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Dave:
How did you get Windows media player to play and output 44.1 16? What did you and in detail please. I could never get Windows media player to work with anything other than Windows native kmixer (XP not Vista) and my DAC would always receive a 48 kHz signal. I thought I tried everythingwichway but that was quite a while ago. I remember I would get an error message about missing windows components. What am I not understanding here?
Dave,
Welcome to the forum, this is very interesting stuff!
I have always been a little bit suspicious of the foobar2000 KS plug-in, as I never liked the sound of the Surgeon General warnings it came with:
Your observations prove, once again, that the only absolute truth in PC audio is that there are no absolute truths!
I upgraded my study PC to Vista over the weekend (mostly for work reasons) and since M-Audio is not ready with the Vista driver for the 2496 yet, have consequently changed back to the built-in motherboard-based SoundBlaster card to feed my Grace via its Toslink out. Appreciate you thinking about me though!
Interestingly, under Vista the SB no longer seems to insist on up-sampling to 48kS/s as it did under XP. Now it is able to send 44.1kS/s material, ALAC and FLAC files played back using foobar2000 v0.9.5 beta 5 using the following output "DS: Digital Output Device (SPDIF) (High Definition Audio Device)", at 44.1kS/s as confirmed by the LEDs on the front of the Grace.
I have set this all up according to the instructions on the Benchmark Wiki with all volume controls set to max etc. and it sounds pretty good, but I know it is not bit-perfect which is spoiling it for me. Now all I have to do is work out how to bypass DirectSound under Vista and drive the SoundBlaster via KS or ASIO. Hmmm...
Bruce,
I didn't do anything at all to cause Windows Media Player to output at 44.1kHz other than ensuring that the volume control was set to 100%. This was the first time that I had actually used it to play a wave file and I only did it because I was surprised at getting a bit-perfect file from the Foobar DS driver and I wanted to try another player in case Foobar was not actually using the DS driver. I thought it highly likely that WMP would use Directsound. Foobar was also new to me as I use a program I wrote myself to tag, manage and play my music files.
With the Audiophile 2496 the output frequency of the SPDIF Out is controlled by the M-Audio Delta Control Panel software. The hardware tab contains all the possible output frequencies and 44.1kHz is selected by default. With my Lynx card the output frequency is controlled by the Lynx Mixer software. I believe from looking at your previous posts in this thread that you are using an Audigy PCMCIA card. The Audigy software should provide an option for selecting 44.1kHz as the output frequency and I thought you said you had already used this. When I was first setting up a computer for music playing I used a USB external Sound Blaster which would default to 48kHz output and required the installation of the whole software package to get the option to switch to 44.1Khz. Sometimes the option was there and sometimes it wasn't and it would often switch itself back to 48kHz. I concluded that Creative cards were not best suited to digital music playback and moved on at the time to the Audiophile 2496 which is much better behaved.
Wave files have a header which describes the number of channels, bit depth, frequency, length of data etc. When programming with Directsound one sets up this information in a DS structure to tell the DS routines how to interpret the data that is passed to the DS buffers. I would expect this is what WMP does. It cannot change the frequency at this point as the data in the file will only make sense as music at 44.1kHz. The Directsound driver will present the output stream to the sound card driver software for output according to the mixer settings. If you have set the Audigy SPDIF output to 44.1Khz then WMP should not change it (it doesn't with the 2496 or the Lynx) but there could be some interaction between WMP and the Audigy software that causes the Audigy to believe that 48kHz has been requested or it might be that WMP starts by requesting a reset of the card and that resets it to 48kHz. If this is what is going on then I don't think you will be able to override it unless there is a "rate lock" option in the Creative software (there is in the M-Audio and Lynx mixers).
I haven't seen anything to suggest that 48kHz is preferred or a default in the Windows environment, I suspect that what you are seeing is driven by the Audigy's preference for 48kHz. I know that you have said previously in this post that you don't intend to change the Audigy but when using a card that is intended for music playback and/or recording rather than gaming and special effects in Windows XP it is realtively straight forward to get a bit-perfect datastream at 44.1kHz. I will be very surprised if this is achievable with the Audigy.
I hope that helps.
Dave
Bruce,
At last I think I might have found what you are looking for, a simple test to determine if an arbitrary computer digital output is bit-perfect on two-channel audio. You don't need a second sound card but you do need a DTS-capable receiver. I found it over at head-fi posted by thomaspf:
For any folks new to the subject who want to try this at home, I should stress the sound card needs to be connected to the receiver via its digital output (coax or Toslink).
I have played around for a couple of hours today and, as expected, was unable to provoke my Creative Audio Audigy ADVANCED MB sound card to produce bit-perfect output on its Toslink jack via DirectSound regardless of the settings I tried (I have not succeeded in configuring an ASIO driver for this output using ASIO4ALL, if anybody knows how to drive this card via ASIO under Vista please let me know). Unfortunately I can't test my M-Audio 2496 as there is no Windows Vista driver available yet
I really hope you have access to a DTS receiver and are able to try this Bruce, it just might provide the answer you have been seeking for so long. Please let us know!
Wow! But I don't have a DTS receiver!!!! I will borrow one from a friend.
Finally. Thanx man.
Struts,
You are aware the Creative Audio Soundblaster Audigy Advanced MB is a software application and not a piece of hardware?
It installs on top of the hardwares driver. You may or may not have a Creative mb chip.
Can you determine the hardware driver manufacturer? There may be a way to hack the hardware driver. What brand computer?
RG
RG,
Nope, I may have harboured suspicions in my subconscious but I can't say I was actually 'aware' of it. I have a Dell DXP061 (in Europe is was sold as a Dimension 9200, in North America the same model was marketed as an XPS). I assumed 'MB' stood for motherboard since the sound card is built into the machine, i.e. when I open it up I can see that the external sockets are mounted on the motherboard itself, not a separate PCI card.
I noticed of course that Creative installed a whole bunch of other apps (Audio Center, WaveStudio 7, Volume Panel etc.) I have no interest in but I assumed there was a device driver underneath it all. The Creative Apps required the installation of a Sigmatel audio device but I assumed this was just code Creative had licensed. However when you say these are completely separate things of course it makes sense.
So, given the above, any hints on how I can make it pass a bit-stream unmolested?
Hints?
Ha Ha!
Maybe DA guesses!
Are you using ASIO4ALL 2.8 BETA 2 ?
Have you tried ticking "Force wdm driver to 16 bit" in the asio4all control panel?
Or this, from the asio4all site: "There now is a new check box 'Force 16bit Samples', which activates a workaround for a situation where the WDM driver reports a bit depth of e.g. 20 but has no plans to actually support that when asked to. In 'normal' Windows this is no problem because in Windows, everything is just 16 bit. But ASIO4ALL tries to max out your hardware, also as far as bit depth. Most notably this workaround has been reported to get SigmaTel chips to work with ASIO4ALL".
If you still cannot get it to work, some have had success with this:
go to settings->control panel->sounds and audio devices->hardware->sigmatel c-major audio->click "properties".
Now, open "midi devices and instruments", select "wdm based devices" and click properties.
Now, select "do not use midi features on this device", apply the changes and asio4all should be working!
If all this fails you may want to try ditching the Creative software unless there is some fuctionality there you want to retain, but it's my understanding that in Vista it offers no more features than the SigmaTel driver. Then try all the above again.
It would seem if you fiddle with it enough you should be able to get it to work. Will it pass unmolested audio, hell I dunno!
Rg
RG,
Thanks for the pointers!
I am indeed using version 2.8 Beta 2. Checking 'Force WDM Driver To 16 Bit' succeeded in getting the music flowing through the ASIO driver, but it stubbornly insists on sending the output at 48kS/s (as reported by ASIO4ALL and confirmed by the Grace), even if I disable that sampling rate in the Audio control panel. This was as far as I got with this same machine under Windows XP, and I concluded that this card performs hardware upsampling to 48kS/s. Everything seems to point to this except the fact that with DirectSound I can get it to produce a (non-bit perfect) 44.1kS/s output. Aaaaaagh!
Elias Gwinn's words just keep ringing in my ears "In the world of computer audio THERE ARE NO ABSOLUTES"...
So, to bypass the evil SigmaTel I will need another card, which is of course why I originally bought the M-Audio. However M-Audio apparently won't have a Vista device driver ready until sometime next year and although I have many virtues patience is not one of them. So I guess I am going to have to get yet another sound card. Thank goodness for ebay, that's all I can say!
I bit of poking around has led me to the ESI MAYA44. It has both Coax and Optical digital outs and is cheaper than their popular Juli@ which lacks a wired digital out but seems to have better specified DACs which I don't need. Vista support? Check. Click! MAYA44 on the way...
Stay tuned for the next exciting installment of "I'll get bit-perfect out of this thing if it kills me".
Just be sure to post before you keel over.
Ditching the evil would have been my number one hint but I knew you were looking for an interim solution.
I did run across a couple posts where folks had to set 16/48 parameters in their DAW software in order to get the Sigmatel to work, if that tells you anything.
User manual on the Maya reads like you want! Will it do it?
I see their on their 2nd version Vista BETA driver. A little scary but par for the course. Certainly better than most suppliers who are still without any for some of their cards, as you well know. There
Thanks Struts,
My bad. That XPS reference got me thinking laptop. Had I taken the time to look up the model number you provided I would have known otherwise.
RG
Okay, just got back from a business trip and the MAYA44 is waiting at the Post Office. Kudos to audioart.se for great service and quick delivery.
Some first impressions:
Sooooooooooooooooooo..........the $64,000 question. IS IT BIT-PERFECT???? In short, I dunno. Sorry, haven't had time to test yet, THE BOSS is watching TV and I can't steal the receiver.
But hey, you've waited this long so another day won't hurt.
Stay tuned!
Damn! I've been thinking Foobar with KS Plugin was your only hope.
See what you've got at present.
If no luck maybe worth trying Foobar 0.9.4.5 with KS.
RG
ESI is working on it's Vista drivers. Early adopter must suffer in between. I must thank you though: if I upgrade(?) to Vista someday, all bugs will be fixed thanks to early adopters.
Ask about bit-perfection in the ESI's forums. Big boss Klaus will answer you, and it's free .
Now try some ASIO + Winamp + ASIO plug-in + upsampling 176 Ultra + 24bit + dithering, and describe me how it feels. Be sure to raise the latency to 2048 samples to avoid pops and clicks.
PS: What a shame Stereophile doesn't have anyone capable of reviewing decent computer performance. See "Computer Music" part here.
Here
A useful post regarding Vista and audio software compatibility:
http://audioforums.com/forums/showthread.php?t=16677
I have been planning to buy a new PC for a while now, but have been holding off until these issues get sorted out. ARGH!
So I always heard of people using either Foobar and others Winamp for audiophile listening. As I told before, my testing proved Winamp + ASIO + upsampling Ultra to be the better than Foobar with upsampling. Now I understood why.
The Winamp ASIO plug-in uses an open-source codec to do the upsampling called SSRC. The same codec was used in previous versions of Foobar, but they eventually decided to switch to another codec called PPHS. The reasons are listed here:
But there's no need for Foobar users to switch to Winamp, because there's the SSRC plug-in for foobar:
http://otachan.com/foo_dsp_ssrc.html <- SSRC resampler for fb2k 0.9+
So, Foobar users, please listen to SSRC in 176.4kHz Ultra mode and tell me how it feels.
Well, after keeping you all in suspense for a few days I finally found time to test the bit-transparency of my new Vista-based music server, and YES Bruce, I finally got there! At least as evidenced by the 'thomaspf DTS test' outlined above. How appropriate that the dulcet voice issuing forth from my speakers should be a beautiful Swedish bass. The monumental significance of this may be lost on those who haven't been 'living' this thread for the last year. I feel like I have found something as elusive and mythical as the holy grail or the G-spot.
Anyway, to summarize here is my recipe for this miraculous achievement:
I would just like to point out that this test demonstrates that Windows Vista Home Premium Edition, foobar2000, and the ESI MAYA44 are all capable of bit-perfect playback when appropriately configured. I have no idea which of these variables (other than Output Device) one can change without affecting bit-transparency versus which would break it, and I certainly don't have time to test all the permutations.
Does this prove that this playback chain will be bit-perfect for all 16/44.1 material? No!! Does this imply that my playback chain is bit-perfect at other word lengths and sampling frequencies? Who knows? The test file is 16/44.1-only. Strictly scientifically this doesn't prove anything about any general case. It is an observation that in at least one case the chain under test was capable of passing the bits unmolested.
But hey, that'll do me for today! Now I'm off to crack open the champagne...
PS. When I tested KS it once again bluescreened my PC and I had to type this whole post again. I would give KS in the aforementioned MAYA44 beta device driver a wide berth. You have been warned!!
PPS. Special thanks to rgibran for pointers to the last elusive pieces of the puzzle.
PPPS. Still standing, Elk
Well done, Struts!
Great info.
I posted on another thread last evening that I really needed to get a handle on this holiday drinking.
So with my morning cup of Jo I
WOW! I am more than impressed. So should I go ahead and implement this? No I can't as I have XP on both my machines and they are old, low memory machines - no chance of running Vista.
I still have not tested my system using that DTS file as I don't have a processor - plan on borrowing one.
Regarding that Holy Grail - forget it.
Regarding the G-Spot - email me and I will put you in touch with a relationship in Ohio - Yes it does exist and it is glorious.
Bruce,
I have no reason at all to believe that the same would not work under XP. Download ASIO4ALL and have a play, once you've configured the off-line settings to activate the digital output on the Audigy you should find that OD appears in the drop-down list in foobar. Just patch the L and R channels using 'Edit' in 'ASIO virtual devices' and you should be off to the races. Then you can compare directly against DS just as I have done and even KS if it is stable on your machine.
Sounds intriguing. Remind me to call you if I ever find myself in Ohio!
I can't tell the difference between ASIO or Foobar's Direct Kernel Streaming either. 'Course I did not do the DTS test, but the fact remains that I can not hear any difference. Go figure.
Only thing surprising to me about this post is you had the balls to admit it.
Told Santa what I wanted for Christmas. Soon as I get loaded up I'll see ya over there.
Thanks,
RG
Been there. Done that. Many times. <sigh>
My latest was comparing dithering methods. I can hear the difference between dither and no dither when going from 24 to 16 bit, and I can hear the difference between no noise shaping and noise shaping on some sources. But I can't tell the difference between various sophisticated schemes, even though some claim tremendous differences.
It's hard not to still obsess over it, though isn't it?
I can only assume it's the same as everybody else still reading this thread. A buyer for their megabux DAC and a ticket to Ohio!
Join the queue
Hi All, just read the whole thread from beginning to end. Excellent stuff ............ and a happy ending of sorts. Does anyone out there have the same level of knowledge on Mac based systems. I am currently re-ripping my CD collection to Apple Lossless and have invested in a headphone rig (hoping it all gets to me before Christmas) The set up will be iMac(Leopard OS) via USB or optical to MHdt Paradisea+ DAC(non over/up sampling) to Musical Fidelity X-Can V3 headphone amp to Sennheiser HD650. Is there any software out there that will improve the output from the Mac to the DAC? Any views on whether I should link via USB or optical?
When I first tested ASIO and heard all the mid-range more natural, just to be shure I asked a friend to download and tell me what he thought. It was no surprise that he also heard signifficant differences between ASIO and DirectSound.
It really surprises me you couldn't.
Matias,
I assume you and your friend are running Windows XP. If you heard a marked difference between ASIO and DS under Vista all I can say is your ears are a lot better than mine!
struts,
Excellent, you are right. I forgot that Vista has its new sound architecture rebuilt. Indeed, in XP they are both very clearly distinguishable. I also doubt I could hear differences in 2 bit-perfect signals!
I found DS not to be bit-perfect (see above), I think speed-reading let you down there
However I still couldn't hear any difference so my ears say it must be bloody close!
Just a brief update. I have now successfully eliminated ASIO4ALL from the equation. The native ASIO driver for the MAYA was there all along, it was just not visible because I hadn't configured it. User error on my part.
So I now have foobar2000 (upgraded to 0.9.5 beta 8 in the process) driving the MAYA via its native ASIO 2.0 driver (once again mapped Ch 3->L and Ch 4->R). The 'Default' settings for word length and sample rate in the Windows control panel now have no effect and I can see the sample rate indicated by both the MAYA and the Grace changing instantly to reflect the source file being played (16/44.1, 24/88.2 and 24/96 all work flawlessly). DTS test shows bit-perfect pass-through for 16/44.1.
I must mention the prompt and comprehensive support I received from Claus R over on the ESI products board (here is the thread if you're interested), exactly as predicted by Matias. Great product and great after-sales service. As far as I am concerned anybody looking for a well made PCI card supporting bit-perfect output (even under Vista) and the convenience of on-card TOSLINK and Coax outputs need look no further.
And even if I can't hear the difference, just knowing that it is bit-perfect helps me relax into the music that bit more. I feel I have achieved one of those rare (and no doubt temporary) states of audio bliss. So if you'll excuse me, I'm off to listen to some music...
Steve McQueen I guess
Oh dear. Have I become predictable?
Not at all. Not for this thread but I have just discovered a band called The Pearlfishers who have been making excellent music since the early 90's and somehow or other I completely missed them. Very Prefab Sprout/Beatles but original. Ordered their back catalogue from their record company but it probably won't get here until the New Year. I downloaded their Christmas ep from iTunes because I was impatient. It is brilliant - highly recommended. Get it and play it on loop between now and the big day
Thanks Joe, I'll certainly check them out.
Now, just to get this thread back on track, two things:
That's all for now. Happy holidays folks!
I admittedly know far less about audio then I do about computers. Fortunately, I know quite a bit about computers (for example I could write a quickie program for you that would deal with your telnet woes). So shut me up if I'm wrong here - but my advice would be:
1. remove the sound card from your PC. Smash it. Burn the pieces - there is no such thing as faithful digital output via 90% of consumer sound cards. (Although M-Audio are great)... but I digress. Get one of these http://www.tnt-audio.com/sorgenti/trendsaudio_ud10_e.html
It is a high fidelity USB DAC with several straight through digital outs if you prefer to use your own DAC as you so obviously do!
2. Go to ebay and pick yourself up a second hand old G4 mac mini. (you may also want to get cheap old small screen and peripherals to go with it)
3. Re-Rip all your music using iTunes Apple Lossless.
4. Pick up a programmable universal remote and have it learn the infra red signals from your Mac remote (all mac minis come with a remote) and your stereo remote (proprietry or not, a proper universal remote can be taught to mimic it).
...and for the love of god re-think your backup strategy. I recommend a RAID 1 NAS with two 500gig drives with your usb drive hooked into it for redundant backup in case the nas itself fails.
Thanks for the post. It's great to shake things up.
The Trends Audio device is intriguing - for that small amount of money it is worth experimenting.
I second your RAID advice. I know someone with a huge NAS based music system. One of the 750GB drives failed. He was saved by his RAID. The idea of re-ripping 750GB of FLAC files is terrifying - just figuring out which ones were lost would be bad enough.
The Trends DAC uses a PCM2704 as its core device. This chip supports USB 1.1 (not 2.0) and 16-bit word lengths with a maximum sample rate of 48 kHz. I would seriously doubht that the jitter performance of this device would be any better than mediocre, and hi-rez audio (at least 24-bit/88.2kHz) cannot be played.
Good info!
But don't you think that for $90.00 it would be a fun toy? I love little things which I can tweak and experiment with.
Perhaps it would just be frustrating.
But for $40 more, you could get the EMU 0202 that includes a DAC that supports 24-bit/192kHz and claimed jitter performance of < 100ps over USB 2.0. For fun, I'd bet you could tweak this box into a very serious contender for minimal cash.
Very cool!
I think you are correct, the yuck/buck ratio would be better with the EMU.
Too bad the jitter spec is for the clock/chips, not USB throughput.
Me too. No wonder this is such a male-dominated hobby
LOL!
The E-MU looks interesting (I may actually investigate this further myself for line-in recording), but the point of my link the Trends Audio device, was that it has optical outs, enabling you to use your own DAC. I.e. it takes the digital signal directly from USB and pumps it into whatever high end DAC you own. M-Audio have some devices that do this also, even better they have outboard gear feeding from a card sitting in a PCI express slot on the motherboard. However, I didn't recommend this as if you want to go down the Apple Mac road (as I would recommend) this will require you to have a Mac Pro (expensive overkill). Also the M-Audio gear itself can be pricy for the high end stuff. The Trends Audio device on the other hand when coupled with an old Mac Mini is a cheap way to get a true digital signal from your NAS/HD to the digital-in connector on the rig of your choice, sans windows media player "enhancements"/flaky workarounds.
Since when was USB 1.1 not fast enough to transmit full fat digital Audio signals by the way?
Hey guys, I just installed a brand new Benchmark DAC1 USB in my system! Now that was a big upgrade from my ESI Juli@. I tested Winamp's DirectSound+Benchmark's USB and Winamp's ASIO plugin+Juli@'s SPDIF, I found that the seconds sounds way better, so I'll keep the Juli@ in the signal path. I'm already talking to Elias, a Benchmark's engineer, about my findings: USB+DirectSound is definitively not as good as the clean native ASIO path, despite the tests described in their Wiki.
Add that to a new interconnect cable and my system sounds GREAT!
Excellent.
Let us know what you learn.
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