bypasses everything and goes straight to the audio device. In Vista this will be really interesting because there are even more mixers and other APOs for a signal to go through. I added a picture to the original post to make it clearer.
Just start listening to Britney Spears and you will save yourself $75.
I am guessing x-fi supports asio, I installed asio component to foo bar but all I got was this, and got 24bit locked in.
Suzka, you don't need anything special to get bit-perfect playback on the X-fi. Go into "Audio Creation Mode", select "output" (I think that's it, it's on the lower left of the Audio Creation Mode control panel) then select "bit matched playback" at whatever the bitrate is of your source (44.1 khz for CD, FLAC, mp3 from CD sources).
Be aware all eq functions and software volume control will go away...but that's what you want...you want to control volume and eq in the analog domain, to avoid having to resample and losing information in the process. Not only are you avoiding the horrible re-sampling done in k-mixer, but you're avoiding resampling altogether (hopefully you're using the analog volume on an external amplifier).
You can use ASIO to achieve bit-perfect playback, but ASIO != bit perfect. Often, ASIO is simply used to handle resampling outside of Windows K-mixer, or to provide low latency (for applications where latency is a factor) to recording devices and the like. To have bit perfect, you have to have a soundcard or device that doesn't internally re-sample to 48 khz as a hardware function. Most Creative Labs soundcards prior to X-Fi, and the vast majority of USB soundcards are worthless when it comes to trying to achieve bit-perfect playback, because they resample to 48khz irregardless of how you feed them (If you own an Audigy 2 NX, Extigy, or Turtle Beach Roadie, I don't think there's anything you can do to achieve bit-perfect sound from 44.1 sources...you could, however, perform software resampling of your files outside of k-mixer, IIRC). If you want a cheap means to get bit-perfect playback from a PCI soundcard and stream it digitally to an external decoder/reciever via optical, then check out the Chaintech AV-710. It's $25.
I believe it's not an issue in Vista because the new version of k-mixer simply uses floating point math to determine the optimum numbers to send out in the resulting digital stream, when you're doing something like asking windows to control volume or eq. Yes, there can be numeric differences in the result, but these differences are well, well below the level of human hearing, and this process is nothing like the butchery exhibited by k-mixer. For handling audio, Vista is a fine craftsman where XP is a hamfisted dolt.
I'm lost in one regard. Why does resampling to a higher bit rate negatively affect sound quality? Rounding errors? Phase errors? Nyquist errors and jitter? How?
I hope we are not confusing sampling rate (number of samples per second) with bit depth (16-bit vs 24-bit vs 32-bit) - using a larger "word" size should not resample anything, should it, but would merely fit the top 16 bits up front into the 24-bit word, right? The extra bit depth is very useful for performing DSP calculations with greater accuracy, so I certainly hope it is considered a good thing, especially since every single professional grade plug-in performs a bit-depth conversion upwards to 24-bit or 32-bit before applying its DSP algorithms (some automatically converting back down to 16-bit as an option afterwards). The bit depth permits finer variation between the sampled levels, since it permits many more numerical values, being a higher bit-depth word.
Terry
PS - I do see the value of avoiding resampling down to 15-bits by windows, and the benefits of ASIO there and elsewhere, just missing why the Creative Labs cards of yore that always resampled up to 48 were considered troublesome.
__________________ "What you visualize is what you could create... So... Be Careful What Sorts of Images You'll Accept to be Put Into Your Mind! (By you or anyone else!)"
Headphones I currently own: Ultrasone Proline 750. Koss Portapro, Sennheiser PXC-250 noise cancelling (I like), Shure E4c (very heavenly!),
No longer used: Sony MDR-EX71 in-ears, Sony MDR-V600 Equipment: Home: Yamaha RX-V750 Receiver, Sony DVP-NS975V SACD/DVD/CD player, Polk Audio LSi9 (front), LSi7 (rear) LSiC (center) speakers, Subwoofer=2-15 inch PA-system woofers fed by Crest FA-800 @~400W/Ch. Portable: Sony PCM-1 portable DAT recorder, Sonic Studios DSM-6S/M stereo mics, iPod Nano 2nd Generation, Panasonic SL-CT710 portable CD player. My avatar is a photo of a EMS Putney VCS3 synthesizer - my first synth if you don't count my Farfisa Mini-Compact organ. I had the priviledge of learning electronic music composition from Jay Lee Jaroslav at Boston's School of the Museum of Fine Arts back in 1972-1975, and this was the synthesizer we had (well, had two of) in our lab.
I do use the same amp as you do. The Porta Corda MkIII-USB. You will indeed get better SQ by using ASIO because you are streaming through a virtual device instead of the Kmixer which is the windows default...
I am sorry if you missed some of these things. Some things have been edited after you read it most likely. Thanks for replying. I hope I made some things clear here since it is kinda hard to explain all this in plain english.
Edit: Here is the answer in plain english: YES!
Thanks and I appreciate your time on this topic, seeing this thread got me to finally try ASIO with foobar.
Thanks and I appreciate your time on this topic, seeing this thread got me to finally try ASIO with foobar.
I plan to compare the ASIO to the "stock" SQ.
Thx m8.
@ Terry
I expected someone to show up with that question. If I described some things not exactly as I should have to be exactly correct I did it to make this thread readable and to keep it somewhat more simple.
Now, for the answer. Resampling is not necessarily a bad thing. Not at all. Conventional resampling however will never increase the SQ and if done by Windows it will actually decrease the SQ. ( I am talking about bitdepth here) You are aware of this ofcourse.
If the resampling is done by a good soundcard like some X-Fis this should work really well. The main problem however is that, when using DS, the Kmixer allready resampled the bitdepth. Only afterwards the X-Fi will resample the signal which has already been set to 15-bit. (for example)
If you are using ASIO all the resampling will be done by the soundcard. (driver) And then there is no problem at all.
The new audio-architecture which is implemented in Vista will maybe present a problem for Creative. I am not sure whether they made a driver they are happy with or not.
In short: Resampling wouldn't be a problem if only the drivers used to do this in Windows would be better. I do not expect Microsoft to change this anymore since they, most likely, will concentrate on Vista now.
If the resampling is done by a good soundcard like some X-Fis this should work really well. The main problem however is that, when using DS, the Kmixer allready resampled the bitdepth. Only afterwards the X-Fi will resample the signal which has already been set to 15-bit. (for example)
If you are using ASIO all the resampling will be done by the soundcard. (driver) And then there is no problem at all.
I expected someone to show up with that question. If I described some things not exactly as I should have to be exactly correct I did it to make this thread readable and to keep it somewhat more simple.
Now, for the answer. Resampling is not necessarily a bad thing. Not at all. Conventional resampling however will never increase the SQ and if done by Windows it will actually decrease the SQ. ( I am talking about bitdepth here) You are aware of this ofcourse.
If the resampling is done by a good soundcard like some X-Fis this should work really well. The main problem however is that, when using DS, the Kmixer allready resampled the bitdepth. Only afterwards the X-Fi will resample the signal which has already been set to 15-bit. (for example)
If you are using ASIO all the resampling will be done by the soundcard. (driver) And then there is no problem at all.
The new audio-architecture which is implemented in Vista will maybe present a problem for Creative. I am not sure whether they made a driver they are happy with or not.
In short: Resampling wouldn't be a problem if only the drivers used to do this in Windows would be better. I do not expect Microsoft to change this anymore since they, most likely, will concentrate on Vista now.
I hope this answer satisfies you a bit.
Sure, EnOYiN, I understood the benefit of avoiding letting Windows get its mits on the signal. As you felt might be coming across, there was a general impression that all Creative cards that upsampled were "bad" (they did this to match the audio to their soundfont standards, as one reason, and because for a time there was a popular movement in favor of 48 over 44.1 for any card that did recording, and for direct support of SPDIF.) If Windows gives even the old AWE-32 a good signal, it can do a fine job with it - as fine as one usually would need (though its digital out for some reason had worse distortion specs than the analog out! But we're talking inaudible differences there.)
Anyway, this is a great article (and is evolving nicely!) and I hope it will get everyone to using this little piece of software!
__________________ "What you visualize is what you could create... So... Be Careful What Sorts of Images You'll Accept to be Put Into Your Mind! (By you or anyone else!)"
Headphones I currently own: Ultrasone Proline 750. Koss Portapro, Sennheiser PXC-250 noise cancelling (I like), Shure E4c (very heavenly!),
No longer used: Sony MDR-EX71 in-ears, Sony MDR-V600 Equipment: Home: Yamaha RX-V750 Receiver, Sony DVP-NS975V SACD/DVD/CD player, Polk Audio LSi9 (front), LSi7 (rear) LSiC (center) speakers, Subwoofer=2-15 inch PA-system woofers fed by Crest FA-800 @~400W/Ch. Portable: Sony PCM-1 portable DAT recorder, Sonic Studios DSM-6S/M stereo mics, iPod Nano 2nd Generation, Panasonic SL-CT710 portable CD player. My avatar is a photo of a EMS Putney VCS3 synthesizer - my first synth if you don't count my Farfisa Mini-Compact organ. I had the priviledge of learning electronic music composition from Jay Lee Jaroslav at Boston's School of the Museum of Fine Arts back in 1972-1975, and this was the synthesizer we had (well, had two of) in our lab.
SB Live! Audio [9000] 1 - 32 bit - Left
SB Live! Audio [9000] 2 - 32 bit - Right
SB Live! Audio [9000] 3 - 32 bit - <None>
SB Live! Audio [9000] 4 - 32 bit - <None>
SB Live! Audio [9000] 5 - 32 bit - <None>
SB Live! Audio [9000] 6 - 32 bit - <None>
+ and - in Foobar works volume fine, Windoze WAV/MP3 Volume does not work ... does that mean I got it working correctly? And the Offline settings are what you said to do.