This to me is the holy grail. I've done endless searches on this and www.thegreenbutton.com, but can't seem to find a definitive anwser. So let me pose the question, and feel free to flame me if I've missed the answer somewhere before!

I run MCE2005 as my OS, as it looks kinda slick, and my wife can drive the remote control. I have an AV-710 sound card hooked up via optical to my Yamaha DSP-A2 amp. I have an ATI 9000 series card to drive both my Pioneer plasma (via a DVI to HDMI cable) and my woidescreen monitor simultaneously. (So you don't need the plasma running while you're only playing music.)

I have all my music CD's ripped to WMA lossless on the machine, and I intend to use it for playback of audio, mpeg video, and DVD's.

Now the tricky bit. I would like the machine to provide the appropriate bitstream to the amplifier - ie. play back a CD, and get 44KHz PCM. Playback a DVD and get 48KHZ DD/DTS. So - no messing with the sound. If I want to mess with the sound I'll do it with the amp.

You see where I'm going. A single audio connector to the PC, and it passes through the native bitstream (OK it will decompress the WMA lossless) to the AMP.

However, I don't think its possible. Apparently, the kmixer component of Windows will always resample audio to 48KHz to allow it to multiplex system and other sounds to the output device.

You can bypass kmixer by using cards that support ASIO or kernel streaming, but they then rely on a media player that supports ASIO or kernel streaming. MCE2005 uses WMP10, and YOU CANT USE ANYTHING ELSE.

So either I use the intuitive media center interface and get crummy sound (believe me it sounds rubbish) OR I use seperate applications for DVD, mpeg playback and audio playback.

Is there any way of getting what I want? I've spent ages building the machine, ripping the CDs, setting up the amp. But it looks like my dream of a fully integrated media centre with bit perfect playback has fallen at the last hurdle.

I too would love to make this happen......alas......

You should probably change your subject line and add a question mark at the end.

I'm not sure if this is a good suggestion but, if its the remote functionality and Wife Acceptance factor that leads you to use MCE, Why not try Mediaportal. You can use this over MCE OS without having to use the MCE app.

I find media portal to have a very friendly user interface similar to MCE. They even have an MCE skin if you want the familiarity.

The great thing about MP is that you can do almost everything you can do with MCE but having the flexibility of using any app with it to play your media and not just windows mediaplayer.

I believe Foobar can output bitperfect audio. MP has a Foobar Plugin. Also I use both the internal Mediaplayer of MP using FFDshow and Nvidia Codecs and WinDVD6 as the DVD player for DNM. I believe MP is a very flexible App best of all its free. Try experimenting with it. I do everything I need my HTPC to do under MP without even leaving the interface.

Vectorzero, I am fairly sure you *can* do what you want.

I asked and asked and searched everywhere and could find no solution. But then in a distant thread in this forum, one guy made a one line comment that no-one picked up on, and in that single one-liner I think there was the answer.

Basically, he said "use reclock".

At the time I didn't figure out what he meant and it went unnoticed. But I think that is the answer.

Re-clock (as perhaps you know) is a tool that helps sync a soundtrack to a movie (and not the other way around) in order to help stop stuttering of the video content. But there's much more to it than that, and - critically - it doesn't have to be used just for movies.

***It has one incredible feature***. You can get it to override the normal output from WMP10 and to be loaded as the sound device, instead of directsound. Insodoing, you are halfway there. By bypassing directsound, you are setup ready to be able to bypass kmixer. All you need to do is to configure reclock to output via kernel streaming.

Now the good news and the bad news. Reclock supports Kernel Streaming. Hurrah!!!!! So you can configure re-clock to load instead of directsound and then for it to use kernel streaming to output bit-perfect content to your AV processor. And since MCE2005 basically uses WMP10, this will work EVEN IN MCE 2005!

The bad news is that Kernel Streaming is disabled by default and to get it enabled, you have to send the author (of reclock) some money. I emailed him about this and he never replied. :-( In fact he's disappeared and even in the reclock forum, no-one has seen him for 2 months. So if he doesn't reply, I have no kernel streaming mode :-(

So I have yet to test it, but from what I can tell, it will surely work.

Chip

Yeah if you can get this to work succesfully i would be VERY happy. i would even be willing to send the author some money myself

I have read posts which state that bit perfect output in MCE is possible, by using a M-Audio Transit USB sound adapter. If you find the topics you can enter the info here.

I'll get on the reclock bandwagon and send the guy some bucks...let you know how it goes.

http://www.avsforum.com/avs-vb/showthread.php?threadid=527242

this looks like this might shed some light on the subject

http://www.avsforum.com/avs-vb/showthread.php?threadid=527242

this looks like this might shed some light on the subject

Funny. There was a throw away line about reclock in that thread too that nobody picked up on. I am sure reclock is the answer, yet no-one seems to grasp it.

Chip

I assume you guys are talking about me since I throw it out there whenever somebody asks. Yeah, Ogo isn't very responsive. That might turn around if you pay him so you could chance that route if you're really interested in bitperfect.

tm

Funny. There was a throw away line about reclock in that thread too that nobody picked up on. I am sure reclock is the answer, yet no-one seems to grasp it.

Chip

I have read posts which state that bit perfect output in MCE is possible, by using a M-Audio Transit USB sound adapter. If you find the topics you can enter the info here.

I am also looking to do the same, and have been down the M-Audio Transit path. While it is possible to get bit perfect playback of wma loseless (my amp even shows 44.1 khz) it's not possible to get DD/DTS passthrough without changing a setting in the driver. Trouble is, when you change that setting it works fine for DD/DTS, but does not play back any other non DD/DTS audio :mad:

jpadua - Mediaportal looks like it has potential but needs to mature. The TV guide was a mess - I couldn't get it working....and the program didn't even go full-screen with that option enabled. I browsed the forums a bit and it looks like a definite work in progress....nothing wrong with that - I think open source is great - but I gave up on meedio for that exact reason - too much tweaking and not enough enjoying.

I would send some money as well
if I knew he would respond
too bad he does not know there is a group of people trying to shove money at him!!

does anyone have this version and have tested it?

I wish I could try it too. But I fully expect it will work.

With Foobar and Kernel Streaming, everything works fine: 44.1kHz CD's play at 44.1 and DTS CD's play just fine. HDCD encoded CD's work too, with the HDCD logo lighting up on my processor (unlike with WMP, which manages to scramble the HDCD encoding somehow).

This demonstrates that Kernel Streaming will do what we want. All we need is some way of getting Kernel Streaming to work from within WMP10 and then we should be away. And reclock will do just that.

So all the pieces are in place...

Chip

This is what I do, set the transit usb as your sound device for WMP. and as previous posters have said this will give you good output for pcm.

however keep your default sound card as the onboard sound card, and MCE will use this for everything else (DTS/DD, TV etc) that way you don't have to switch the settings for the transit all the time. music from MCE will come out transit everything else from onboard
You will have to change inputs on your amp, but usuually thats easy by remote.

RD

viggster,

yeah I had trouble at first also with media portal. It's kinda daunting at first. But now I'm so used to setting it up I can set it up in minutes. Ive got the same problem with meedio, to much configuring, too many plugins dont know which ones to use etc. Anyways since Ive switched to media portal, everything was a breeze (after the steep learning curve of configuring it)....

Sorry.. I know this is a bit perfect audio thread :) just wanted to help out, but I really know nothing about it.

How do I know if my audio output is really 44.1khz not 48? I tried using Foobar, the status bar reports 44100 but I dont really know if my Soundstorm is sending out 44.1 or 48... how can I tell? Does the output have to be PCM? using soundstorm my output is 5.1

stay alive.... need an answer

Ok, I think I've found our answer. I was going to mess around with reclock, etc, but another then discovered a usb device that should do the trick. It's the M-AUDIO TRANSIT. It's tiny, so it will be unseen behind my htpc. According to another thread on avs, it does bitperfect for everything. Search and you'll find the thread...I'm too lazy at the moment.

I'll let you know if it works. Just ordered one for $67.

Yup, the m-audio Transit is the way to go. Just got mine. Installed 1.022 drivers from website, plugged in, instant bit-perfect from EVERYTHING, MCE2005 included. I don't know the technical details, it just works. After messing with ASIO etc I can say it's well worth the $70.

One downside: It does not autosense the source. That means a manual switch from 16-bit 44.1khz to DTS in the device's control panel. But there is another solution: now that I have 2 soundcards, I'm going to set up the old one for DTS and switch inputs at the stereo with the remote. No movin' the lazy ass.

Try it! It Works!

If you don't mind dropping some cash, I beleive an RME card will do both bitperfect pcm and DD/DTS without haveing to switch any settings. I haven't personally tried one as they are too rich for my poor a$$. The Transit does work but the lack of auto sensing sucks.

This post is now linked from the MCE 2005 FAQ (http://www.avsforum.com/avs-vb/showthread.php?t=465419) for your convenience, and I have summarized findings of all in this thread below. Special thanks to chung_chang for the MSI motherboard write-up.

Below is my attempt to condense hugely useful and IMO hard-to-find information on giving MCE 2005 / WMP10 better audiophile-grade sound output.

For all the lurkers out there wondering the same thing I am, here's the answer I think:

To get bit-perfect 44.1 kHz audio passed to your pre/pro or receiver while playing audio from MCE 2005 (which uses Windows Media Player 10) without EVER having to switch programs from MCE to something that uses ASIO, there are five things you can do, explained below:

1) Use the M-Audio Transit USB configured in audio mode and set this as the default sound option in Windows Media Player 10. If you do this, your DTS/AC3 (Dolby Digital) recordings will not be passed properly unless you manually switch output in the M-Audio control panel before you watch an AC3 track, but as many have suggested, all you have to do is leave your system's on-board audio SPDIF output set as the player for DVD and system sound, and you're all set. So, you could use the Transit over USB for your music (e.g. receiver's optical input 1) and use your system's soundcard or onboard sound (e.g. receiver's coaxial SPDIF input) for DVD/DTS/DD/Gaming and normal Windows sounds. You could manually change the Transit USB to DTS/DD mode but that defeats the point of mouse-free (remote control) bit-perfect audio listening for me. ***Edit 4/30/06: Transit USB installed and working perfectly. Also note: reports of a PPAI brand (model #1455) USB sound card achieving bit-perfect output have been verified here (http://www.avsforum.com/avs-vb/showthread.php?p=7575646&&#post7575646) , so this off-brand model might work for you as well if you can find it.

2) Use a Creative Audigy 4 card and select the "Bit-accurate" output option in creative software as detailed here (http://www.avsforum.com/avs-vb/showthread.php?p=12511114#post12511114).

3) Configure an MSI K8NGM2-FID motherboard with onboard HD Audio according to the following writeup by Chung_Chang (http://www.avsforum.com/avs-vb/member.php?u=7495040) :

Q: Is there any onboard sound that is bit-perfect under MCE2005?

A: Yes. The MSI K8NGM2-FID (nVidia 6150) can achieve bit-perfect sound under MCE 2005 using the onboard Realtek 880 chip. It can handle passing DD and DTS in both 44.1KHz and 48KHz and it has been reasonably proven that 44.1KHz 2 channel PCM (typical stereo music CD) is also bit-perfect. The best part of using this board is that switching between 44.1KHz and 48KHz source sampling rates is done automatically by the driver. It is also believed that other boards using the Realtek HD-Audio Codecs can produce bit-perfect sounds but that has yet to be proven.

Q: How do I set up for bit-perfect audio with MCE 2005 using the MSI 6150 Motherboard with HD-Audio?

A: First, follow this excellent guide (http://forums.anandtech.com/messageview.aspx?catid=29&threadid=1803985&enterthread=y) to complete your basic setup. Note: as of this writing, the only Realtek drivers confirmed to work as desired are v5253. These v5253 drivers are being graciously hosted here (http://tbaudio.com/realtek/WDM_R137_v5253_bitperfect.zip) by bling24 if you can't find elsewhere. Once drivers are installed, find a DTS sound file (like this one (http://www.sr.se/laddahem/MultiKanal/Dts/SURROUNDTEST_011212.zip)) and play it with WMP. If you can't hear the sound correctly, go into the sound driver configuration, click Audio I/O tab, click the configuration button next to Digital, select 44.1KHz and click the Auto Lock (http://www.avsforum.com/avs-vb/attachment.php?attachmentid=57724) button (all while the DTS sound is still playing). You should now hear the DTS sound perfectly and all other sounds will now be bit-perfect too under MCE. There is also no need to change those sound settings again even after reboots.

4) Use ReClock, the tool developed by an AVS forum member (http://reclock.free.fr), to alter the actual frequency the Microsoft DirectShow (standard for Windows/WMP10/MCE) drivers use to reproduce audio and video. I have not tried this, and probably will not, as I'm going to use solution #1 above. I am inherently skeptical of tinkering with DirectShow / DirectX for fear of jeopardizing other computer uses as well. If anyone finds more success with this, let me know.

5) Purchase an expensive RME Audio Hammerfall card that supposedly bypasses Windows kmixer.sys using its own driver, and hence does not corrupt the digital audio output in the process. Thomaspf shares that "Their windows XP drivers are not native WDM audio drivers but Win9x/NT style MME drivers with some additional glue (power mgmt, PNP) to make it run under W2K and XP. One of the open issues for them is to support native WDM applications." For more speculation on this solution, check here (http://www.hydrogenaudio.org/forums/index.php?showtopic=5755&st=25#) . This is not an option for me due to both price of the card and inherent riskiness of using a driver that must "stay broken" in order to keep working.

So that's basically it...finally I think I've found my solution, and it's the Transit USB. I will be ordering an MSI motherboard to test onboard sound, for now direct questions to chung_chang.

Remote-controlled bit-perfect audio using MCE 2005 and it's streamlined "10-foot" interface will be a great thing, because now I don't have to teach my wife to use JRiver, Foobar or even Winamp with the ASIO plugin for music, and we can use all the categorization and integration features of MCE 2005. Easy and WAF-approved...Yippee!

I hope others seeking to do this find my post useful.

Best,
Adam

An intersting paragraph in the M-Audio Audiophile USB's manual.

"If you have this box (DD/DTS Pass-Thru) checked, you can still send stereo digital audio to the S/PDIF port without any problems."

I'm going to get ahold of one of these to see if it works. They are like $99 on eBay right now.

An intersting paragraph in the M-Audio Audiophile USB's manual.

"If you have this box (DD/DTS Pass-Thru) checked, you can still send stereo digital audio to the S/PDIF port without any problems."

I'm going to get ahold of one of these to see if it works. They are like $99 on eBay right now.

Interesting find, I'll have to try that myself. Remember to use a DTS CD or .wav test file to verify you're passing bit-perfect info when you get it installed. Bottom track (14MB) on this page makes a good start (If you hear DTS and not ugly noise, you're bit-perfect:

http://www.sr.se/multikanal/english/e_index.stm

BTW, $79 shipped from zipzoomfly last time I checked.

-Adam

BTW, $79 shipped from zipzoomfly last time I checked.

-Adam

You got a linky? All I can find is $129. This is the USB version, not the PCI.

Looks like ReClock has kernel streaming enabled for everyone now. The beta is being updated and is the current work on getting it to run in MCE.

You got a linky? All I can find is $129. This is the USB version, not the PCI.

Just double-checked and apparently Bob's guidelines indicate that we are allowed to post parts deals in the HTPC forum, so here ya go:

http://www.zipzoomfly.com/jsp/ProductDetail.jsp?ProductCode=123304

Where did you see $129? The MSRP on M-Audio's site is only $99 I think...

Anyway, good luck!

-Adam

Try setting reclock as your default system renderer and set it up to use kernel streaming for audio output.

This should get you bit-perfect playback with most sound cards that do not resample to 48Khz in hardware.

Cheers

Thomas

Just saw this today....might be worth looking at.... http://www.auzentech.com/products_xplosion.html

Try setting reclock as your default system renderer and set it up to use kernel streaming for audio output.

This should get you bit-perfect playback with most sound cards that do not resample to 48Khz in hardware.

Cheers

Thomas

Alright, you've tempted me and I think I'm going to go futz with DirectShow by trying out ReClock with my Asus Nforce2-based Mobo...

And we now have conversations in three threads Thomas... :D

Cheers to you,
Adam

Does the Nforce 2 support 44.1KHz output?

Does the Nforce 2 support 44.1KHz output?

Actually, the Asus A7N8X motherboard's onboard audio uses an Nvidia Soundstorm chip that is very capable of actually encoding AC3, but for passthrough playback I think it uses an Nforce 2 "MCP" chip and a plain old "Realtek ALC650 6CH w/built in HP amplifier" that locks things at 48kHz. Worth a shot to try, anyway. I'll use the test track to check it out tomorrow.

-Adam

I know about the realtime encoding but I thought it could only do 48Khz?

Cheers

Thomas

I know about the realtime encoding but I thought it could only do 48Khz?

Cheers

Thomas

Actually, you're right. No dice on this one with my existing onboard sound, it's locked at 48khz.

Well, I have a Chaintech AV710 on order and an M-Audio Transit USB, both of which are capable of bit-perfect 44.1kHz output, though I will not need ReClock to use the M-Audio Transit with MCE 2005. I'll probably play with the Chaintech using ReClock just for kicks to see if it works and report my results here.

-Adam

Try setting reclock as your default system renderer and set it up to use kernel streaming for audio output.

This should get you bit-perfect playback with most sound cards that do not resample to 48Khz in hardware.

Cheers

Thomas

OK, reporting in here with a question:

First, the question: how do I set ReClock as the default system renderer in MCE or WMP settings? For some reason, I can't find it. If you mean setting the "force" option in the ReClock configuration panel, I've done that.

I've got my Chaintech AV710 in and have been tinkering away. 2 successes and 1 failure so far.

First, my setup:

Asus A7N8X Deluxe (Nforce2) with Athlon XP 2400+
Windows XP MCE with Rollup 2
Foobar2000 version 0.8.3 with Kernel Streaming output .dll enabled
Chaintech AV710 Soundcard, using ViaArena.com latest Envy24 Drivers (Envy24_Family_V500a)
ReClock Beta (1.7b4)

Next, my settings:

Via Envy Control Panel: 2-channel mode set to 44.1kHz, SPDIF set to PCM only and 44.1kHz.
Foobar2000 Preferences:Output set to kernel streaming, device set to Envy24 (Chaintech card)
ReClock Config Panel: Outputs set to kernel streaming, devices set to Envy24, Force ReClock to be renderer for DirectSound enabled

Now, my successes:

Foobar2000 in kernel streaming mode plays the 44.1kHz DTS test track and all my lossless and VBR MP3 audio files flawlessly. It sounds fantastic, and vastly improved from Nforce2 onboard sound being muddled by Kmixer to 48kHz.

Windows Media Player plays the 44.1kHz DTS test track, some VBRAltPresetStandard EAC-encoded Mp3's and some Windows Lossless songs through ReClock's rendering engine (forced in ReClock settings) with a delayed start, a pop now and then, and a few skips here and there, but once it's going, it sounds equally as fantastic as Foobar2000.

Finally, my problem so far: no MCE bit-perfection with Chaintech yet

Media Center, however, tells me when I look for my normal collection of WMlossless and VBR MP3 files that the files are "missing or corrupt" which they most certainly aren't, and no audio plays at all. I remember someone mentioning in some Envy or Chaintech AV710 thread that when files are played on a drive other than the primary C: drive in the same machine or over a network, problems arise. Perhaps this is the cause, as my audio files reside on a 400GB SATA drive I installed solely for media storage. Unless someone (thomaspf?) knows something I don't in that I haven't reset MCE properly.

I'm still awaiting my M-Audio transit (I know it will jitter thomas, but at least it will hopefully play MCE audio at bit-perfect 44.1kHz ;) ). I'll install that and run same tests as above when I get it and report here.

In the meantime, does anyone have any ideas about making ReClock and MCE happy together?

Thanks,
Adam

I have no more data. I have only tested this with Windows Media player not with MCE.

I was under the apparently wrong impression that this trick would work there as well.

Thomas

I have no more data. I have only tested this with Windows Media player not with MCE.

I was under the apparently wrong impression that this trick would work there as well.

Thomas

From Ogo's own ReClock forums on his site, I understand that he's working on MCE compatibility right now and that several users have in fact, made it work. I'm still not sure what's messing up my config, we'll see. I should receive my M-Audio Transit later this week and will report in here.

Thomas thanks as always for your input.

-Adam

I recently installed the "Transit-plus-onboard-sound" solution suggested above and it works flawlessly. I tested with the DTS test track linked above.

Big improvement in sound quality over the onboard-only setup.

DE

I assume those using two sound devices are not using MCE for all media playback? Unless I'm mistaken, there is no way to set video playback to use one device and music to use another. Thats one thing I liked about Meedio, you could specify different sound devices for different media.

I recently installed the "Transit-plus-onboard-sound" solution suggested above and it works flawlessly. I tested with the DTS test track linked above.

Big improvement in sound quality over the onboard-only setup.

DE

Fantastic! Looks like I'll be relegating my Chaintech to another system or to supplemental duty when my Transit arrives. Victory is ours. :-)

-Adam

I assume those using two sound devices are not using MCE for all media playback? Unless I'm mistaken, there is no way to set video playback to use one device and music to use another. Thats one thing I liked about Meedio, you could specify different sound devices for different media.

Meedio and MediaPortal both have that functionality I believe. In my case, it's a matter of the wife's comfort and "training" on MCE and my desire to boost audio performance without teaching her a new system. My path will be:

DVD and HDTV with multichannel audio -> MyHD MDP-130 -> onboard sound SPDIF out -> DVD input on my Denon receiver

MCE sound from Standard Def TV, videos and 2-channel music -> M-Audio Transit USB -> CD input on my Denon Receiver

So that's my setup to be.

-Adam

I also got "Transit-plus-onboard-sound" soultion. Works great with MCE, except whenever I use Transit in MCE's media player, I lose about half a second to a second of sound at very beginning of each song I played. Both no problem at all if I use Foobar or JR Media Center.

I also tried ReClok on MCE. While the audio part work, it causes the live TV to meltfunction. That's why I decide to go with Transit solution.

Just double-checked and apparently Bob's guidelines indicate that we are allowed to post parts deals in the HTPC forum, so here ya go:

http://www.zipzoomfly.com/jsp/ProductDetail.jsp?ProductCode=123304

Where did you see $129? The MSRP on M-Audio's site is only $99 I think...

Anyway, good luck!

-Adam

Missed your reply. I'm talking about the Audiophile USB, not the Transit. I have a Transit. I was the guinea pig who determined the Transit will give you bit perfect PCM with MCE. I recently went exclusivly to MCE and don't want to change settings in the Transit's control panel. I have a Audiophile USB on the way so we will see how it works. Yet another sound device to throw into the pile ;)

http://www.zipzoomfly.com/jsp/ProductDetail.jsp?ProductCode=361125

Missed your reply. I'm talking about the Audiophile USB, not the Transit. I have a Transit. I was the guinea pig who determined the Transit will give you bit perfect PCM with MCE. I recently went exclusivly to MCE and don't want to change settings in the Transit's control panel. I have a Audiophile USB on the way so we will see how it works. Yet another sound device to throw into the pile ;)

http://www.zipzoomfly.com/jsp/ProductDetail.jsp?ProductCode=361125

Ah yes, now I understand why your price was higher. I see no reason why that one shouldn't work too, as it's the USB interface and 44.1kHz hardware compatibility that creates the magic formula I believe. Do let us know how that one works out when you get a chance. Adding to the list of successes is a good thing. :)

What's your planned signal path setup, out of curiosity?

-Adam

I am setting up a HTPC and am interested in the best sound possible for my MCE PC. Earlier in this thread it was stated by a poster that he used onboard SPDIF for DVD/HDTV/ Games and the "transit and onboard" solution for CD's. I am so sorry to be a pain, but if I buy the Chaintech for DVD playback and HD video playback, will the sound suffer of will that card just not work with MCE? I am having a hard time understanding what to expect if I buy the Chaintech for my MCE.

THanks in advance.

AC3/DTS will be fine thru the Chaintech or onbaord sound. PCM audio will probably be better thru the Chaintech because music isn't being resampled to 48KHz. Its still going thru the Kmixer though which is said to degrade sound. The only way to get bitperfect PCM from MCE seems to be thru *some* USB audio devices. If all you care about is AC3/DTS from DVDs and game audio, the onboard sound is fine. Don't bother with another card/device. If you want high quality music playback, the Chaintech only gets you halfway there when using MCE.

Hi all,

M-Audio Transit USB - Bit-perfect Audio in MCE 2005 Success
Happy to report that I downloaded fresh M-Audio Transit drivers, installed and rebooted, then connected my Transit and rebooted for good measure, and Voila! I have bit-perfect audio coming right out of WMP 10 and Media Center itself! My lossless and VBR alt-preset standard audio rips sound fantastic. Hooray!

Odd Problem Likely Unique to My System - Help Anyone?
But, there's a caveat. This is wierd and it happened after I installed the Chaintech as well, and when I uninstalled the Chaintech and its associated drivers, the problem went away:

After installing the sound card, the TV streams from both My Hauppauge PVR150 *and* my MyHD MDP-130 stop displaying, reporting "signal strength too low." Obviously, as the MyHD is completely independent of Media Center, my possible theories are that:

1) My weak power supply is pushed too hard when I add either the Chaintech or Transit USB and somehow this keeps the live signal from passing through properly.

2) A PCI Bus or IRQ Conflict (do those still happen in WinXP?) is created resulting in the problem.

3) Somehow, the TV signal is degraded by adding a sound card with optical out. (I find this highly unlikely as optical wires have no RF interference and I'm shielded and cable-tied well, and because the problem repeats across both a PCI device (Chaintech) and USB device (Transit) and across MyHD and The Hauppauge tuner).

Anyone have any theories to help me troubleshoot this problem?

My system is an older A7N8X Deluxe Nforce2 mobo with an Athlon XP 2400+. The power supply reports the following in Speedfan, and I don't know enough to know if this is too low:

+3.3v: 3.31
+5v: 4.95
+12v: 12.22
-12v: -11.39
-5v: -4.84

Thoughts? Ideas? Help please!

Edit: Reinstalled MCE and completely uninstalled Chaintech drivers and all is well again, not sure what the problem was. Perhaps some piece of beta audio software I was tinkering with left behind some gremlins. Wiped clean, sounds great now.

Thanks,
Adam

I assume those using two sound devices are not using MCE for all media playback? Unless I'm mistaken, there is no way to set video playback to use one device and music to use another. Thats one thing I liked about Meedio, you could specify different sound devices for different media.

I'm using MCE for all playback. Within Windows Media Player, you can select the sound device you want WMP to use. This can be different from the Windows default sound device. If the device (Transit) is selected in WMP, then MCE uses it for My Music only. The default device (on-board) is used for everything else.

Crazy - but it works...

DE

An intersting paragraph in the M-Audio Audiophile USB's manual.

"If you have this box (DD/DTS Pass-Thru) checked, you can still send stereo digital audio to the S/PDIF port without any problems."

I'm going to get ahold of one of these to see if it works. They are like $99 on eBay right now.


have you tied M-audio audiophile USB to get bit-perfect sound? It seems like a much better value than m audio transit and can be used for other purposes as well.
Thanks a lot!

Ruslan

The Audiophile USB doesn't work. You can't get bitperfect audio with it and no matter what the settings, MCE or WMP will not pass or even output AC3/DTS. I could get AC3/DTS with TT but who cares if you can't get bitperfect PCM. I can't tell you how many sound devices I have bought in the last 4 years. Luckly, I have a relative who wants it.

Any suggestions for someone who doesn't have a digital output with the onboard sound? I have a Dell 8400 and while it has 5.1 analog output, there is no digital output. I currently have the 5.1 analog outputs plugged into my receiver via a 5.1 analog input. Should I still be able to get pit-perfect sound simply by adding a USB Transit device?

Also, is there any benefit to adding TheaterTek for DVD playback when it comes to bit perfect sound. They have a TheaterTek with DTS encoders - should I look into that since I have only analog 5.1 outputs? As you can see, this whole Bit Perfect thing really throws me for a loop.

Thanks.

Haven't posted here for quite a while since my MCE has been working very well for months. My bit-perfect MCE setup is similar to most folks here. That is, onboard sound for 48khz and usb sound (cheap $20 at Fry's) for 44khz. Both play back DTS flawlessly. Only issue (small issue) is I have to use remote to switch between the 2 sound inputs on my receiver whenever I switch from movie to music and vise-versa. Not a big deal for me, but not 100%WAF.

So, here is a thought. Has anyone tried one of those cheap optical spliters? The ones that let you split one optical into 2. Sort of like an "Y" adapter. I remember seeing it online for < $10 but can't find it now.

I was thinking if you could connect both audio outputs from the PC (onboard and usb) to this "Y" adapter and only have one optical cable goiong into the receiver/amp. That way, you won't even have to use the remote to switch between the 2. Everything will just be automatic. I haven't tried this because my onboard S/PDIF is coax and the usb is optical. What do you think? Will combining 2 opticals completely mess up the digital signals even if only one will have active signal at a time?

I purchased a $6 Optical for the very same use and I am extremely satisfied. I don't have the equipment to measure the audio throughput, but on my $2000 HT setup, the sound is awesome.

Well, I just found this (http://www.soundprofessionals.com/cgi-bin/gold/item/SP-TOC-SPLIT) . Looks like what I said earlier might not work as it says:
only one device can be turned 'on' at a time.

With onboard sound and usb sound setup, both sound devices are 'on' at the same time. I've notice that even when no signal is being passed, and the device is turned on, there is still a light through the toslink cable. Thus I had doubts that simply combining the 2 toslink cables would work. Oh well.

Texasengineer, can you elaborate on your setup? It shouldn't be working according to the link above.

I would like to know this also. I currently have a macro on my remote for 'Music' and that works OK, but it would be nice if I didn't have to...

DE

PS - I found this item

35-442 FIBER-OPTIC TOSLINK "Y" ADAPTOR-WITH 3 TOSLINK FEMALE CONNECTORS $24.95

on this (http://www.electronicplus.com/content/ProductPage.asp?maincat=oc&subcat=oca) site

texasengineer, are you around? Please see my question 2 posts above.

I found several places that sell this toslink splitter/combiner for < $10:

http://www.partsexpress.com/pe/pshowdetl.cfm?&DID=7&Partnumber=180-964
http://www.cablestogo.com/product.asp?cat%5Fid=102&sku=27027
http://www.cablewholesale.com/cgi-bin/search.cgi?body=Search&text=toslink+splitter
http://www.monoprice.com/products/product.asp?c_id=104&cp_id=10401&cs_id=1040108&p_id=966&seq=1&format=2&style=
http://www.ramelectronics.net/html/audio_toslink_adapters.html#pof-810
http://www.trianglecables.com/pof-810.html
or several on e-bay (search for toslink splitter)

Anyone who has 2 toslink connectors care to invest <$10 and test? I would if I had 2 toslink connectors, but I have a toslink and a coax. Even if it doesn't work, you can still use it as a regular coupler. But if it does work, HOLY GRAIL!!!!

I have an M-Audio Delta Dio 24/96 card and my recollection is that it outputs bit-perfect audio... Has anyone else tried this card under MCE?

Thanks,
Jeff

I have an M-Audio Delta Dio 24/96 card and my recollection is that it outputs bit-perfect audio... Has anyone else tried this card under MCE?

Thanks,
Jeff

It outputs bitperfect with ASIO or keneral streaming. Neither of which are available in MCE.

Should I still be able to get pit-perfect sound simply by adding a USB Transit device?
Thanks.


I'm not sure if your question has been answered yet, but yes, you absolutely should. The little Transit USB will bypass windows completely and pump your bit-perfect audio at true 44.1kHz out of its digital audio jack. If you want 48kHz, you have to jump into the M-audio control panel to alter it accordingly (for Dolby Digital encoded info, etc.).

Good luck!

-Adam

I found another way to get bit-perfect from WMP. I've been using this method for a couple of years now, and never had a chance to share it.

I have 2 soundcards in my system. One M-Audio Delta DIO, the other an XPlosion 7.1 (former for music, latter for games and movies).

I use an M-Audio Delta DIO for music, but I installed the Hoontech DSP 24 Value drivers for it (it works!). The cards must use the same chip, but The Hoontech drivers do not use KMixer. You set the hoontech drivers to send WavOut to the SPDIF (as opposed to its Monitor Mixer). WMP then passes bit perfect audio (demonstrable using DTS tracks).

In WMP, I explicitly choose the SPDIF output of the Delta DIO as my output device (in WMP, Tools/Options/Devices/Speakers).

The default Windows sound card is my Xplosion, so all other audio, including DVD output, goes out my XPlosion. (This is because most the DVD decoders rely on the default Windows sound card only, even though you're technically using WMP within MCE 2005)

Both the DIO and the Xplosion are connected to my Surround Processor via SPDIF cables, and I just switch between the two devices using my remote to listen to music or plays games, watch a movie, etc.

If any of you guys still use the Delta DIO, it's definitely worth it.

Here's another tip for this card (and others like it that have digital input, and allow syncing to an external clock). Use TWO SPDIF cables. One going out from the sound card to your processor/receiver as normal. The other going out from the processor's digital out, back into the digital input of the sound card. Set the soundcard's drivers to use "External Master Clock". This forces the sound card to use exactly the same clock as the DAC in your processor/receiver, and it greatly reduces jitter. The difference can be very, very audible.

-Jeff

Thanks for sharing but some of us are already doing this except using onboard sound on the motherboard and an external M-Audio Transit usb sound (or in my case, a cheap $20 Fry's USB sound). We set them up in MCE just like you said and get perfect DTS music and movies. However, just like you, we still have to use the remote to switch between the 2 digital input sources on our receivers. We are trying to alleviate that altogether so that everything is automatic when switch between 44khz music and 48khz movies. I know, call me lazy but that is the last step for me to achieve 100% WAF (and yes I have a wife who regularly forgets to switch receiver input when switching from movies to music and vise versa). Thus, I have proposed the cheap optical combiner solution in a few posts above which I don't believe anyone has tried but I have doubts that it'll work.

I found another way to get bit-perfect from WMP. I've been using this method for a couple of years now, and never had a chance to share it.

I have 2 soundcards in my system. One M-Audio Delta DIO, the other an XPlosion 7.1 (former for music, latter for games and movies).

I use an M-Audio Delta DIO for music, but I installed the Hoontech DSP 24 Value drivers for it (it works!). The cards must use the same chip, but The Hoontech drivers do not use KMixer. You set the hoontech drivers to send WavOut to the SPDIF (as opposed to its Monitor Mixer). WMP then passes bit perfect audio (demonstrable using DTS tracks).

In WMP, I explicitly choose the SPDIF output of the Delta DIO as my output device (in WMP, Tools/Options/Devices/Speakers).

The default Windows sound card is my Xplosion, so all other audio, including DVD output, goes out my XPlosion. (This is because most the DVD decoders rely on the default Windows sound card only, even though you're technically using WMP within MCE 2005)

Both the DIO and the Xplosion are connected to my Surround Processor via SPDIF cables, and I just switch between the two devices using my remote to listen to music or plays games, watch a movie, etc.

If any of you guys still use the Delta DIO, it's definitely worth it.

Here's another tip for this card (and others like it that have digital input, and allow syncing to an external clock). Use TWO SPDIF cables. One going out from the sound card to your processor/receiver as normal. The other going out from the processor's digital out, back into the digital input of the sound card. Set the soundcard's drivers to use "External Master Clock". This forces the sound card to use exactly the same clock as the DAC in your processor/receiver, and it greatly reduces jitter. The difference can be very, very audible.

-Jeff

chung_chang,

Why not call any of the places selling the toslink y-splitter and ask them if your idea will work? You just need to know if the splitter is bi-directional.

The other option is to get a 2:1 toslink/spdif auto-sensing switch and pipe the outputs from either soundcard or motherboard to receiver input.

- Steve O.

I don't think either idea will work. You can't combine the digital signals with a simple Y-connector because both cards are always sending clock data. The auto switcher wont work for the same reason. Every auto switcher I have seen just looks for clock data. They don't look for which is playing music or a movie. They are desinged to switch inputs when sources are powered on and off.

Thanks for confirming my doubts.

I don't think either idea will work. You can't combine the digital signals with a simple Y-connector because both cards are always sending clock data. The auto switcher wont work for the same reason. Every auto switcher I have seen just looks for clock data. They don't look for which is playing music or a movie. They are desinged to switch inputs when sources are powered on and off.

For the people using analogue out, would it be possible/worth to use an M-Audio Transit USB adapter, connected digitally to the soundcard performing D/A conversion?

I'm thinking about a combination of Transit + X-Fi.

Or has anyone used the X-Fi directly with Reclock and Kernel Streaming through MCE?

I am having some problems with getting this Sound device to work within MCE 2005 or work period. I want to get the maximum pontential of this Sound Card. Here is my setup:

Mobo: ASUS A7N8X Deluxe w/ NForce 2 Sound
CPU : AMD 2600 XP
RAM : 1 gig
A/V Receiver: Pioneer VSX-1014
Speakers: Athena Audition Series 6.1 setup

I found another way to get bit-perfect from WMP. I've been using this method for a couple of years now, and never had a chance to share it.

I have 2 soundcards in my system. One M-Audio Delta DIO, the other an XPlosion 7.1 (former for music, latter for games and movies).

I use an M-Audio Delta DIO for music, but I installed the Hoontech DSP 24 Value drivers for it (it works!). The cards must use the same chip, but The Hoontech drivers do not use KMixer. You set the hoontech drivers to send WavOut to the SPDIF (as opposed to its Monitor Mixer). WMP then passes bit perfect audio (demonstrable using DTS tracks).

In WMP, I explicitly choose the SPDIF output of the Delta DIO as my output device (in WMP, Tools/Options/Devices/Speakers).

The default Windows sound card is my Xplosion, so all other audio, including DVD output, goes out my XPlosion. (This is because most the DVD decoders rely on the default Windows sound card only, even though you're technically using WMP within MCE 2005)

Both the DIO and the Xplosion are connected to my Surround Processor via SPDIF cables, and I just switch between the two devices using my remote to listen to music or plays games, watch a movie, etc.

If any of you guys still use the Delta DIO, it's definitely worth it.

Here's another tip for this card (and others like it that have digital input, and allow syncing to an external clock). Use TWO SPDIF cables. One going out from the sound card to your processor/receiver as normal. The other going out from the processor's digital out, back into the digital input of the sound card. Set the soundcard's drivers to use "External Master Clock". This forces the sound card to use exactly the same clock as the DAC in your processor/receiver, and it greatly reduces jitter. The difference can be very, very audible.

-Jeff


Jeff, excellent write-up and thank you for sharing this! I may add this to my post with methods for achieving bit-perfect output above.

Thanks,
Adam

I found another way to get bit-perfect from WMP. I've been using this method for a couple of years now, and never had a chance to share it.

I have 2 soundcards in my system. One M-Audio Delta DIO, the other an XPlosion 7.1 (former for music, latter for games and movies).

I use an M-Audio Delta DIO for music, but I installed the Hoontech DSP 24 Value drivers for it (it works!). The cards must use the same chip, but The Hoontech drivers do not use KMixer. You set the hoontech drivers to send WavOut to the SPDIF (as opposed to its Monitor Mixer). WMP then passes bit perfect audio (demonstrable using DTS tracks).

In WMP, I explicitly choose the SPDIF output of the Delta DIO as my output device (in WMP, Tools/Options/Devices/Speakers).

The default Windows sound card is my Xplosion, so all other audio, including DVD output, goes out my XPlosion. (This is because most the DVD decoders rely on the default Windows sound card only, even though you're technically using WMP within MCE 2005)

Both the DIO and the Xplosion are connected to my Surround Processor via SPDIF cables, and I just switch between the two devices using my remote to listen to music or plays games, watch a movie, etc.

If any of you guys still use the Delta DIO, it's definitely worth it.

Here's another tip for this card (and others like it that have digital input, and allow syncing to an external clock). Use TWO SPDIF cables. One going out from the sound card to your processor/receiver as normal. The other going out from the processor's digital out, back into the digital input of the sound card. Set the soundcard's drivers to use "External Master Clock". This forces the sound card to use exactly the same clock as the DAC in your processor/receiver, and it greatly reduces jitter. The difference can be very, very audible.

-Jeff

Thanks for the sharing, may I ask what processor offer digital output to clock slave your DIO 2496? Any DAC option around $500 which could do that?

Thanks for the sharing, may I ask what processor offer digital output to clock slave your DIO 2496? Any DAC option around $500 which could do that?

I'm using a Rotel 1066. I imagine any receiver, processor, or dac that has a digital out should work.

-Jeff

Thanks for sharing but some of us are already doing this except using onboard sound on the motherboard and an external M-Audio Transit usb sound (or in my case, a cheap $20 Fry's USB sound). We set them up in MCE just like you said and get perfect DTS music and movies. However, just like you, we still have to use the remote to switch between the 2 digital input sources on our receivers. We are trying to alleviate that altogether so that everything is automatic when switch between 44khz music and 48khz movies. I know, call me lazy but that is the last step for me to achieve 100% WAF (and yes I have a wife who regularly forgets to switch receiver input when switching from movies to music and vise versa). Thus, I have proposed the cheap optical combiner solution in a few posts above which I don't believe anyone has tried but I have doubts that it'll work.

Actually, if I were to only use the Delta Dio w/ Hoontech drivers, I wouldn't need the remote at all. The card auto-switches fine between PCM and AC-3 (Dolby/DTS) without manually changing anything in its control panel. I could just use it as my default Windows device and be done with it.

The reason I continue to use two soundcards (which necessitates the remote switch) is because I also use my HTPC for PC gaming. The Xplosion encodes my games into 5.1 DTS and sounds fantastic (and all over a single digital cable)

Now, if I could get the Xplosion to output bit-perfect audio, then I'd be all set. But it doesn't work, even when using "reclock".

Personally, I think that manual switch limitation with the transit is LAAAAME, and m-audio could fix it. The Delta Dio, Revolution, and Audiophile cards don't have that problem - so why the Transit?

One thing I like about the Transit and other external USB audio devices (assuming music PCM use only) is that you can place the device very far from the HTPC without losing any sound quality. You could use a 15 foot USB cable (or several of them if you use a USB repeater), and still get away with a 3 foot or less digital audio cable (coax or toslink). There's no loss in sound quality over a 15+ feet of USB like there could be with a 15+ digital audio cable.

Back on topic:

Sound professional sound cards (and their control panels) may allow you to do what you want. You'd need to run the digital output from Card A (any ol' card) into the digital input of Card B (a pro quality card). The digital output of Card B would go into your processor/receiver as normal. You'd use the "monitor mixer" or "mixing board" portion of Card B's control panel to send WAV audio out the SPDIF, **AND** to redirect the SPDIF Input out the SPDIF output as well. This should essentially "combine" them. You'd still need to explicitly set WMP to use your SPDIF of Card B, and Windows to use Card A - but your processor/receiver would only be on a single device (output of Card B).

The Hoontech control panel does not allow the above configuration exactly, so I can't prove it to be 100% true. However, I could have sworn the original M-Audio drivers/control panel would allow this. Someone with an RME or other monster pro card could test.

-Jeff

No, no....we are not switching between Dolby and DTS. I said we're switching between 44khz (music) and 48khz (movies dolby/DTS). Is your setup able to switch between 44khz and 48khz automatically without use of remote and 2 sound cards? Please let us know.

Actually, if I were to only use the Delta Dio w/ Hoontech drivers, I wouldn't need the remote at all. The card auto-switches fine between PCM and AC-3 (Dolby/DTS) without manually changing anything in its control panel. I could just use it as my default Windows device and be done with it.

The reason I continue to use two soundcards (which necessitates the remote switch) is because I also use my HTPC for PC gaming. The Xplosion encodes my games into 5.1 DTS and sounds fantastic (and all over a single digital cable)

Now, if I could get the Xplosion to output bit-perfect audio, then I'd be all set. But it doesn't work, even when using "reclock".

Personally, I think that manual switch limitation with the transit is LAAAAME, and m-audio could fix it. The Delta Dio, Revolution, and Audiophile cards don't have that problem - so why the Transit?

One thing I like about the Transit and other external USB audio devices (assuming music PCM use only) is that you can place the device very far from the HTPC without losing any sound quality. You could use a 15 foot USB cable (or several of them if you use a USB repeater), and still get away with a 3 foot or less digital audio cable (coax or toslink). There's no loss in sound quality over a 15+ feet of USB like there could be with a 15+ digital audio cable.

Back on topic:

Sound professional sound cards (and their control panels) may allow you to do what you want. You'd need to run the digital output from Card A (any ol' card) into the digital input of Card B (a pro quality card). The digital output of Card B would go into your processor/receiver as normal. You'd use the "monitor mixer" or "mixing board" portion of Card B's control panel to send WAV audio out the SPDIF, **AND** to redirect the SPDIF Input out the SPDIF output as well. This should essentially "combine" them. You'd still need to explicitly set WMP to use your SPDIF of Card B, and Windows to use Card A - but your processor/receiver would only be on a single device (output of Card B).

The Hoontech control panel does not allow the above configuration exactly, so I can't prove it to be 100% true. However, I could have sworn the original M-Audio drivers/control panel would allow this. Someone with an RME or other monster pro card could test.

-Jeff

No, no....we are not switching between Dolby and DTS. I said we're switching between 44khz (music) and 48khz (movies dolby/DTS). Is your setup able to switch between 44khz and 48khz automatically without use of remote and 2 sound cards? Please let us know.

That's what I said...

Like yours, my setup does not - but only because I use two soundcards.

However, if I used only my Delta Dio, I would not need to switch via the remote. The Delta Dio (and most, if not all, m-audio cards) automatically switches between PCM (music) and AC3 (movies, Dolby/DTS). The Transit does not (that sucks)

-Jeff

I think I'm getting confused. I thought DVD also supports PCM at 48Khz, no?

Anyways, can you play a DTS music at 44khz and then a DTS movie DVD at 48 khz back to back inside MCE without touching any settings or switching inputs on your receiver using M-Audio Delta DIO alone? If so, THANK YOU!!!! Please let us know.

That's what I said...

Like yours, my setup does not - but only because I use two soundcards.

However, if I used only my Delta Dio, I would not need to switch via the remote. The Delta Dio (and most, if not all, m-audio cards) automatically switches between PCM (music) and AC3 (movies, Dolby/DTS). The Transit does not (that sucks)

-Jeff

I'm trying to find the hoontech drivers you used for your M-audio delta dio... i found 2 drivers:

E-WDM Drivers for Windows 2000/XP new 8.2rc5 1.3MB 03/02/2004 readme dsp24ewdm82rc5.zip

Drivers/Applications for Windows 2000/XP 7.2.1014 668kB 10/14/2002 readme dspw1022.zip

Which ones did you use?

Thanks,
Jeff

I think I'm getting confused. I thought DVD also supports PCM at 48Khz, no?

Anyways, can you play a DTS music at 44khz and then a DTS movie DVD at 48 khz back to back inside MCE without touching any settings or switching inputs on your receiver using M-Audio Delta DIO alone? If so, THANK YOU!!!! Please let us know.

Yes, DVD supports anything it wants (PCM, Dolby Digital, DTS) - whatever. Let me be more specific - YES - I can play a DTS music track (which is technically "hidden" inside a 44.1khz PCM track) and then fire up a movie and listen to Dolby or DTS, without touching a thing. I haven't done this in a long time, as I now use two soundcards. I will check it again, to be absolutely sure. I will let you know. If you don't hear back from me by the end of the day, pmail me to remind me.

-Jeff

I'm trying to find the hoontech drivers you used for your M-audio delta dio... i found 2 drivers:

E-WDM Drivers for Windows 2000/XP new 8.2rc5 1.3MB 03/02/2004 readme dsp24ewdm82rc5.zip

Drivers/Applications for Windows 2000/XP 7.2.1014 668kB 10/14/2002 readme dspw1022.zip

Which ones did you use?

Thanks,
Jeff


Neither. Use these:

ST Audio (Hoontech) Drivers (http://www.staudio.de/data/dspw1022.zip)

Unzip the file. There is no setup program, you can uninstall the m-audio drivers (preferred), reboot, and use "Have Disk" when windows finds the "new" device. OR, you can go into Device Manager, and choose Change Driver on the Delta Dio. I prefer to uninstall the m-audio drivers because if you don't, I beleive it leaves the control panel applet hanging around (harmless, but annoying)

When you select the driver location, you may be presented with a choice of ST Audio cards. Choose the DSP 24 Value, if asked.

Once installed, open the ST Audio control panel applet and choose "Wav SPDIF Out" for "SPDIF Out". And in the Hardware Settings section, select at least 256 for the buffer size. The settings will stick between reboots, so you should only have to do this once. Always test bit-perfect with a DTS track (especially after your first reboot). If you hear nothing but hiss, make sure you have "Wav SPDIF Out" selected, and not "Monitor Mixer". Also, make sure that the SPDIF is your default Windows audio device, or explicitly select it in Windows Media Player (Tools/Opions/Devices/Speakers). Whenever you change the device in WMP, you must restart it.

-Jeff

If you can confirm it works, I will be very very happy!!!! Please do let us know either way.

Yes, DVD supports anything it wants (PCM, Dolby Digital, DTS) - whatever. Let me be more specific - YES - I can play a DTS music track (which is technically "hidden" inside a 44.1khz PCM track) and then fire up a movie and listen to Dolby or DTS, without touching a thing. I haven't done this in a long time, as I now use two soundcards. I will check it again, to be absolutely sure. I will let you know. If you don't hear back from me by the end of the day, pmail me to remind me.

-Jeff

Jeff -

Thanks for your help on this. Here's where I stand:
1. Delta Dio is installed with the Hoontech drivers. Setup as you said
2. Tested inside WMP with DTS test, worked perfectly. My surround processor shows DTS 3/2 exactly as it should and sound comes out each channel properly.
3. Ditto inside MCE
4. Movie playback works, but not getting the correct audio output - i think it's sticking to 44.1 khz - my surround processor shows Digital PCM and doesn't see it as dolby digital (this worked perfectly with the sound blaster).
5. Updated my purevideo decoders in hopes that that would help (using .196 now). No luck.

Any idea how to get the correct audio output from my movies/videos ? Considering this is what i use MCE for the most.... help would be very much appreciated.

Also, there is some screwyness with MCE - when you start it up, it thinks it's in mute mode (although sound comes out ok). Movies play with closed captioning (which i just shut off). Clicking on the unmute button sets the volume level to 0 - have to go to the control panel to push it up to max as MCE won't allow you to. This isn't a big deal, but the lack of Dolby Digital output for movies really is...

Thanks!
Jeff

OK, so I updated to Rollup2 and all the latest security fixes/patches. Still having the same problems - No DD/DTS only PCM.

Come to think of it, I've NEVER been able to get MCE to output DD/DTS! If I exit MCE's interface and run TheaterTek, it outputs everything fine... so... what's MCE doing wrong?

Thanks,
Jeff

Well, I'm sorry to say, I could not get the m-audio working properly for anything but PCM output. I'm back to my Audigy 2ZS - getting AC3 playback through SPDIF - sounds great.
No bit-perfect music... oh well. I guess i could spend the time getting the m-audio working for music playback and using the 2ZS for everything else, but honestly, the music sounds pretty darn good as is... As i get older, it really becomes less of an issue!

Yes, DVD supports anything it wants (PCM, Dolby Digital, DTS) - whatever. Let me be more specific - YES - I can play a DTS music track (which is technically "hidden" inside a 44.1khz PCM track) and then fire up a movie and listen to Dolby or DTS, without touching a thing. I haven't done this in a long time, as I now use two soundcards. I will check it again, to be absolutely sure. I will let you know. If you don't hear back from me by the end of the day, pmail me to remind me.

-Jeff

Jeff, thanks for your contribution to solving the bit-perfect puzzle. I did want to clarify one thing: the main thrust of this thread is to achieve bit-perfect audio at 44kHz and 48kHz from playback of Windows XP Media Center Edition, not from any other application such as Theatretek, foobar, or JRiver media center. Given the post above this one, however, I'm worried that you possibly spoke too soon when assuming that the old M-Audio Delta DIO could bypass Kmixer.sys from Windows XP Media Center Edition's playback.

From Thomaspf's posts here and elsewhere (http://www.hydrogenaudio.org/forums/index.php?showtopic=5755&pid=58590&mode=threaded&show=&st=&#) , we know the older Delta DIO works as a 44.1kHz output device with JRiver Media Center using ASIO/kernel streaming output, even without the Hoontech drivers, because the card itself mixes and sets the digital frequency of its output when passed an appropriate signal without any manual switching by the user. However, I believe that not all applications are capable of getting an unmixed signal to the card (bypassing kmixer). Apparently, the "magic" of the card you are speaking of is best described in this excerpt from an audiomidi.com review:

"The Delta DiO 2496 even offers onboard digital mixing and routing, monitor control, SCMS (Serial Copy Management System) output on S/PDIF, and multi-platform operation."

Thus, the crystal of this card can control the digital frequency output, enabling proper switching between 44kHz and 48kHz from within the card itself, when it knows what kind of signal it has received. But the issue here is getting around Kmixer.sys using Windows Media player and Windows XP MCE's player, and so far, we have no direct reports of this working without "hard" switching the Transit USB.

If I misunderstood and you have already done this, I apologize, but would you volunteer to test this card's output with AC3 content as well as 44.1kHz DTS wav content (http://www.sr.se/laddahem/MultiKanal/Dts/SURROUNDTEST_011212.zip) using the latest version of XP MCE with rollups/patches? Until you or someone else verifies this as a known working solution, I think it best to remain skeptical and content ourselves with the 2-card or manual output switching solution that at under $80, only the M-Audio Transit USB (or a similar cheap USB digital sound card?) allows.

By the way, the newer M-Audio Audiophile card has been tested and we have a report from MadChemist that it does not work (http://www.avsforum.com/avs-vb/showthread.php?p=7270072&&#post7270072) above.

Cheers,
Adam

Jeff, was just wondering if you had a chance to test it out yet. Appreciate you taking the time and let us know whatever you find out.

... only the M-Audio Transit USB (or a similar cheap USB digital sound card?) allows...

My setup uses PIAA Inc usb sound card that can be purchased from Fry's Electronics store for $40. I was mistaken about it being $20. That was for another usb sound that I bought which didn't work. My friend just bought the same PIAA from Fry's couple of days ago and got bit-perfect DTS music and movies under MCE (using 2 sound card method).

My setup uses PIAA Inc usb sound card that can be purchased from Fry's Electronics store for $40. I was mistaken about it being $20. That was for another usb sound that I bought which didn't work. My friend just bought the same PIAA from Fry's couple of days ago and got bit-perfect DTS music and movies under MCE (using 2 sound card method).

Good to hear, sounds like we might be able to add another USB card to the list. Did your friend use the sound test file above to test 2-channel DTS output and verify 44.1kHz fidelity?

Cheers,
Adam

Good to hear, sounds like we might be able to add another USB card to the list. Did your friend use the sound test file above to test 2-channel DTS output and verify 44.1kHz fidelity?

Cheers,
Adam


He used a DTS music CD which only produced noises before using the PIAA. I used a ripped DTS music wave. Both of our receivers' DTS indicator came on and my Panny showed 44khz. His Pio does not have khz indicator.

Is that good enough or should both of us try that test sound file also? You are talking about this (http://www.sr.se/laddahem/MultiKanal/Dts/SURROUNDTEST_011212.zip)?

He used a DTS music CD which only produced noises before using the PIAA. I used a ripped DTS music wave. Both of our receivers' DTS indicator came on and my Panny showed 44khz. His Pio does not have khz indicator.

Is that good enough or should both of us try that test sound file also? You are talking about this (http://www.sr.se/laddahem/MultiKanal/Dts/SURROUNDTEST_011212.zip)?

Sounds like a working unit to me. Yes, that is the file I use to test, which is a 44.1kHz DTS-encoded .wav file that would also "light up" your receiver's DTS indicator. If you can hear it, you've got 44.1kHz bit-perfect audio. If it was passing through kmixer.sys and mangled to 48kHz, you'd hear sound corruption and noise.

Would you please list the exact manufacturer part, model number and source for this "PIAA" card you're using? Thanks for the input!

Cheers,
Adam

Sounds like a working unit to me. Yes, that is the file I use to test, which is a 44.1kHz DTS-encoded .wav file that would also "light up" your receiver's DTS indicator. If you can hear it, you've got 44.1kHz bit-perfect audio. If it was passing through kmixer.sys and mangled to 48kHz, you'd hear sound corruption and noise.

Would you please list the exact manufacturer part, model number and source for this "PIAA" card you're using? Thanks for the input!

Cheers,
Adam


Sorry. The company is actually called PPA Int'l. Here is the link to the exact usb sound (http://www.ppa-usa.com/product_pages/soundcard/1455.htm) I have.

I also did test out the DTS wav file above and it worked, though I could not understand the language.

If we don't hear back from Jeff on his single card solution, I won't be sweating on this much longer since Vista is right around the corner. And from what I've read on this forum, Vista appears to be more bit-perfect sound friendly.

Sorry. The company is actually called PPA Int'l. Here is the link to the exact usb sound (http://www.ppa-usa.com/product_pages/soundcard/1455.htm) I have.

I also did test out the DTS wav file above and it worked, though I could not understand the language.

If we don't hear back from Jeff on his single card solution, I won't be sweating on this much longer since Vista is right around the corner. And from what I've read on this forum, Vista appears to be more bit-perfect sound friendly.

Added your card note to my original post above, thanks for the link. Yep, Vista will apparently have an "exclusive" audio mode indeed that locks the output at a given frequency. Kmixer.sys is gone. Of course, release version of Vista is likely a year off given Microsoft's desire to work out all the kinks, I'd say. But for the next year, we're enjoying bit-perfect audio anyway with our USB solutions, eh? :-)

Cheers,
Adam

Correction...

I am able to get bit-perfect audio out Windows Media player and MCE using the Delta Dio (no ASIO needed). You just need to install the aforementioned Hoontech/ST Audio drivers instead of M-Audio's and select the SPDIF Out. I've been doing this for years - bit perfect audio in MCE, easy. Demonstrable via a DTS CD. Period.

However - (and here's my correction) - I cannot get AC-3 (Dolby Digital or DTS) out of WMP or MCE. HOWEVER - this appears to be more a problem with the audio decoders than it is the device. If I use the PowerDVD application, it works fine. No matter what I do (including manually modding the registry), I cannot get Dolby/DTS passthrough out WMP/MCE or even GraphEdit using the nVidia, PowerDVD, or WinDVD decoders. BUT IT IS TECHNICALLY POSSIBLE, since it can be done in PowerDVD.

That's a real bummer. Bit perfect audio via WMP/MCE using Delta Dio. AND Dolby/DTS passthrough - just not in WMP/MCE. The card apparently auto-detects and switches between PCM and AC3 too - it's just not in the one app (MCE) :(

Sorry to have misinformed anyone. It's been a long time since I did such testing and comparisons. I still use (and prefer to) two sound cards. I use the Delta Dio w/ ST Audio drivers for bit-perfect audio, and an HDA Xplosion 7.1 for games, windows audio, movies. I explicitly select the ST Audio SPDIF in WMP, and the Xplosion is my default windows device. I simply switch devices on my receiver/processor to switch between the two.

Note for DCypher: You CAN get bit perfect audio from an Audigy 2. Go to Start/Programs/Creative/Soundblaster Audigy 2 and run "Creative Audio Console". On the "SPDIF I/O" tab, choose "44.1 Khz". On the "Bit Accurate" tab, select "Enable Bit Accurate Playback". The latter checkbox, however, MAY be reset when you reboot the machine (check after your first reboot). When these two settings are selected, you can get bit perfect audio over the SPDIF in WMP/MCE and anything else (without ASIO or Kernel Streaming)

-Jeff

Jeff, thanks for your contribution to solving the bit-perfect puzzle. I did want to clarify one thing: the main thrust of this thread is to achieve bit-perfect audio at 44kHz and 48kHz from playback of Windows XP Media Center Edition, not from any other application such as Theatretek, foobar, or JRiver media center. Given the post above this one, however, I'm worried that you possibly spoke too soon when assuming that the old M-Audio Delta DIO could bypass Kmixer.sys from Windows XP Media Center Edition's playback.


Just to reiterate from my previous post. I did not speak too soon :) The M-Audio Delta DIO does bypass Kmixer IF you install the Hoontech/ST audio drivers for it, instead of M-Audio's. I've been using this setup for over 2 years. And this is in WMP, MCE, and any other player, for that matter. ASIO and Kernel Streaming are no required. I only use MCE to listent to my bit perfect music.

-Jeff

Does anyone use Girder to switch inputs on your external process/receiver between the mobo digital out and the M Audio USB device? I'm wondering if this could automate switching the reciever's input based upon whether you're playing movies or music in MCE?

In the alternative, could I use the M Audio USB device for bit-perfect audio in MCE, and continue to use the M Audio for DD/DTS for movies by launching movies in theatertek from MCE via the MyMovies plugin?

For what its worth, I don't yet have MCE, but I am considering it for my next HTPC build. I currently have a M Audio Revo.

Note for DCypher: You CAN get bit perfect audio from an Audigy 2. Go to Start/Programs/Creative/Soundblaster Audigy 2 and run "Creative Audio Console". On the "SPDIF I/O" tab, choose "44.1 Khz". On the "Bit Accurate" tab, select "Enable Bit Accurate Playback". The latter checkbox, however, MAY be reset when you reboot the machine (check after your first reboot). When these two settings are selected, you can get bit perfect audio over the SPDIF in WMP/MCE and anything else (without ASIO or Kernel Streaming)

-Jeff

I wasn't aware you could do this on the Audigy 2ZS - i will have to check it out. Do you know if this will allow AC3 passthru on the card, or do i need to manually switch it back when i want DD/DTS?

Thanks,
Jeff

I have great news!

I bought an MSI K8NGM2-FID mobo with AMD64 3000+ cpu to replace my aging Media Center PC. I just finished the build a couple of hours ago and one of the first things I tested was the onboard RealTek ALC 880 sound via coax spdif to my Panny digital receiver. And guess what.....

THIS ONBOARD SOUND ON A CHEAP $80 MOBO DOES BIT-PERFECT SOUND IN MCE FOR BOTH 48khz AND 44khz AND AUTO-SWITCHES!!!!!!!

That's right folks! I played "Toy Story" (48khz DTS) and the sample DTS wave (44khz) and both sounds come out perfect. My receiver's DTS light comes on and the frequency indicator changes between 48khz and 44khz as I switch back and forth.

I'm a very happy camper right now! No need to install my USB sound anymore. I still have yet to figure out how to bring the PC out of S3 standby using my Harmony remote. So, I'm back to working on it some more...

If you want to find out more about my setup, there is an excellent guide here (http://forums.anandtech.com/messageview.aspx?catid=29&threadid=1803985&enterthread=y) which I followed (just read the first post).

Congrats. What app did you use to play it?
I can get 5.1 when using WinDVD, but not when using MCE.
Any idea how I can get it to work?

I have great news!

I bought an MSI K8NGM2-FID mobo with AMD64 3000+ cpu to replace my aging Media Center PC. I just finished the build a couple of hours ago and one of the first things I tested was the onboard RealTek ALC 880 sound via coax spdif to my Panny digital receiver. And guess what.....

THIS ONBOARD SOUND ON A CHEAP $80 MOBO DOES BIT-PERFECT SOUND IN MCE FOR BOTH 48khz AND 44khz AND AUTO-SWITCHES!!!!!!!

That's right folks! I played "Toy Story" (48khz DTS) and the sample DTS wave (44khz) and both sounds come out perfect. My receiver's DTS light comes on and the frequency indicator changes between 48khz and 44khz as I switch back and forth.

I'm a very happy camper right now! No need to install my USB sound anymore. I still have yet to figure out how to bring the PC out of S3 standby using my Harmony remote. So, I'm back to working on it some more...

If you want to find out more about my setup, there is an excellent guide here (http://forums.anandtech.com/messageview.aspx?catid=29&threadid=1803985&enterthread=y) which I followed (just read the first post).

Does 48kHz AC3 (Dolby Digital 5.1) pass through properly as well?

Great Anand link.

-Adam

I used MCE 2005 Rollup 2 with nVidia PureVideo set to spdif out to test everything.

It has played everything I've thrown at it so far:

44khz DTS (sample and ripped DTS music)
48khz DTS 5.1 ES (Toy Story 1)
48khz DTS 5.1 (Santana Live Concert)
48khz DD 5.1 EX (Finding Nemo)
48khz DD 5.1 (most movies and HDTV)
.....
have I missed anything? Oh, and of course your typical stereo stuff.

I still can't get over how the sound card just simply locks in to the new sampling rate automatically without me doing anything! There are no skipping, stuttering or anything and believe me, I've sat and watched plenty over the weekend.

Now, isn't RealTek ALC880 a common sound chip on many newer motherboards? If so, then couldn't all of them do this? The driver came from RealTek, not something MSI hacked up. Are there any standalone PCI sound cards with this chip also? Then can't those do bit-perfect sound on any mobo then?

All Motherboards with HD-audio support and the Realtek 88x codecs are bit-perfect in my experience.

Even my new laptop with an HD-audio Realtek sound chip provides bit transparent playback on it's S/PDIF output.

Cheers

Thomas

would these solutions work for playing back wma lossless files?

They work for me in WMA lossless, FlAC, APE, etc..

Cheers

Thomas

All Motherboards with HD-audio support and the Realtek 88x codecs are bit-perfect in my experience.

Even my new laptop with an HD-audio Realtek sound chip provides bit transparent playback on it's S/PDIF output.

Cheers

Thomas

Fascinating ...how did I miss the development (and release) of this apparently much-improved PC Audio specification?! I have not seen major press on this spec, which apparently has existed since 2004, but I'd say this amounts to a minor revolution in PC audio in stepping away from the old AC'97 spec and supporting bit-perfect audio out of the box. Admittedly, we're not mainstream audio consumers here. ;-)

Interesting read here: Intel HD-Audio Specification (ftp://download.intel.com/standards/hdaudio/pdf/HDAudio_03.pdf)
From this page: Intel HD-Audio (http://www.intel.com/standards/hdaudio/)

Thomas, do you know if this implements bit-perfect output at the windows driver level as well? How does this get around Kmixer.sys?

Might just be time to upgrade my venerable Asus Nforce2 mobo in the HTPC...

Best,
Adam

I used MCE 2005 Rollup 2 with nVidia PureVideo set to spdif out to test everything.

It has played everything I've thrown at it so far:

44khz DTS (sample and ripped DTS music)
48khz DTS 5.1 ES (Toy Story 1)
48khz DTS 5.1 (Santana Live Concert)
48khz DD 5.1 EX (Finding Nemo)
48khz DD 5.1 (most movies and HDTV)
.....
have I missed anything? Oh, and of course your typical stereo stuff.

I still can't get over how the sound card just simply locks in to the new sampling rate automatically without me doing anything! There are no skipping, stuttering or anything and believe me, I've sat and watched plenty over the weekend.

Now, isn't RealTek ALC880 a common sound chip on many newer motherboards? If so, then couldn't all of them do this? The driver came from RealTek, not something MSI hacked up. Are there any standalone PCI sound cards with this chip also? Then can't those do bit-perfect sound on any mobo then?

That's pretty much the gamut of normal digital audio to throw at it. Good show!

I'll try to do a little more research to verify a few more motherboard/chipset/driver combinations before we pronounce HD-Audio as the magic formula, though it sure sounds like we have another winner.

Sounds like I'll be updating my post above with another solution for those with newer HD-Audio compliant motherboards...congratulations and thanks for sharing this find!

-Adam

No prob. I wasn't expecting this board to do bit-pefect but when I saw a sampling rate "auto lock" button in the driver, it got me curious. Glad I tested it. Now, I have to untrain my wife from switching the receiver inputs.

All Motherboards with HD-audio support and the Realtek 88x codecs are bit-perfect in my experience.

Even my new laptop with an HD-audio Realtek sound chip provides bit transparent playback on it's S/PDIF output.

Cheers

Thomas

My motherboard has a Realtek ALC658, and is supposedly HD audio (though not sure). I installed the latest Realtek drivers (A387 = 5.10.0.6070) and I don't see any sampling rate auto lock feature in the properties page. The ALC658 specs say it supports 44.1k and 48k but there is no software setting visible... I cannot get passthrough of DTS wav files. Any tips?

Thanks,
Jason

All Motherboards with HD-audio support and the Realtek 88x codecs are bit-perfect in my experience.

Even my new laptop with an HD-audio Realtek sound chip provides bit transparent playback on it's S/PDIF output.

Cheers

Thomas

Does anyone know if the Realtek 850 is bit-perfect? I am looking for a new motherboard and the only ones that are nForce4 Ultra seem to have the Realtek 850 on-board sound. I would like to find an 880 board but I don't need the matx form factor and would rather have 4 pci slots.

Any board suggestions?

Aaron

My motherboard has a Realtek ALC658, and is supposedly HD audio (though not sure). I installed the latest Realtek drivers (A387 = 5.10.0.6070) and I don't see any sampling rate auto lock feature in the properties page. The ALC658 specs say it supports 44.1k and 48k but there is no software setting visible... I cannot get passthrough of DTS wav files. Any tips?

Thanks,
Jason

You are using a newer driver than mine. I think mine is 5.10.0.5354 ? (R1.37)

I double-click the audio icon in the taskbar, then click Audio I/O, Digital, there is a button in the lower right hand corner for Auto-lock. I had to click on it once after installing the driver, then everything worked automatically after that. Good luck.

Oh, too bad. I tried A387 5.10.0.6070 and R137 5.10.0.5910 and my menus look different than yours. There is no "Audio I/O" tab. Just "SPDIF In" and "SPDIF Out" and no auto lock option. Was it really 5.10.0.5354 that you have? I coudn't find it. It's probably my chip, not the drivers, but one can always hope...
Thanks,
Jason

Does anyone know if the Realtek 850 is bit-perfect? I am looking for a new motherboard and the only ones that are nForce4 Ultra seem to have the Realtek 850 on-board sound. I would like to find an 880 board but I don't need the matx form factor and would rather have 4 pci slots.

Any board suggestions?

Aaron

In case anyone is interested, I did find another RealTek 880 motherboard with no on-board graphics and more pci slots at NewEgg:

AOpen nCK804Ua-LFS Socket 939 NVIDIA nForce4 Ultra ATX AMD Motherboard

But it has some other faults. So, I'm still looking around.

Aaron

Oh, too bad. I tried A387 5.10.0.6070 and R137 5.10.0.5910 and my menus look different than yours. There is no "Audio I/O" tab. Just "SPDIF In" and "SPDIF Out" and no auto lock option. Was it really 5.10.0.5354 that you have? I coudn't find it. It's probably my chip, not the drivers, but one can always hope...
Thanks,
Jason

Sorry. It's 5.10.0.5253.

Attached is a screenshot.

I think the ALC658 and ALC850 are older AC97 sound chips.

The nforce4 chipset does not support HD-audio anyway.

Sorry

Thomas

I used MCE 2005 Rollup 2 with nVidia PureVideo set to spdif out to test everything.

It has played everything I've thrown at it so far:

44khz DTS (sample and ripped DTS music)
48khz DTS 5.1 ES (Toy Story 1)
48khz DTS 5.1 (Santana Live Concert)
48khz DD 5.1 EX (Finding Nemo)
48khz DD 5.1 (most movies and HDTV)
.....
have I missed anything? Oh, and of course your typical stereo stuff.

I still can't get over how the sound card just simply locks in to the new sampling rate automatically without me doing anything! There are no skipping, stuttering or anything and believe me, I've sat and watched plenty over the weekend.

Now, isn't RealTek ALC880 a common sound chip on many newer motherboards? If so, then couldn't all of them do this? The driver came from RealTek, not something MSI hacked up. Are there any standalone PCI sound cards with this chip also? Then can't those do bit-perfect sound on any mobo then?

Does anything happen if you reduce the volume below 100% in MCE while playing 44 kHZ DTS audio? If not, that means that DTS WAV is somehow recognized as encoded material and is being passed through unmolested.

However, I'm still skeptical as to whether you're truly achieving bit-perfect for 2 ch. PCM (CD audio) with this setup. Unless you're bypassing kmixer altogether, which requires using ASIO or WDM drivers, there will still be digital volume adjustment applied to all PCM audio.

It seems to me that achieveing bit-perfect with encoded material using onboard sound is nothing new, EXCEPT for 44.1 kHz DTS audio that was problematic because most older AC97 codecs can't output digital audio at 44.1 kHz (and resample to 48 kHz). Although you gain this possibility with the Realtek HD codecs (plus the autoswitching), I don't think you are outputting bit-perfect PCM. But if I'm wrong I'd love to be corrected, for indeed it would be great news.

Does anything happen if you reduce the volume below 100% in MCE while playing 44 kHZ DTS audio? If not, that means that DTS WAV is somehow recognized as encoded material and is being passed through unmolested.

However, I'm still skeptical as to whether you're truly achieving bit-perfect for 2 ch. PCM (CD audio) with this setup. Unless you're bypassing kmixer altogether, which requires using ASIO or WDM drivers, there will still be digital volume adjustment applied to all PCM audio.

It seems to me that achieveing bit-perfect with encoded material using onboard sound is nothing new, EXCEPT for 44.1 kHz DTS audio that was problematic because most older AC97 codecs can't output digital audio at 44.1 kHz (and resample to 48 kHz). Although you gain this possibility with the Realtek HD codecs (plus the autoswitching), I don't think you are outputting bit-perfect PCM. But if I'm wrong I'd love to be corrected, for indeed it would be great news.

I think you might be right!

The way I tested for kmixer is to re-enable sound effects under MCE (dings and such while navigating menu). If I played a 44khz DTS music, navigating the menu will cause the music to momentarily drop and then resume. I conclude this behaviour as bypassing the kmixer which results in only one sound source to be played through the audio device at any given time. Also, I tested your suggestion of changing the master audio volume. It had no impact to the music at all like you said.

Now, here is what really puzzled me. I played a 44khz stereo music. My receiver indicates a 44khz source which suggests the sound is not being upsampled to 48khz by Windows (kmixer?). I also changed the master audio volume and it had no effects on the music which again suggests no digital volume molestation by Windows (kmixer?). However, if I navigated through MCE menu while the music is playing, the dings and the music can both be heard at the same time without interruptions as if they were being mixed by Windows (kmixer?)

So, what exactly is happening here? No upsampling or volume molestation but definitely getting mixed. Can kmixer mix sounds but not upsample or control volume? Is it remotely possible that this Realtek HD codec is doing the mixing and still bypassing kmixer? I don't know. I'm confused.

I think you might be right!

The way I tested for kmixer is to re-enable sound effects under MCE (dings and such while navigating menu). If I played a 44khz DTS music, navigating the menu will cause the music to momentarily drop and then resume. I conclude this behaviour as bypassing the kmixer which results in only one sound source to be played through the audio device at any given time. Also, I tested your suggestion of changing the master audio volume. It had no impact to the music at all like you said.

Now, here is what really puzzled me. I played a 44khz stereo music. My receiver indicates a 44khz source which suggests the sound is not being upsampled to 48khz by Windows (kmixer?). I also changed the master audio volume and it had no effects on the music which again suggests no digital volume molestation by Windows (kmixer?). However, if I navigated through MCE menu while the music is playing, the dings and the music can both be heard at the same time without interruptions as if they were being mixed by Windows (kmixer?)

So, what exactly is happening here? No upsampling or volume molestation but definitely getting mixed. Can kmixer mix sounds but not upsample or control volume? Is it remotely possible that this Realtek HD codec is doing the mixing and still bypassing kmixer? I don't know. I'm confused.

Well it sure sounds like kmixer IS being bypassed so long as you're playing a single source -- either DTS wav or regular PCM -- because the Windows volume control has no effect.

kmixer can function happily at 44.1 kHz, it is because of the codec that audio gets resampled to 48 kHz. So yes, it can mix without resampling. However, it can't mix or even pass a single source without applying digital volume correction, so that even at 100% volume you're really getting something like 99.9% (ie very small attenuation).

What happens if you play PCM music, lower the volume volume below 100% and THEN add system sounds on top of it? Does the volume dip while the system sound is playing simulatneously? If so, that would mean that kmixer is being bypassed so long as there is a single source playing and no mixing is required, but it kicks in when mixing is required, causing a dropout in DTS wav playback, but no noticeable effect in PCM playback unless the volume is set much lower than 100%.

If that is the case, your findings would make sense and confirm that you ARE in fact getting bit-perfect PCM for music too!

I'm having some problems acchieving bit perfect playback on my PC.

I'm running the Audigy 2ZS, which, according to previous posts is actually capable of bit perfect playback.
Via MII12000 mini-itx mother board (VIA CLE266 chipset)

Here's what I've done so far:

Go to Start/Programs/Creative/Soundblaster Audigy 2 and run "Creative Audio Console". On the "SPDIF I/O" tab, choose "44.1 Khz". On the "Bit Accurate" tab, select "Enable Bit Accurate Playback".
Downloaded and installed the ASIO_out.dll. Set the card as my card and for Gapless Audio, I also increased the buffer to it's max to start with. I copied over the SSE version of the .dll since I've learned that my chip is equivilant to about a PIII.
Installed ASIO4ALL, and increased the buffers to their max to start with


Now, I'm having a problem. When I hit play on Winamp, nothing happens. It will act like it's going to play, but it just doesn't. Not quite sure how to describe it. It's like it can't find the file to play or something...

have I done something wrong? have I set-up something incorrectly?

I'd appreciate any help anyone can give!

Thanks,

Jan

Well it sure sounds like kmixer IS being bypassed so long as you're playing a single source -- either DTS wav or regular PCM -- because the Windows volume control has no effect.

kmixer can function happily at 44.1 kHz, it is because of the codec that audio gets resampled to 48 kHz. So yes, it can mix without resampling. However, it can't mix or even pass a single source without applying digital volume correction, so that even at 100% volume you're really getting something like 99.9% (ie very small attenuation).

What happens if you play PCM music, lower the volume volume below 100% and THEN add system sounds on top of it? Does the volume dip while the system sound is playing simulatneously? If so, that would mean that kmixer is being bypassed so long as there is a single source playing and no mixing is required, but it kicks in when mixing is required, causing a dropout in DTS wav playback, but no noticeable effect in PCM playback unless the volume is set much lower than 100%.

If that is the case, your findings would make sense and confirm that you ARE in fact getting bit-perfect PCM for music too!

Thanks for the explanation. I tested as you have outlined above. Here is the strange part. No matter where I set the master audio volume to, both 44khz stereo music and windows sounds don't change volume. I even tabbed outside of MCE to play some windows sounds and master volume has no effects at all.

What does this mean? This is beyond me as I don't know how kmixer, master volume control and Realtek HD codec all fit and work together.

Here are my observations so far inside MCE:

* 44khz stereo music and MCE sound effects - both sounds are mixed together perfectly with no drops
* 44khz DTS music and MCE sound effects - DTS will drop out momentarily while MCE sound is playing
* 48khz DD movie and MCE sound efffects - no dropouts on DD but MCE sound is never heard
* Master volume control has no effects on any of the above

Oh, too bad. I tried A387 5.10.0.6070 and R137 5.10.0.5910 and my menus look different than yours. There is no "Audio I/O" tab. Just "SPDIF In" and "SPDIF Out" and no auto lock option. Was it really 5.10.0.5354 that you have? I coudn't find it. It's probably my chip, not the drivers, but one can always hope...
Thanks,
Jason

I think our different chips use different drivers. I just looked at the driver download section on RealTek's site and found out I have the latest driver for my ALC880 chip. Another strange thing is if you look at the screenshot I attached a few posts above, the Windows title says MSI. But I promise I downloaded my driver directly from Realtek. So, somehow, Realtek's generic driver is detecting which mobo I have. I find that very odd!

Chung_chang,

Thanks for your post. I too am using the alc880 chipset sound although I am using a Gigabyte board instead of the MSI board you are using. I am having problem getting the results you are getting. I seem to only get PCM out no matter what I try to do. I am using the nVidia bronze decoder. I want to ask if you could help me troubleshoot my problem a little bit.

1: In the realtek setup software, I see the "Auto Lock" button you are refering to in the SPDIF setup but clicking it doesn't seem to do anything. I have the SPDIF set to digital out only. When you play an audio file or DVD, do you have the information filed in for the sampling rate, data validation and copyright protection. No information is being displayed for me.

2: I doubled checked that the nVidia decoder is set to SPDIF passthrough. You stated above that you are not able to change the volume in media center, I seem to be able to do so which leads to believe that something is not getting configured properly. So what exactly to you see in terms of volume control in MCE when audio is set to SPDIF.

3: Are there any other steps that you went through other than setting the realtek software to SPDIF digital out and setting the nVidia decoder to SPDIF.

Any info you can provide is highly appreciated.

Thanks

Chung_chang,

Thanks for your post. I too am using the alc880 chipset sound although I am using a Gigabyte board instead of the MSI board you are using. I am having problem getting the results you are getting. I seem to only get PCM out no matter what I try to do. I am using the nVidia bronze decoder. I want to ask if you could help me troubleshoot my problem a little bit.

1: In the realtek setup software, I see the "Auto Lock" button you are refering to in the SPDIF setup but clicking it doesn't seem to do anything. I have the SPDIF set to digital out only. When you play an audio file or DVD, do you have the information filed in for the sampling rate, data validation and copyright protection. No information is being displayed for me.

2: I doubled checked that the nVidia decoder is set to SPDIF passthrough. You stated above that you are not able to change the volume in media center, I seem to be able to do so which leads to believe that something is not getting configured properly. So what exactly to you see in terms of volume control in MCE when audio is set to SPDIF.

3: Are there any other steps that you went through other than setting the realtek software to SPDIF digital out and setting the nVidia decoder to SPDIF.

Any info you can provide is highly appreciated.

Thanks

I'm using basic nVidia PureVideo and MCE rollup 2.

1. Sometimes I see info, sometimes not. I don't think it's reliable. In the very beginning, I had to push "Auto Lock" while a DTS source was playing. Then, everything started working after that even after reboots.

2. Correct. Volume control in MCE (or Windows) has no effects at all with any sounds for me. The volume indicator goes up and down but has no effects. I strictly use receiver to control volume.

3. Nope. It's a very basic load of MCE and drivers. Other than hitting that "Auto Lock" button once in the beginning, nothing else special that I can think of.

I assume you are using the latest RealTek driver (R1.37). Does your RealTek audio Window title say GigaByte (like mine says MSI)? No sure what else I can do to help you. Just let me know. Good luck.

Chung_Chang,

Thanks for your response. I do see the Gigabyte logo in the realtek software. I also loaded in the latest release of their driver.

I'll keep playing with it. I sure I'll get it to work at some point. I'll post back with any success.

Thanks,

Chung_Chang,

Thanks for your response. I do see the Gigabyte logo in the realtek software. I also loaded in the latest release of their driver.

I'll keep playing with it. I sure I'll get it to work at some point. I'll post back with any success.

Thanks,

Wow! RealTek driver not only detects what mobo the driver is installed on, it also has the company logo (Gigabyte, MSI) to go with it? I don't think the company logo is stored anywhere on the mobo if no board specific drivers have been installed. So, where did Realtek driver pull the MSI or Gigabyte graphical logo from? Pre-loaded as part of the driver?

I hope that's the only customization Realtek has done to their driver, and nothing else that would impede your ability to do bit-perfect over spdif.

How can I enable an SPDIF option in MCE? I only have Stero and SAP.

Anyone using M-Audio Transit device successfully to playback DTS Cd's via Windows Media Player 10, chime in: I cannot.

I have recently purchased M-Audio Transit . . . installed per M-audo directions (drivers first, then reboot, then conntect USB device). I have enabled the device in Media Player . . . Tools . . . Options . . . Devices . . . Speakers). I have closed and restarted media player. I have rebooted.

I can play regular audio through the Transit, but not any test tracks of DTS tracks. I do not hear static. I do not hear anything.

Media player volume is set to max. Setting in Transit are 24-bit 8000 Hz to 48000 Hz. I have tried 16-bit 8000 to 48000 Hz. I have tried changing the latency. Nothing works. I can play regular CDs or WMA lossless, but not DTS cds. My external reciever does have built-in DTS decoders, I can play movies with DTS soundtracks no problem . . . if I change the transit settting to passthrough or if I use another sound device.

Suggestions? Otherwise, I say this solution for bit-perfect with WMP 10 only works with some systems.

OS: Windows MCE 2005 SP2
CPU: AMD FX-60 dual core
2GB Corsair RAM

Anyone using M-Audio Transit device successfully to playback DTS Cd's via Windows Media Player 10, chime in: I cannot.

I have recently purchased M-Audio Transit . . . installed per M-audo directions (drivers first, then reboot, then conntect USB device). I have enabled the device in Media Player . . . Tools . . . Options . . . Devices . . . Speakers). I have closed and restarted media player. I have rebooted.

I can play regular audio through the Transit, but not any test tracks of DTS tracks. I do not hear static. I do not hear anything.

Media player volume is set to max. Setting in Transit are 24-bit 8000 Hz to 48000 Hz. I have tried 16-bit 8000 to 48000 Hz. I have tried changing the latency. Nothing works. I can play regular CDs or WMA lossless, but not DTS cds. My external reciever does have built-in DTS decoders, I can play movies with DTS soundtracks no problem . . . if I change the transit settting to passthrough or if I use another sound device.

Suggestions? Otherwise, I say this solution for bit-perfect with WMP 10 only works with some systems.

OS: Windows MCE 2005 SP2
CPU: AMD FX-60 dual core
2GB Corsair RAM

I use WMP 10 with DTS CDs' with M Audio Transit successfully. Try going into Tools -> Options -> Device tab -> then highlight Speakers and click on Properties. Make sure the audio device to use list the DirectSound M-Audio Transit USB. Then restart your computer and see if that works. Hope that helps.
.

I use WMP 10 with DTS CDs' with M Audio Transit successfully. Try going into Tools -> Options -> Device tab -> then highlight Speakers and click on Properties. Make sure the audio device to use list the DirectSound M-Audio Transit USB. Then restart your computer and see if that works. Hope that helps.
.

Yeah, I already did this. It doesn't work for my DTS CD's or ripped DTS audio tracks. I have enabled the device in Windows Media Player as you describe, and as I described in my post, but no dice. I can play other audio tracks via the Transit device on Media Player 10, but not DTS tracks. The funny thing is that others describe hearing static or noise when they do this, and the pathway is not bit perfect . . . I hear nothing at all . . . no static . . . nothing.

Thanks for the explanation. I tested as you have outlined above. Here is the strange part. No matter where I set the master audio volume to, both 44khz stereo music and windows sounds don't change volume. I even tabbed outside of MCE to play some windows sounds and master volume has no effects at all.

What does this mean? This is beyond me as I don't know how kmixer, master volume control and Realtek HD codec all fit and work together.

Here are my observations so far inside MCE:

* 44khz stereo music and MCE sound effects - both sounds are mixed together perfectly with no drops
* 44khz DTS music and MCE sound effects - DTS will drop out momentarily while MCE sound is playing
* 48khz DD movie and MCE sound efffects - no dropouts on DD but MCE sound is never heard
* Master volume control has no effects on any of the above

Can't explain that either. The Realtek probably has a hardware mixer that is used when bit-perfect is selected in the control panel, so kmixer is still being bypassed. Regardless, you are still getting bit-perfect, even for PCM audio, that's the important part.

Another thing you can do to test - go into Windows Media Player and play one of the DTS surround files from this website -

http://www.sr.se/multikanal/english/e_index.stm

I used the "Multichannelcheck" file from the bottom of the page.

If you have bit-perfect playback in Windows, you will hear the file being played. If not - silence.

DE

I will test that DTS out tonight. I already have a feeling that it'll play fine. I'll let you know.

Does playing DTS bit-perfect also prove that stereo is bit-perfect?

How can I prove that I'm really bypassing kmixer besides the volume control test?

Most of my music are in stereo so I'm more interested in bit-perfect stereo 44.1 khz.

Can I rename/delete kmixer.dll (if there is such a dll) to prove it? Or will that break Windows? Any other ways to prove this (for stereo)?

You've already tested 44 kHz DTS wav files, no need to do so again I would think.

In your case I think it is pretty obvious that both DTS and PCM are being played bit-perfect given the similar playback behaviour and the lack of volume control in both cases. Unless you are using ASIO or WDM drivers, I think the lack of volume control is the best sign you have that kmixer is bypassed. Unless anyone has another idea?

Ultimately the proof is in what you hear, but I'd consider it settled.

Any new suggestions? I don't think the M-Audio Transit USB device is a bit-perfect solution for everyone . . . it doesn't work for me.

ok. I went ahead and tested those files from the website. I believe I have already tested that exact same DTS file earlier in this thread. I also went ahead and tested the DD file. Is the DD file supposed to be 44.1 khz? That's what showed up on my receiver. I think that was the first time I've ever played a DD sound at 44.1 khz. And yes, both DTS and DD files worked as I have expected.

crabnebula, I too will consider my setup bit-perfect unless someone else has other ideas. I'm all ears and willing to test. I did try to delete kmixer.sys under windows\system32\drivers. But the file just reappears automagically in 1 second so I was not able to test that.

xAragornx, have you any luck with ALC880 chip on your Gigabyte board? I'd hate to think there are different implementations of the same chip on different mobos causing different spdif behaviors.

Chung_Chang,

I'm still working on it. I'll post back with any headway. It's thanks to your success that I'm plowing ahead. I'm sure I'll get it working at some point.

Can anyone help on this topic: http://www.avsforum.com/avs-vb/showthread.php?t=678413

Briefly, USB sound card stop sending AC-3/DTS digital signal through SPDIF. It worked before.

Thanks.
Roy

Chung_Chang,

I'm still working on it. I'll post back with any headway. It's thanks to your success that I'm plowing ahead. I'm sure I'll get it working at some point.

Try this:

1. Play a 44.1 khz DTS wave using Media Player
2. Open up Realtek audio control panel, go to Audio I/O and Digital
3. Select 44.1 khz and press Auto-Lock (while wave is playing)
4. Apply changes

I don't remember if that's what I did as I was messing with the settings quite a bit but see if that works for you.

ok. I went ahead and tested those files from the website. I believe I have already tested that exact same DTS file earlier in this thread. I also went ahead and tested the DD file. Is the DD file supposed to be 44.1 khz? That's what showed up on my receiver. I think that was the first time I've ever played a DD sound at 44.1 khz. And yes, both DTS and DD files worked as I have expected.


I tried the files from the above link (both dd and dts) and get nothing but noise. DVD's with DD are played properly as are WMV 1080P clips from Microsoft (http://www.microsoft.com/windows/windowsmedia/musicandvideo/hdvideo/contentshowcase.aspx#sysreq), MP3's and standard CD's. The appropriate indicator: Dolby Digital or Linear PCM lights on my surround sound receiver. I don't have any DTS to try, but I suspect it will work also if it's on a DVD or video clip. I can play the video and music files in both MCE and XP.

Basically, DVD's and video clips work fine with surround, but audio clips from http://www.sr.se/multikanal/english/e_index.stm don't. Is there something I have to do to associate .wav files with a particular codec?

Another strange symptom, although surround is working, the surround sound demos do not work within the Realtec Audio Manager. Might be that it is also useing .wav. Although theAUTOLOCK seems to be working, it always says "Unlocked" and the selected bit-rate does not change on the display.

Current codec is Nvidia purevideo. Mobo is the MSI K8NGM2-FID 6150. Receiver is an old JVC RX-9000v. Any ideas?

"DTS" lights up on my receiver when playing a DTS movie, but then when i try the multi-channel audio test file (mentioned earlier) within WMP 10 it fails (ping noise). I am not bit-perfect like i thought i was, correct? Is it even possible with my set up?

Set up:
Turtle Beach Montego DDL Sound card (With spidif enabled connected through a optical cable to my Sony receiver)

nvidia purevideo codec (latest ver)
ECS nFORCE-3 mobo
MCE 2005

Is the dream possible......

Media player volume is set to max. Setting in Transit are 24-bit 8000 Hz to 48000 Hz. I have tried 16-bit 8000 to 48000 Hz. I have tried changing the latency. Nothing works. I can play regular CDs or WMA lossless, but not DTS cds. My external reciever does have built-in DTS decoders, I can play movies with DTS soundtracks no problem . . . if I change the transit settting to passthrough or if I use another sound device.

Suggestions? Otherwise, I say this solution for bit-perfect with WMP 10 only works with some systems.

OS: Windows MCE 2005 SP2
CPU: AMD FX-60 dual core
2GB Corsair RAM

To try and eliminate windows as the culprit:

Set your M-Audio control panel to the topmost setting in the menu (2-in, 2-out, 16-bit, 8000 to 48000), latency medium, mic boost 0dB (off).

Obviously, under WMP10 preferences, select the M-Audio as default sound device. Set it under Sound Control Panel as well.

Make sure your volume sliders in main and WMP10 are pegged at the top.

Also, what driver versions for ASIO and USB are reported on this M-Audio screen for you?

Now, what kind of receiver are you sending your HTPC optical digital signal to? Make/Model? It's possible that the DTS wav file info played over your USB connection is buried in a PCM stream that is not immediately recognized by your receiver because it does not necessarily conform to the DTS digital spec. Do you have access to another receiver to test with?

Sorry to hear of your disappointment, hope we can help.

Best,
Adam

"DTS" lights up on my receiver when playing a DTS movie, but then when i try the multi-channel audio test file (mentioned earlier) within WMP 10 it fails (ping noise). I am not bit-perfect like i thought i was, correct? Is it even possible with my set up?

Set up:
Turtle Beach Montego DDL Sound card (With spidif enabled connected through a optical cable to my Sony receiver)

Is the dream possible......

As far as I know, no one has successfully tested your Turtle Beach sound card to produce bit-transparent / bit-perfect audio output from all sources including DTS audio.

I and others here continue to have success with the Transit USB, and Chung Chang and others are experimenting with HD-Audio capable onboard sound on some new motherboards, though I'm concerned by Sleep Doc's difficulties with the Transit and want to get to the bottom of his issues. I can say that the Transit USB has worked flawlessly for me and still does.

Good luck,
Adam

I'm having some problems acchieving bit perfect playback on my PC.

I'm running the Audigy 2ZS, which, according to previous posts is actually capable of bit perfect playback.

Now, I'm having a problem. When I hit play on Winamp, nothing happens. It will act like it's going to play, but it just doesn't. Not quite sure how to describe it. It's like it can't find the file to play or something...


In my experience, Creative cards are not capable of bit-perfect output out of the box at this time, but I could be wrong. If it's solely bit-perfect audio out of Winamp ASIO you're after, (or JRiver Media Center or Foobar which works like a charm as well), then I'd suggest picking up a very inexpensive Chaintech AV710 card, which I believe is about $27 last I checked. Should work well.

Good luck,
Adam

BTW, there is a new full-size MSI motherboard for the AM2 platform which carries the 883 RealTek on-board sound and has 3 pci slots. I was all set to go with the previous MSI board discussed here but I need to look into the new AM2 motherboard and all the ramifications that come with it.

Aaron

Here's an update on my quest for bit-perfect audio using the ALC880 onboard sound.

I'm very close to having my setup working. Here is where I am right now.

I have the DTS or Dolby digital light coming up on my receiver when I play movies in WMP but no such luck when playing back the same files in media center. I did at one point have the lights coming up when playing the files in media center but everything went back to PCM audio when I rebooted the system. Now I am not able to get DTS or Dolby coming out of media center again no matter what I try in terms of settings.

I have tried many different settings in the realtek driver and have double and triple check that the nVidia decoder is set to SPDIF. Unfortunately, I can't exactly point to what driver settings got the bit perfect audio to work in WMP. I'm trying to back track to find out what setting made it work. I will post back when I find it out. I think that maxing out all the volume levels made it work but I'm still verifying that.

As far as MCE, it seems like it is configured properly. Changing the volume in MCE doesn't change the volume of the SPDIF output. I have gone thru the speaker configuration setup and set the output to SPDIF. I running out of options to try in MCE. I'm so close, I just need to find the "magic" setting that makes it work. Like I said I had it working at one point but it reverted back to PCM when I rebooted.

So close yet so far. Hopefully someone else has had the same issue and found the solution. Any advice is greatly appreciated.

Meanwhile I'll keep chipping away. I'll post back with updates.

Thanks all!

Here's an update on my quest for bit-perfect audio using the ALC880 onboard sound.

I'm very close to having my setup working. Here is where I am right now.

I have the DTS or Dolby digital light coming up on my receiver when I play movies in WMP but no such luck when playing back the same files in media center. I did at one point have the lights coming up when playing the files in media center but everything went back to PCM audio when I rebooted the system. Now I am not able to get DTS or Dolby coming out of media center again no matter what I try in terms of settings.

I have tried many different settings in the realtek driver and have double and triple check that the nVidia decoder is set to SPDIF. Unfortunately, I can't exactly point to what driver settings got the bit perfect audio to work in WMP. I'm trying to back track to find out what setting made it work. I will post back when I find it out. I think that maxing out all the volume levels made it work but I'm still verifying that.

As far as MCE, it seems like it is configured properly. Changing the volume in MCE doesn't change the volume of the SPDIF output. I have gone thru the speaker configuration setup and set the output to SPDIF. I running out of options to try in MCE. I'm so close, I just need to find the "magic" setting that makes it work. Like I said I had it working at one point but it reverted back to PCM when I rebooted.

So close yet so far. Hopefully someone else has had the same issue and found the solution. Any advice is greatly appreciated.

Meanwhile I'll keep chipping away. I'll post back with updates.

Thanks all!


Keep it up!

I'll need to reinstall MCE soon to hopefully resolve my Avermedia A180 not being able to receive 2 OTA HD channels. When I do, I'll be sure to take note of all the settings I change (and hopefully still able to get bit-perfect after re-install).

I read about 2/3 of this thread very excidedly, hoping to find a plausible solution, but I guess I missed the post explaining what I needed to buy to make this work. Bit-perfect audio nirvana from MCE has always been a wet dream of mine since i have over 50gb of music that I personally ripped to APE and FLAC from my extensive CD collection. All that ripping effort and no MCE bit-perfect playback to show for it -- what a waste! So sadly, it's remained only a dream of mine on my current MCE setup (Asus A8N-SLI Premium mobo, M-Audio Audiophile 24/96, nForce4 onboard sound (not used much), MCE'05 with RU2, Integra DTR-5.6 A/V receiver). In fact, I can't even get my receiver to detect a DD or DTS signal from the SPDIF! :(

I now have a chance to get it right: I'm gonna build a dedicated HTPC in the next week or so, and having bit-perfect PCM, DD, and DTS audio is a top priority. However, I will not purchase a motherboard with integrated graphics since they simply don't allow for enough future expansion. Right now my list of components includes:

SilverStone LC16B case (http://www.newegg.com/Product/Product.asp?item=N82E16811163053)
Abit KN8-SLI mobo (nForce4) (http://www.newegg.com/Product/Product.asp?item=N82E16813127225) -- I chose this because it was one of the cheapest mobos with a passive chipset cooling solution. Those chipset fans can be LOUD!
XFX 7600GS (silent) (http://www.newegg.com/Product/Product.asp?item=N82E16814150146)
...and some more stuff that doesn't need to be included in this post.

What must I buy to ensure I can get bit-perfect PCM/DD/DTS from 44k and 48k sampling rates without adjusting anything? I have no problem spending the money on something to make this work -- I just want to make damn sure it works. :)

I would be eternally grateful for any help provided for either MCE setup. Thanks in advance.

What must I buy to ensure I can get bit-perfect PCM/DD/DTS from 44k and 48k sampling rates without adjusting anything? I have no problem spending the money on something to make this work -- I just want to make damn sure it works. :)

I would be eternally grateful for any help provided for either MCE setup. Thanks in advance.

Hope we can help. As far as I know save for one errant report, everyone that has used the M-Audio Transit USB in the configuration I described in my post above (http://www.avsforum.com/avs-vb/showthread.php?p=7200938&&#post7200938) has successfully achieved bit-perfect playback from within MCE from all sources, including DTS wav files and DTS audio CD's. No other sound cards I know of can achieve this save for the RME card and possibly some HD audio onboard sound on newer mobos.

The trick is to either use your onboard sound or another sound source (in my case, the MyHD card) for DD/DTS audio, or if you use the M-Audio for everything, you'll have to switch settings within the M-Audio control panel each time you want to switch between 44kHz audio and 48kHz audio (DD/DTS). This is actually not such a hard thing to do, though it does require mouse intervention in an otherwise remote-controlled environment.

If price is no object, you could explore the use of an extremely high-end RME sound card, but I'd call them first to verify MCE compatibility. This card apparently auto-switches between 44 and 48kHz in its own driver and bypasses windows kmixer.sys. I have not tested this.

Good luck!

Regards,
Adam

Thanks for your help -- let me see if I've got it straight: so if I used my onboard nForce4 SPDIF for DD/DTS (48k & 96k) and the Transit for PCM (44k) I wouldn't have to adjust any settings when, say, moving from "My Music" to "Play a DVD" while using the MCE interface? But if I used the Transit for everything, I would have to minimize MCE and change a setting in between? And if I didn't change that setting, would I not hear anything or would the sound just not be bit-perfect?

Conversely, if I bought a mobo with the new onboard Realtek 880 HD audio codec would I still have to minimize MCE and use a mouse to change settings when moving from PCM to DD/DTS? It's pretty ridiculous how complicated and confusing this whole thing is; I consider myself a pretty advanced user and this is still kinda hard for me to follow. I'm assuming Microsoft is to blame for it all...

Anyone have any information on how Vista changes all of this? I hear they're getting rid of KMangler.

Apparently Vista will completely change everything; audio implementation is supposed to be very different and significantly improved. From what I've heard, you can adjust the audio settings and default volume level for every individual program you run (if you so choose). But waiting for Vista, which is at least another 7+ months away, is not an option I'd recommend since pretty much all the hardware you can buy right now (with the exception of video cards) will work just fine with Vista.

Anyone have any information on how Vista changes all of this? I hear they're getting rid of KMangler.

That's a thread here (http://www.avsforum.com/avs-vb/showthread.php?t=649421).

-Adam

Anyone have any information on how Vista changes all of this? I hear they're getting rid of KMangler.


Vista will have a exclusive mode for audio. Under this mode, the audio stream will not be mixed by the global audio engine and will be rendered directly to the audio hardware.

In case anyone is interested, I did find another RealTek 880 motherboard with no on-board graphics and more pci slots at NewEgg:

AOpen nCK804Ua-LFS Socket 939 NVIDIA nForce4 Ultra ATX AMD Motherboard

But it has some other faults. So, I'm still looking around.

Aaron
I just checked AOpen's website and it appears the newegg listing is a typo. AOpen's site (http://usa.aopen.com/products/mb/nCK804Ua-LFS.htm) clearly states the onboard audio solution is the "Realtek AC'97 CODEC on-board (ALC850)". Oh well.

Conversely, if I bought a mobo with the new onboard Realtek 880 HD audio codec would I still have to minimize MCE and use a mouse to change settings when moving from PCM to DD/DTS? It's pretty ridiculous how complicated and confusing this whole thing is; I consider myself a pretty advanced user and this is still kinda hard for me to follow. I'm assuming Microsoft is to blame for it all...

I have MSI with onboard Realtek 880 HD codec. It has onboard video but also has a PCI-E x16 for another video card. I don't have to change anything to listen to any sounds. And supposedly, it is bit-perfect for everything unless someone can prove otherwise for 2 channel PCM 44.1khz music. Read my posts in this thread for more details.

I have MSI with onboard Realtek 880 HD codec. It has onboard video but also has a PCI-E x16 for another video card. I don't have to change anything to listen to any sounds. And supposedly, it is bit-perfect for everything unless someone can prove otherwise for 2 channel PCM 44.1khz music. Read my posts in this thread for more details.
Wow, that board has a PCI-e x16 slot and two PCI slots! I wrote integrated gfx boards off because I was under the (apparently false) impression that they usually only have 1-2 expansion slots. Looks like I will be purchasing the exact same board as you along with the 7600GS I linked above and a Hauppauge PVR-500MCE. Wish me luck!

Now, if only it were possible to listen to my collection of lossless music either from within MCE or another program with a large enough display interface to be legible & navigable from the couch...

Wow, that board has a PCI-e x16 slot and two PCI slots! I wrote integrated gfx boards off because I was under the (apparently false) impression that they usually only have 1-2 expansion slots. Looks like I will be purchasing the exact same board as you along with the 7600GS I linked above and a Hauppauge PVR-500MCE. Wish me luck!

Please post your bit-perfect results once you have everything setup. As far as I'm aware, mine is the only bit-perfect setup using the onboard Realtek 880 codec under MCE. I'll be reinstalling MCE soon to make sure my current setup is not a fluke. You can try without the 7600GS to see if onboard video is sufficient. Mine does 720/1080 HD flawlessly and I'm using a 3000+ Venice.

Please post your bit-perfect results once you have everything setup. As far as I'm aware, mine is the only bit-perfect setup using the onboard Realtek 880 codec under MCE. I'll be reinstalling MCE soon to make sure my current setup is not a fluke. You can try without the 7600GS to see if onboard video is sufficient. Mine does 720/1080 HD flawlessly and I'm using a 3000+ Venice.
Thanks again for your help! Do you think it's possible to hold off on reformatting until after my board arrives next week? I'm gonna try to duplicate your setup exactly, and I'd really appreciate some help with getting the settings to match yours exactly.

I will be installing only the bare minimum components to insure nothing screws up my bit-perfection: A64 3200+ CPU (venice), 1gb Crucial RAM, NEC DVD drive, an SATA 300gb HDD, and using onboard video. As per your instructions, I will be installing MCE'05 OEM & RU2 before trying to tweak the settings. Only after I have verified it to be BP will I go forward with the discrete video card (I'm gonna be running some 1080p video which is beyond the capabilities of the 6150), TV tuner, etc.

*crosses fingers*

Thanks again for your help! Do you think it's possible to hold off on reformatting until after my board arrives next week? I'm gonna try to duplicate your setup exactly, and I'd really appreciate some help with getting the settings to match yours exactly.

I will be installing only the bare minimum components to insure nothing screws up my bit-perfection: A64 3200+ CPU (venice), 1gb Crucial RAM, NEC DVD drive, an SATA 300gb HDD, and using onboard video. As per your instructions, I will be installing MCE'05 OEM & RU2 before trying to tweak the settings. Only after I have verified it to be BP will I go forward with the discrete video card (I'm gonna be running some 1080p video which is beyond the capabilities of the 6150), TV tuner, etc.

*crosses fingers*

No problems. Main reason for reformat is to try getting my Avermedia A180 card working again. I lost OTA CBS and ABC DTV about 2 weeks ago and could never get them back again (ATSC tuner in my TV works fine with the same antenna). I'm beginning to wonder if my A180 is defective as it never really worked 100% on my previous PC either (lost WB). I still have plenty of DVD movies and music I can enjoy so I can wait on reinstall. Good luck to you and let us know how it goes.

My requirements are:

1. By-passes Kmixer.
2. Bit perfect PCM (no resampling of 44.1kHz to 48kHz).
3. No manual switching between 44.1 and 48.
4. DD/DTS pass-through.
5. Doesn't use ASIO or KS (so that it is compatible in MCE and any other app).

So, as if understand correctly, the only PC audio solution that does the above is the onboard ALC88x. No non-motherboard sound cards can do all of the above?

I have not managed automatic switching of the digital output between 44.1 and 48khz with my onboard realtek sound solution. That is a checkbox in the control panel for me.

Cheers

Thomas

My requirements are:

1. By-passes Kmixer.
2. Bit perfect PCM (no resampling of 44.1kHz to 48kHz).
3. No manual switching between 44.1 and 48.
4. DD/DTS pass-through.
5. Doesn't use ASIO or KS (so that it is compatible in MCE and any other app).

So, as if understand correctly, the only PC audio solution that does the above is the onboard ALC88x. No non-motherboard sound cards can do all of the above?

I believe my current setup fulfills all of your requirements. However, I'm waiting for bling24 to finish up his near identical setup sometime this week to confirm that he can also duplicate my bit-perfect setup before I'd consider it a sure bet.

As for standalone cards, I believe someone in this thread mentioned an RME card that might work but I think it's expensive.

I believe my current setup fulfills all of your requirements.
As for standalone cards, I believe someone in this thread mentioned an RME card that might work but I think it's expensive.

I think RME cards have a problem passing AC3/DTS over SPDIF. I think you can get bitperfect PCM but whats the point if you can't get AC3/DTS movies to pass. Something about directshow vs wavout. RME wants one, MCE the other :(

If i am not all that concerned with my music being "bit perfect" but i want all my DVD's (DD & DTS) to sound as good as possible, is the M-Audio Transit USB something i even need to consider? With my current setup (Turtle Beach Sound Card w/optical out to a Sony receiver) the DTS badge lights up when playing a DVD with a DTS track, am i as good as i am going to get? Thanks in advance for any opinions.

Thanks again for your help! Do you think it's possible to hold off on reformatting until after my board arrives next week? I'm gonna try to duplicate your setup exactly, and I'd really appreciate some help with getting the settings to match yours exactly.

I will be installing only the bare minimum components to insure nothing screws up my bit-perfection: A64 3200+ CPU (venice), 1gb Crucial RAM, NEC DVD drive, an SATA 300gb HDD, and using onboard video. As per your instructions, I will be installing MCE'05 OEM & RU2 before trying to tweak the settings. Only after I have verified it to be BP will I go forward with the discrete video card (I'm gonna be running some 1080p video which is beyond the capabilities of the 6150), TV tuner, etc.

*crosses fingers*

How did your install go?

What mobo are you guys using?

Sorry I couldn't jump in here sooner guys -- a tree fell in my back yard on Saturday morning and took my cable line with it! That meant no TV or internet for me until probably later this afternoon when the cable company can come out to reattach the cable to the telephone pole...

Anyway, I finally got a chance to build the system last night using only vital components (aka no discrete gfx card, no TV card -- just the MSI board with onboard video, CPU, RAM, HDD, and DVD drive). I installed MCE'05, but I don't think my OEM CDs have RU2 included on them. Luckily I had a copy of the RU2 patch saved on my desktop, but alas it will not install on my HTPC because it requires the latest .NET framework and the latest patch for .NET -- none of which is obtainable since my internet is down!!

Regardless, I hooked up the MSI's SPDIF out bracket and installed the Realtek HD audio driver from the included MSI CD since I don't have internet access to get the stand-alone Realtek one. In the driver settings I noticed that I have two extra options in the tab on the screenshot you posted that you do not -- one to select the SPDIF's output and another to disable it. Other than that, I matched your settings exactly. I wish I could post a screenshot for you, but unfortunately that will have to wait until my internet returns.

I tried playing some PCM (mp3), and it was detected as such by the receiver. Adjusting the Windows volume using the tray speaker icon had no effect on the sound level, but using the volume slider in WMP or in Winamp (v5.23) did change the volume. I then tried some lossless PCM (.ape) with the same results. I also tried some DD test files (.vob) and got sound from them, but the receiver still said it was being fed PCM and not DD; I then got no sound at all from the DTS .vob I used. Thus I don't believe I am getting bit-perfect audio yet. :(

By the way, should I be using WMP 10 to test these files or something else like the MCE front-end? In programs like Winamp and Media Player Classic I get to select the output device (ex: WaveOut, DirectSound, etc) -- which should I be using?

Hopefully I can get my internet restored at home so I don't have to resort to using my lunch break at work to post. Maybe my problem is not having RU2, or maybe it's that I used the MSI-bundled Realtek driver instead of a stand-alone one? Here's hoping that I can get my internet restored at home so I can properly update everything...

MCE and WMP use the same core library, they are just different fron ends for the same guts.

You should test with what you want to use, if that's MCE then WMP is a fine substitution.

Sorry to hear about the fallen tree. Hope you get things in order soon.

I captured those screenshots via remote desktop. If I physically went to the media PC, then my screen has the 2 extra options you mentioned.

Try playing that DTS file again except this time, while it's playing, go into the driver and select 44.1khz and then click auto-lock. See if that works.

Unfortunately, it looks like I they won't be restoring my internet or TV until next week (ugh!), but I do have good news!

I went back and uninstalled the MSI drivers and did an *extensive* search for the v5253 Realtek drivers that worked for you and finally found a working link on a Russian website. :)

Try playing that DTS file again except this time, while it's playing, go into the driver and select 44.1khz and then click auto-lock. See if that works.
When I double clicked the DTS .wav file all I heard was clicking and buzzing (same with the DD file), but then I tried what you described above and IT WORKED!! My receiver immediately clicked into DTS mode and decoded it perfectly; same with DD and PCM! You have no idea how happy I am right now...thank you very much for your help!

To show my appreciation, I have decided to host the v5253 drivers for posterity on some webspace I have at http://tbaudio.com/realtek/ . I will leave the zip file up for quite awhile in case anyone else here chooses to go this route and needs working drivers.

Thanks again to CC for keeping this thread alive and helping others out. I'd also like to say thank you to my neighbor for not passwording his wireless network ;)

Congratulations! I feel much better about reinstalling now too.

I didn't even realize Realtek had already updated the driver to R1.39. I'll give the latest driver a go when I reinstall. Worse case is I revert back to the old driver which I have at home.

I think this pretty much proves bit-perfect sound works perfectly under MCE 2005 using the MSI 6150 board. I'd like to hear other boards working using the Realtek HD codec.

I am also looking to do the same, and have been down the M-Audio Transit path. While it is possible to get bit perfect playback of wma loseless (my amp even shows 44.1 khz) it's not possible to get DD/DTS passthrough without changing a setting in the driver. Trouble is, when you change that setting it works fine for DD/DTS, but does not play back any other non DD/DTS audio :mad:


I’ve experienced the exact same thing, after jumping through all the fiery hoops and getting DTS playback from my hard drive I could not play normal 2-ch .ape encoded or .wav files. I would have to go back into the M-Audio console and reset the ASIO driver back to consumer and remove the “Non-Audio” checkmark then restart MC11 and make changes in there as well (back to Direct).

Just isn’t worth it, somebody should make a turnkey solution for this, I know it would sell well. I even tried the new DTS card by Azuntech and it sucked at DTS encoding, my Halo Natural mode does a much cleaner and far superior sounding 5-ch matrix.

Try playing that DTS file again except this time, while it's playing, go into the driver and select 44.1khz and then click auto-lock. See if that works.


When I double clicked the DTS .wav file all I heard was clicking and buzzing (same with the DD file), but then I tried what you described above and IT WORKED!! My receiver immediately clicked into DTS mode and decoded it perfectly; same with DD and PCM! You have no idea how happy I am right now...thank you very much for your help!

I must be blind. I just uninstalled my driver and tried the package from your WEB site. Where is the magic auto-lock button? Could you please post a screenshot?

Cheers

Thomas

chung_chang posted a screenshot on page 4 -- here's the link to his post (http://www.avsforum.com/avs-vb/showthread.php?p=7688413&&#post7688413)

I have a screenshot attached on page 4 of this thread.

Holy! Bling24, we posted at the same exact minute!

Congratulations!

I think this pretty much proves bit-perfect sound works perfectly under MCE 2005 using the MSI 6150 board. I'd like to hear other boards working using the Realtek HD codec.

Note that chung_chang has written a great writeup on installing and configuring the MSI 6150 board which I have added to my original post (http://www.avsforum.com/avs-vb/showthread.php?p=7200938&&#post7200938) above for those arriving from the MCE 2005 FAQ and elsewhere.

Bravo!

Thomaspf, please let us know your test results as well.

Best,
Adam

Great idea Ad-Rok -- that should save alot of time for many first-time viewers of the thread. I always hate having to dig through pages and pages of trials & errors to find the one or two posts which uncover a solution to the OP's problem; I wish more OPs would edit their first posts to include solutions as you have done.

*edit* oops, just realized you aren't in fact the OP. Hopefully the link from the MCE'05 FAQ points to your post and not post #1. Also, you might add that we've only confirmed bit-perfection with the specific Realtek drivers I hosted above, not the driver from the MSI CD.

Holy! Bling24, we posted at the same exact minute!
Hah, what are the odds?

Hopefully the link from the MCE'05 FAQ points to your post and not post #1.

Yup, it sure does. I had PM'ed the MCE 2005 FAQ poster a while ago, and he direct-linked to my post above (#22). My goal was to help others achieve bit-perfection since I was so frustrated trying to find reliable information earlier. Hopefully it will!


Also, you might add that we've only confirmed bit-perfection with the specific Realtek drivers I hosted above, not the driver from the MSI CD.

Done. Thanks for the heads-up, and for hosting the known working drivers.

Best,
Adam

Done. Thanks for the heads-up, and for hosting the known working drivers.

Best,
Adam

Will this hosted driver work on any sound card, I cant find a good ASIO driver for my M-Audio Delta 2496 card as was thinking about using this one posted here. ASIO4All wont work at all on this card and the factory driver is really poor on and pauses and skips no matter how much buffer I add.

Will this hosted driver work on any sound card, I cant find a good ASIO driver for my M-Audio Delta 2496 card as was thinking about using this one posted here. ASIO4All wont work at all on this card and the factory driver is really poor on and pauses and skips no matter how much buffer I add.

Do you mean to ask if a Realtek sound card driver would work for an M-Audio card? I think the answer to that is certainly not, but if I'm missing something, ask Chung Chang or bling24 for more info.

Good luck,
Adam

Do you mean to ask if a Realtek sound card driver would work for an M-Audio card? I think the answer to that is certainly not, but if I'm missing something, ask Chung Chang or bling24 for more info.

Good luck,
Adam

Sorry, never heard of a realtek sound card unless its one of those built into the motherboard ones that are far from audiophile quality and would seem silly to worry about bit perfect output anyway.

I've been trying to do this bit perfect thing for two years and no luck as of yet on many different audiophile cards.

I've been trying to do this bit perfect thing for two years and no luck as of yet on many different audiophile cards.

In our research from efforts in this thread we have found that you cannot and will not achieve bit-perfect output in MCE 2005 unless you use exactly one of the four suggested solutions in my post here (http://www.avsforum.com/avs-vb/showthread.php?p=7200938&&#post7200938). In other words, your current card won't work with MCE 2005.

Not sure what your criteria for an audiophile sound card solution is. Realtek powers the HD Audio solutions in the MSI motherboard, and HD Audio is quite advanced and largely superior to the AC '97 audio specification it replaces, despite being integrated into the motherboard.

If you just want bit-perfect ASIO or kernel streaming without using MCE 2005, then that's not the focus of this thread.

Good luck,
-Adam

Sorry, never heard of a realtek sound card unless its one of those built into the motherboard ones that are far from audiophile quality and would seem silly to worry about bit perfect output anyway.

All we're using the onboard Realtek chip for is passing a digital bitstream untouched and not re-sampled to an external DAC. Thus, debating the quality of the DACs inside the Realtek chip is moot. I have a very nice 2-channel system hooked up in my living room through my HTPC (Vienna Acoustics Mozarts + Bryston 4B-SST, augmented by a Martin Logan Grotto), and I can assure you the sound quality I am experiencing is quite good.

I have no need (nor the physical space) for a receiver + speaker setup on my desktop computer, which uses an M-Audio Audiophile 24/96 --> Creek OBH-11OSE --> Sennheiser HD650. The quality of the sound card absolutely comes into play on my desktop computer but is a meaningless debate when the SPDIF output is used in the HTPC.

In our research from efforts in this thread we have found that you cannot and will not achieve bit-perfect output in MCE 2005 unless you use exactly one of the four suggested solutions in my post here (http://www.avsforum.com/avs-vb/showthread.php?p=7200938&&#post7200938). In other words, your current card won't work with MCE 2005.

Not sure what your criteria for an audiophile sound card solution is. Realtek powers the HD Audio solutions in the MSI motherboard, and HD Audio is quite advanced and largely superior to the AC '97 audio specification it replaces, despite being integrated into the motherboard.

If you just want bit-perfect ASIO or kernel streaming without using MCE 2005, then that's not the focus of this thread.

Good luck,
-Adam

I'm using Windows XP Pro, I really didn’t think the procedures would be that different between the operating systems. What I'm really trying to find is a good ASIO driver for the M-Audio card as the factory one does not seem to work well at all. The Audigy AISO driver with Z2S card works like a champ but unfortunately wont allow bit perfect output to pass a ripped DTS 44.1 signal. I wasn’t trying to derail this thread and talk about other operating systems then the OP had intended, but you seemed quite knowledgeable on this subject so I couldn’t help butting in to try to glean some info.

I think after one more try tonight I will just have to give up until somebody comes out with a turnkey solution, ultimately I think it would be kickass if I could have all my DTS Music and concert video tracks individually ripped so I could play them back in any order in full DTS surround, but I guess it will have to wait for technology to catch up.

All we're using the onboard Realtek chip for is passing a digital bitstream untouched and not re-sampled to an external DAC. Thus, debating the quality of the DACs inside the Realtek chip is moot.

Was going to say that, you beat me to it.

I have a very nice 2-channel system hooked up in my living room through my HTPC (Vienna Acoustics Mozarts + Bryston 4B-SST, augmented by a Martin Logan Grotto), and I can assure you the sound quality I am experiencing is quite good.

Holy awesome 2-ch setup, Batman. Bryston 4B-SST...drool.. ;-)

-Adam

I think after one more try tonight I will just have to give up until somebody comes out with a turnkey solution, ultimately I think it would be kickass if I could have all my DTS Music and concert video tracks individually ripped so I could play them back in any order in full DTS surround, but I guess it will have to wait for technology to catch up.

That's precisely what I do with some of my DTS Surround Music (such as the Eagles Hotel California), which MCE 2005 categorizes, recognizes, and plays bit-perfect through my M-Audio Transit USB via SPDIF into my receiver's DAC. If you're looking for turnkey, you really should read through the solution that Chung Chang and bling24 came up with for the motherboard above. Granted, I suppose you'd have to use Windows Media Player without the MCE 2005 shell that we use, but it should certainly work. Or you could just buy MCE 2005 and install it.

As for ASIO drivers for others, have you tried Foobar's ASIO and kernel streaming plugins? As I recall when I looked, only the older Foobar versions worked properly with my Chaintech AV710. J River Media Center has ASIO as well, and there's the Winamp ASIO plugin which I could never get to work right either. After all, once we made MCE 2005 work, who needs ASIO anyway? ;-)

Good luck,
Adam

I'm using J Rivers MC11 but it didnt come with a ASIO driver, I'll google it and see what I can find.

Thanks for taking the time to reply.

What I'm really trying to find is a good ASIO driver for the M-Audio card as the factory one does not seem to work well at all.

If your having problems with M-Audio's ASIO, there is something wrong with your setup. These are pro cards and alot (most?) pro software uses ASIO. If ASIO didn't work, no one would buy their cards. I've used various M-Audio cards over the years with JRiver MC, Winamp, Meedio, BPM, Cooledit, and fb2k and never had any problems with ASIO.

Edit: JRiver MC11 does come with ASIO output. Its not a "driver".

Sorry, never heard of a realtek sound card unless its one of those built into the motherboard ones that are far from audiophile quality and would seem silly to worry about bit perfect output anyway.

Pardon my ignorance, but if bitperfect is your goal, then does it matter if the sound card is pro audiophile grade or not? Bitperfect to me means the 1's and 0's from the digital audio source (be it CD, DVD, sound bits on a harddrive,...) are passed to your external amp/receiver untouched. So, in essence, the sound card simply acts as a pass-through device, passing the 1's and 0's out of the computer untouched by Windows or whatever app you are using. If a $5 sound card can accomplish that, I simply don't see how a $500 audiophile card can pass 1's and 0's any better. It is up to your external amp/receiver/speakers to faithfully reproduce the original sound.

I'm no sound expert but my limited understanding tells me that if those digital sound's 1's and 0's are not being faithfully passed out of the PC untouched, you can hear the flaws expecially in the case of DTS source. Even if 1 bit is passed incorrectly, a DTS sound can produce a nasty hiss, pop or silence that can easily be detected by human ears. Since I'm not hearing any of those artifacts out of my onboard sound from DTS sources through long sittings, my assumption is the sound is faithfully and consistently being passed to my receiver perfectly and untouched. If it can handle DTS solidly, I see no reason for it to handle other sound sources (DD, PCM) any less faithfully.

Let me know if any of my assumptions/comments are flawed. I'm always willing to learn.

Jriver MC11 does come with support for playback via the ASIO api

Options -> Playback -> Output Mode -> ASIO

Your card needs an ASIO driver

Cheers

Thomas

Pardon my ignorance, but if bitperfect is your goal, then does it matter if the sound card is pro audiophile grade or not? Bitperfect to me means the 1's and 0's from the digital audio source (be it CD, DVD, sound bits on a harddrive,...) are passed to your external amp/receiver untouched. So, in essence, the sound card simply acts as a pass-through device, passing the 1's and 0's out of the computer untouched by Windows or whatever app you are using. If a $5 sound card can accomplish that, I simply don't see how a $500 audiophile card can pass 1's and 0's any better. It is up to your external amp/receiver/speakers to faithfully reproduce the original sound.

I'm no sound expert but my limited understanding tells me that if those digital sound's 1's and 0's are not being faithfully passed out of the PC untouched, you can hear the flaws expecially in the case of DTS source. Even if 1 bit is passed incorrectly, a DTS sound can produce a nasty hiss, pop or silence that can easily be detected by human ears. Since I'm not hearing any of those artifacts out of my onboard sound from DTS sources through long sittings, my assumption is the sound is faithfully and consistently being passed to my receiver perfectly and untouched. If it can handle DTS solidly, I see no reason for it to handle other sound sources (DD, PCM) any less faithfully.

Let me know if any of my assumptions/comments are flawed. I'm always willing to learn.


Distortion and jitter, it can be added by the soundcard after the bits are processed from my understanding, anyway that was not really that important but I had heard that motherboard based sound cards have a lot of distortion and someone even had a site with measurements and comparisons and motherboard sound cards were by far the worst. Also if you think there is no difference in sound between different sound cards I'll have to disagree with that idea, I have had multiple sound cards in my machine and swapped cables between them and can hear a difference in SQ between them.

Do you know where I can get a good ASIO driver for the M-Audio delta 2496 card, ASIO4ALL wont work at all on this card?

...but I had heard that motherboard based sound cards have a lot of distortion and someone even had a site with measurements and comparisons and motherboard sound cards were by far the worst.
This comparison was almost certainly done using the analog outputs (and thus the onboard DACs), whereas our solution keeps the signal in the digital domain from the source until it exits the computer. Onboard sound is absolutely a poor choice if you are going to plug speakers or headphones directly into your sound card, but it's just fine if you are only trying to pass a digital signal through the SPDIF.

Also if you think there is no difference in sound between different sound cards I'll have to disagree with that idea, I have had multiple sound cards in my machine and swapped cables between them and can hear a difference in SQ between them.

Once again, this is because of the different DACs being used on the different cards. If you bypassed the onboard DAC by transmitting the signal digitally to an external decoder (such as an A/V receiver), I can assure you there would be zero difference in sound quality between any of the cards you used.

Once again, this is because of the different DACs being used on the different cards. If you bypassed the onboard DAC by transmitting the signal digitally to an external decoder (such as an A/V receiver), I can assure you there would be zero difference in sound quality between any of the cards you used.

I would say it the following way.

If you bypassed the onboard DAC by transmitting the signal digitally to an external decoder (such as an A/V receiver), you would transmit identical bits. In addition to the actual bit values the amount of jitter in the signal generated by the sound card can have an impact on the sound quality. The degree to which jitter impacts the sound depends on the design of the receiver/DAC.

Cheers

Thomas

I've read about jitters before but never really understood how it really impacted bitperfect digital sounds.

I fully agree that analog sounds would be quite different on various sound cards. Let's keep this discussion to the digital sound realm where the goal of the sound card is to pass the 0's and 1's out of the PC to an external decoder as they appear on the source, aka bitperfect.

So, using a DTS source as an example, if even 1 bit is off between the source and the external decoder, the decoder will produce a very audible distortion in the form of hiss, pop or silence. So, if jitters on a sound card causes some bits to be passed incorrectly, the end results would be extremely obvious distortions that is audible even to amateurs, not just a mild sound quality degradation only pros and sound measurement devices can detect.

So, if I can hear no obvious distortions from listening to my DTS sound source through hours of sitting, even if there were jitters, I would assume it has zero impact to bitperfect sound delivery. If the high jitters on my onboard sound is not changing a bit from 0 to 1 or vise versa between the source and external decoder, then it is performing no different from an audiophile grade sound card from the point of view of the external decoder. The decoder would see the same exact bit streams. Further more, if the sound card can deliver a DTS stream perfectly (even with high jitters), then passing DD or PCM would be no different as they are all just streams of 0's and 1's to the sound card. I would be surprised that jitters on my sound card suddenly increase when switching from DTS to DD or PCM sources.

Does this make sense? I know the many articles explaining jitters don't make sense to me from a practical point of view if you simply want bitperfect. Jitter or no jitter, bitperfect is bitperfect. There is no "almost bitperfect" due to jitters that go undetected by any human ears simply by testing using a DTS test source.

Actually, even if the receiver receives identical bits the sound quality can be quite different. If the jitter is so large that the bits change that is a transmission error rather than a jitter problem.

For the purpose of this dicussion jitter is simply the variation in the clock intervals for the duration of one cycle. The clock cycle on a S/PDIF link is roughly 5.6 Mhz for a 44.1Khz word clock. If the jitter in the word clock is higher than 20ps (pico) the resulting distortions approach the level of changing the lowest bit in a sample. Very good DACs with multiple dejitter circuits measure around 100-200ps.

A DAC needs to extract this word clock somehow from the incoming signal to drive its DAC chip that translates the numeric values into analog voltages at the intervalls. The fact that you decompress DTS in the middle somewhere does not change this at all.

How much jitter ends upin the final conversion stage depends on the amount of jitter in the input signal and the amount of attenuation the DAC can apply after it receives the signal. Higher end DACs spend a lot of effort on reducing jitter.

The digital outputs from a sound card can vary quite a bit and so can the resulting sound quality. A/V receiver are often very susceptible to this type of jitter since lip sync cencerns prevent the implementation of dejittering circuits with deep buffers.

In the professional space you can find wordclock connections from the DAC back to the source so that the source (e.ga soundcard) sends it samples frequency locked to the clock in the DAC. There is a large amount of discussion on jitter in past threads.

Cheers

Thomas

I've read about jitters before but never really understood how it really impacted bitperfect digital sounds.

Thomaspf and I have had several discussions about the subject of "jitter" in digital signal transmission (read page two of this thread), and suffice it to say that the jury is still out on any audible difference in sound quality (or affect on same), with several leading audio engineers arguing various esoteric issues, all of them requiring very expensive gear to even approach any sort of measurable test.

For the purposes of this thread, the term "bit-perfect" is one we have adopted to mean something completely different than the presence or lack of jitter. "Bit-perfect" for the purposes of this thread might as well mean "bit-accurate" because the core focus here is getting Windows machines to output the proper kilohertz sound frequency when the source material was recorded in either 44.1 or 48 kHz. And the difference in a "remixed" (by kmixer.sys) 44.1 to 48 kHz track and one in which bits have been unmolested is definitely audible, and we have proven that consistently.

Thomas will disagree with me that bit-perfection can be achieved without expensive DAC's and jitter compensators, but I know he shares my opinion that audio meant to be passed to a DAC at 44.1kHz that is resampled to 48kHz is always a bad thing when audio clarity and fidelity is the name of the game. This is one reason that the Vista architects completely retooled the audio code stack for the next version of Windows. It's kind of funny, but all our work to force XP to do what we want might be a quirk of the past after Vista comes out. Can't say we didn't blaze new ground here though!

Let's keep up the experimentation guys, I know this is helpful to others.

Hope this clears up a little confusion and Thomas please feel free to chime in.

Edit: LOL Thomas I see you already beat me to it. May all your anti-jitter quests prove fruitful and rewarding. :-)

Best,
Adam

Thanks Thomas for the explanation although I must admit I still don't quite understand. Maybe I need to educate myself on Audiophile 101. I've read your post over several times and my limited understanding leads me to believe this has more to do with timing than bitperfect issue. So, with incorrect timing, some samples may not be decoded by the DAC at all? Thus the end result is missing samples in the sound? Am I even close to understanding this issue? I'll definitely take your word for it and do more research on jitters.

Thanks Chung_Chang,

the factor that matters here is when exactly you will convert the samples you receive in the digital stream. There is a single pin on a DAC chip that triggers the conversion from the digital value in the sample to an analog voltage that you send to your amplifier.

Even if you translate all the right bits but you don't translate them at exactly equidistant time intervals the resulting adio signal is slightly distorted. If you don't want to wade through the AES papers there is a pretty readable paper on the Lavry web site.

You don't actually need super special gear to watch the effect. There is a special J-signal that you send to your DAC and then you can look at the impact of jitter in the recorded result.

Cheers

Thomas

I've read quite a few articles on jitters in the last couple of hours and am beginning to see the issues. I don't think my current audio setup (receiver, speakers, room, placement, ambient noise) calls for worries on jitters yet. But for my next house with a proper listening room and higher end audio equipment, I will definitely take jitter issues into considerations.

Optical links like TOS-Link will narrow the bandwith of the digital signal because of dispersion and it also weakens the electric signal to a couple of mili-Volt at the phototransistor of the receiver. So every (ambient) noise introduces jitter! (Most audiophiles use TOS-Link instead of COAX (which hasn't the above problem ;)))

A known jitter form is the Data Jitter (to difficult for me to explain in english as I'm Dutch), which occur when the clock and data signal are transferred over one connection like SP/DIF. As these are no 'random jitter errors' it will lead often to hearable audio distortion. Almost all home equipment use internally separate 'connections' for data and clock signal so don't suffer from this problem.

Funny fact: You have an home cinema with SP/DIF COAX connections you don't have to worry about jitter as it is almost not there. If your an audiophile with a separate DAC which you connected through a TOS-Link, then you really have jitter problems and should worry about it! ;)

So my opinion on Jitter is:
Home cinema users: Don't worry as you probably don't have jitter
Audiophiles: Yeah worry about it as the high end audio equipment (like a separate DAC) can introduces jitter big time.

PC Soundcards: If there are soundcards which use the same circuitry for transferring the data and clock signal the these will introduce (hearable) noise in the audio signal. So in my opinion I think it does mind which soundcard to use even when using it only for SP/DIF output. (But if even the cheap SP/DIF soundcard use a separate data and clock signal, then there will be almost no difference IMHO).

Just my 2 cents. :)

There is not much difference between S/PDIF over coax vs. optical. Both have jitter. Even with an I2S connection where the clock signal travels over a separate connection you need to worry about the quality of your clock.

Optical was for a long time considered to introduce more jitter due to the optoelectrical conversion but for example the optical output on the RME card was measured to have lower jitter in Stereophile. For a digital connection to a PC Toslink has the advantage to provide complete isolation from interference and ground transmitted noise.

Home cinema systems in particular are often prone to jitter since they need to be concerend about lip sync. This means they can only utilize very shallow buffering techniques and many just rely on the PLL in the receiver chips. Even expensive gear like the Bryston SP1.7 or SP2 are susceptible to jitter in that space.

Cheers

Thomas

All of this discussion is certainly interesting, and as much as I enjoy arguing electrical theory as it relates to acoustics, I personally consider all this stuff about jitter a non-issue. From what I am hearing in my living room right now, the signal coming from my HTPC is pretty darn perfect. I spent about 3-4 hours after work last night working on the placement of my Viennas and finally nailed it in my room; I've been in audio nirvana ever since.

Until someone can reliably identify an audible difference caused by the presence of jitter in a system like mine (using a double-blind listening test), I'm gonna sit on the sidelines of this debate and enjoy my Viennas. :)

What I meant in my last post was that jitter is probaby the least of my worries given the low end equipment I have compared to most of you guys. Fixing jitter in my current system will probably not improve the low end sound produced by my system significantly or even noticed in my current poor listening environment.

However, something is still not sitting quite well with me on this issue. Why isn't my receiver which contains both the DSP (to decode signal such as DTS) and the DAC, both probably using the same internal clock generator, rely on its own internal clock to trigger the data conversion? In other words, regardless of when the 1's and 0's arrive from the PC to the receiver, as long as they arrive in reasonable time for the receiver to correctly identify the bit (1 or 0) in a given cycle (using the receiver's own internal clock), the receiver would trigger DAC based on its own internal clock. Then, it would be up to the quality of the receiver's own clock generator to determine how much jitter is present, not the clock generator from the sound card in my PC. So, regardless of how bad the timing is on my sound card, as long as the 1's and 0's are still being interpreted correctly by my receiver in each cycle, I would have to get a better receiver with more accurate clock generator to reduce jitter issues, not a better sound card.

I'm sure there is a perfect explanation for this if you can understand what I'm trying to describe. Regardless, I'm with Bling24 on this in that from now on, I'll sit on the sidelines of this debate until my next major audio equipment/environemnt upgrade.

What you describe is pretty much how a synchronous dejitter circuit works.

However, the trick is to have the local clock follow the clock extracted for the incoming signal. The incoming clock is likely to have an ever so slight small difference in frequency. Over time this would build up to either an overflow or an underflow.

I can only say that I recently sold my Bryston SP1.7 after comparing it to a Lavry DA10. The difference worked double blind very reliably but of course I can't attribute the differences only to the superior dejittering circuitry in the Lavry.

Jitter is easily measurable and you can look at the distortions in the analog wave forms. That was all I wanted to contrbute to this discussion about bit perfect transmission. Anyhow, if you guys are happy that is the most important thing and there are other many more important factors than jitter to sound quality. Enjoy your systems.

Cheers

Thomas

Not trying to pursue this topic further as I'm no expert on the subject, but in the age of atomic clocks having accuracy down to 1 second in 3 thousand years and selling for $5, common sense tells me frequency variations between 2 electronic audio devices, both designed to set standards (44.1KHz, 48KHz,....), should be negligeable in the span of a few hours. Sorry to sound so silly and un-academic about it but as I continue to pursue this knowledge further, I'm sure I'll be shown the light some day and will be better informed to make my own decision as to what is and what isn't important to me in my own sound systems. Thanks Thomas for sharing your knowledge on this and sorry to detract from the subject of this thread.

And if all audio equipment contains atomic clocks then I suppose none of this will be a problem. In the meantime we have real equipment in the market and the output frequencies vary in the percent range.

However even if the frequency would be perfect on average you still might have jitter which are just the small variances in the bit intervals.

Dunn calculated in his original AES paper that even for jitter as low as 20ps the changes in the analog wave form amount to distortions equivalent to making changes to the lowest significant bits of the samples.

If people talk about bit perfect than jitter induced distortions are in the same ballpark.

Cheers

Thomas

Yes. I can now see that bitperfect should not just be the same sequence of bits passed between the source and the decoder, but also at the same time interval. So, jitter is part of the equation. How important the impacts are to the final sound output is something I would have to find out and judge for myself, not something I should just give up and dismiss. Thanks.

Yes. I can now see that bitperfect should not just be the same sequence of bits passed between the source and the decoder, but also at the same time interval. So, jitter is part of the equation. How important the impacts are to the final sound output is something I would have to find out and judge for myself, not something I should just give up and dismiss. Thanks.

I suppose I agree with this "re-definition" terminology. Bit "accuracy" is perhaps what we've achieved here in this thread in creating solutions that accurately pass 44.1 or 48 kHz information without signal corruption. So, without getting too far into a semantic argument of what constitutes "perfect," I'll state that the theoretical audio improvement gained through jitter reduction would only be important to me if I had many tens of thousands more to spend on audio system improvement, and only after many other (as Thomas wisely notes) more important areas for improving sonic accuracy and fidelity to the original recording were exhausted (top tier amplification, processing, room acoustics, and source material, to name a few).

Bit-perfection in the sense of forcing 44.1kHz audio information to pass through to the amplification stage at its intended 44.1kHz and not have it be corrupted to 48kHz is, IMO, a significantly more apparent sonic improvement, and can be achieved for a fraction of the cost of any jitter-reducing technology. Like, tens of dollars versus tens of thousands.

As an aside: I, like others here, still do not understand how a time variation in passing bits to a DAC would have any bearing on distortion unless this variation was inconsistent. That is, a seven second delay on TV broadcasts has absolutely no affect on the picture quality of said broadcast, it just "happens" later. Now, if a TV broadcaster took an image of a television show and colorized it a shade of red before re-broadcasting it, that would make a clear difference, and is to me the same thing as kmixer.sys corrupting the bitstream and "remixing" it, which we've essentially solved in this thread. To me, that is quite an achievement, and jitter reduction, though I will not deny its well-theorized existence, is still not cheap enough or meaningful enough for me to throw many dollars towards correcting yet.

My 2 cents (ha!)

Best,
Adam

You are hitting the point. Jitter is irregular between successive samples. The variance in timing often occurs with a certain frequency but for the purpose of getting jitter induced distortions that does not help.

Jitter reduction does not need to be very expensive.

The latest S/PDIF receiver chip from Crystal (CS8416) has a built in PLL that already reduces jitter to typically 200ps so this comes for free in newer equipment.

Many budget components like the Benchmark DAC1 use an asynchronous resampler chip to reduce this number down to about 120ps and even DACS with a secondary synchronous reclocking circuit (as used in the Lavry DA10) now ship for < $1K. This is one of the main reason these budget DACs are so popular.

Cheers

Thomas

This comparison was almost certainly done using the analog outputs (and thus the onboard DACs), whereas our solution keeps the signal in the digital domain from the source until it exits the computer. Onboard sound is absolutely a poor choice if you are going to plug speakers or headphones directly into your sound card, but it's just fine if you are only trying to pass a digital signal through the SPDIF.

Nope, all my comparisons were strictly the SPDIF outputs and there is an audible difference between the three sound cards I compared.

M-Audio was the best, cleanest with slightly better bass resolution at high volumes. The Soundblaster X-Fi and Azuntech DTS were about the same followed by the Audigy 2ZS. Really high volume levels is the easist way for me to tell the difference, if its not clean then its very annoying at high SPL's. Software drivers can make a difference as well.

Hope we can help. As far as I know save for one errant report, everyone that has used the M-Audio Transit USB in the configuration I described in (URL removed due to the fact that I haven't made 5 posts yet) has successfully achieved bit-perfect playback from within MCE from all sources, including DTS wav files and DTS audio CD's. No other sound cards I know of can achieve this save for the RME card and possibly some HD audio onboard sound on newer mobos.

The trick is to either use your onboard sound or another sound source (in my case, the MyHD card) for DD/DTS audio, or if you use the M-Audio for everything,


This annoys me a bit. Many says that the Transit is the answer to everyones problems. Just use your onboard soundcard for DD/DTS. Bah. What if your onboard card doesn't have that feature, oops?

Sorry for that just a bit frustraded after pursuing this for months.


Anyway, thinking of throwing DD/DTS out the windows and use the DVD-player for that and then wait for Vista and HD-DVD+Blu Ray before I get a passthrough solution that will work for everything.

So, I'm ready to pick up an M-audio transit soundcard to use with my music-collection. The only place that has it is an Mac-store. I guess that they are identical? With perhaps the exception that I have to download the drivers myself (I think I can live with that :p).
(they do state that MAC OS xxxxx is required although the text above says that any computer with an USB connections is required).

Just a bit curious about this satement (from somewhere in the thread, can't find it now but copied the text):
One thing I like about the Transit and other external USB audio devices (assuming music PCM use only) is that you can place the device very far from the HTPC without losing any sound quality. You could use a 15 foot USB cable (or several of them if you use a USB repeater), and still get away with a 3 foot or less digital audio cable (coax or toslink). There's no loss in sound quality over a 15+ feet of USB like there could be with a 15+ digital audio cable.


Is that true?
Can't SPDIF easily handle a 15 feet cable without any problems whatsoever?
I will be needing a ~15 feet cable. USB degrades over distances as well and I know that some devices are more picky than others. So I'm not so confident that running a 15 feet USB cable will work at all.

Anyone tried it?

Also, just wondering whether the DAC in the Transit is decent?
For TV-use I'm thinking (no surround there anyway). I don't know what you are paying for in an external soundcard but for that price they could have theoretically putted in a decent DAC. Especially since it's only dealing with stereo on the analogue connections.
Basically it would replace my old SoundBlaster Live! Value card for TV use.
BTW. can the Transit output both analogue and digital at the same time? That would be great because then I guess I could listen to music through the TV as well as on the receiver?
Yeah I know the sound-quality sucks compared to passthrough (and the TVs speakers aren't that good either) but if you're just testing something or searching for an song you don't need perfect audio and it would improve the WAF (eventhough everyone has to learn how to start the receiver it's allways good to have a backup solution ;)).

I've got over 50 ft of Digital Coax connected between my HT Processor and my M-Audio sound card. BTW, the cheep Delta 2496 will do bit perfect playback as well as others but IMO none are worth the trouble as you cannot easily play both DTS wav files and regular files without changing some parameters in the sound card software, so I gave up.

Thanks. Finally bought myself a Transit though.

The bundled USB cable was quite long so I'm only gonna need about 1.5 meters of optical cable (I'm assuming that won't have any practical loss(?)).

Surround will be a later issue. Perhaps everything will work in Vista as it should (I'm not counting on it but I don't think I will devote myself into this before Vista and HD-DVD/Blu Ray.

I have one minor issue though and that is that I doesn't seem to be able to get any output from the onboard soundcard when playing music (through the transit). That means that you will have to start the receiver to hear anything. Not a big issue since that what you'd want mosly anyways. But can be a bit WAF unfriendly I suppose.

Hi Thomas (and other experts here),

I've found the datasheet for the Realtek ALC880 chip. I saw 2 charts that gave jitter information and they are both attached. Unfortunately, I don't know what I'm looking at so was hoping you can help determine what the values mean, good or bad. There are tons more information from the datasheet, but it is a zipped PDF file which I was unable to attach. The zip file can be downloaded here (ftp://202.65.194.18/pc/hda/alc880d/ALC880%20series%20Data%20Sheet%20Ver%201.4%20RN20050902.ZIP) . I'd appreciate any help.

I am one extremely happy camper right now.

I finally have "bitperfect" playback on MCE. I posted a month or two months ago with a problem were I was getting good playback using WMP but not when I switched to MCE. I have been banging my head around this problem for a long time and I finally found the solution.

First off, I am using Giga-byte's GA-K8N51PVMT-9-RH motherboard. It uses NVidia's 6150 chipset and ALC880 for sound. I can confirm that other motherboards aside from MSI can produce "bitperfect sound"

The problem I had was that I had the MCE karaoke plug-in installed on my system and for some reason it somehow remixed the sound in MCE and basically messed eveything up. Removing the plug-in solved the problem. Great bitperfect sound everytime.

Finallyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy!!!

I hope this helps others how might have had the same problem I had. There is a bug with the MCE karaoke plug-in.

I am one extremely happy camper right now.

The problem I had was that I had the MCE karaoke plug-in installed on my system and for some reason it somehow remixed the sound in MCE and basically messed eveything up. Removing the plug-in solved the problem. Great bitperfect sound everytime.

Congrats! We'll note to steer clear of Karaoke plugin for all who desire bit-accurate / bit-perfect sound output in MCE.

-Adam

Anyone found a PCI sound card that uses the ALC880 chip?

Has anyone been able to determine if the ALC882 chip works as effective as the 880 at providing the bigperfect stream? Thanks.

Has anyone been able to determine if the ALC882 chip works as effective as the 880 at providing the bitperfect stream? Thanks.

Let's bump this thread a bit, with what I think some great info:

auto sensing with the M-Audio transit!

How?
Use AC3filter (I use 1.07a) and configure it:
Go to the System tab and under SPDIF options set a checkmark at "Output SPDIF as PCM"

Works for me! Playing wav bitperfect and DVD with SPDIF without changing the software and/or receiver!

Please let me know if this works for others too!

Interesting.

Are you getting AC3 or DTS showing up on your receiver/proc?

I get Dolby Digital as my receiver is DD only, so no DTS.

So I hope someone with a DD/DTS receiver can tell if this really works.

Has anyone else had any luck with the Transit switching automatically between bit-perfect music and DD/DTS?

First off, I am using Giga-byte's GA-K8N51PVMT-9-RH motherboard. It uses NVidia's 6150 chipset and ALC880 for sound. I can confirm that other motherboards aside from MSI can produce "bitperfect sound"

Did you have to do anything special to this board ot get it to run bit perfect? Any special drivers or configurations?

:confused:
No one tried/tested it yet?

I thought Transit owners would love this feature!

Please can someone confirm this?

Maybe I should open a new thread to get more attention?

:confused:
No one tried/tested it yet?

I thought Transit owners would love this feature!

Please can someone confirm this?

Maybe I should open a new thread to get more attention?

I keep meaning to try it out. I'm not currently using the Transit though as I couldn't be bothered with switching modes. I'll try to test it out on a test PC the weekend if things don't get to crazy.

OK, I finally got it to work on my test PC using both WMP11 and Zoomplayer. What Transit drivers are you using? I had to install the oldest driver (v1022). But yes, I can play music and movies and get PCM and AC3 without changing settings on the Transit control panel. DTS wav file is properly decoded by my processor so I am getting bit perfect audio. I am going to try this on my MCE machine now.

The Transit is set to: 2in, 2out, 16-bit, 8000 Hz to 4800 Hz

AC3 filter: Check "Use SPDIF" on main tab and "Output SPDIF as PCM" under SPDIF options on the system tab.

I'll post a follow up once I test on MCE.

Yep I'm using the v1022 driver too, that together with setting the volume to 100% gives me bit-perfect output.

If it is working on your MCE machine, then I think it is time to make this auto sensing Transit news world wide! :) :D

I think Aurora1313 may have done it!!! This is working in MCE but it took a little more work. Firstly, I am getting bitperfect audio. Secondly, recorded ATSC playsback correctly as Dolby Digital.

However, I couldn't get DVD to playback in Dolby Digital. Just got PCM. MCE insisted on using the Purevideo audio decoder. So I changed the merit of the AC3filter and the purevideo audio decoder but still no deal.

Found a reg hack to force MCE to use the AC3filter at TGB:

http://thegreenbutton.com/forums/thread/13915.aspx

Go to start -> run -> regedit.exe
Go to HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion \Media Center\Service\Video
Change PreferredMPEG2AudioDecoder into {A753A1EC-973E-4718-AF8E-A3F554D45C44}

This did the trick for me. DVDs play back with Dolby Digital. I have read that some people have had playback stability issues after doing the reg hack so I will need to see if the hack is completely successful.

And if your not using MCE, its even easier as you don't need to mod the reg.

But for now, things look good!!! Thanks Aurora1313

Glad it is working for you!
Hope more people will have the same succes too.

I unreged the nvidia audio decoder so that's leaving MCE no other choise then to use AC3Filter :)
No issues here so maybe that's better than the registry hack?

<Edit>
BTW I use the .172 version of nvtrpdemux.ax (all other files are .233). That is the last version of nvidia which allowed attachment from an audio codec other then nvidia's own.

I tried it on Friday night on my MCE machine and it is working for me except for Dolby Digital through DVDs'. I will try the reg hack to see if that works. I do get Dolby Digital when playing Live TV through MCE. I am taking for granted that it would work for Recorded TV of ATSC also. I will report back after trying the reg hack.

What is the best way to put DTS Wave files in MCE My Music with album info.?

Thanks again for the info Aurora!! :)

What is the best way to put DTS Wave files in MCE My Music with album info.?


Convert the to wma lossless so you can tag them.

MCE 2005 comes on machine,

Installed WINDVD6.0

AND.....once set up, don't need to modify settings to get:

play WMALossless bitperfect 44.1Hz within MCE
play DVDs ripped to HD with AC3 or DTS surround at 48Khz within MCE

Confirmed bitperfect with the German "surround test" DTS file.

Don't have any details handy, but the realtec HD audio driver is not the version referenced above for the MSI motherboard.

I don't do TV, High Def, or gaming so wont be able to comment on that.

Just happy to have bitperfect audio and digital surround for me and the non-technical savvy family!

Yep I'm using the v1022 driver too, that together with setting the volume to 100% gives me bit-perfect output.

If it is working on your MCE machine, then I think it is time to make this auto sensing Transit news world wide! :) :D

I haven't heard much about this since your post. Has this solution been replicated by other transit users? I also use Purevideo with my setup, so I am very interested in having DD/DTS passthrough working with Purevideo.

Thanks,
Mike

MCE 2005 comes on machine,

Installed WINDVD6.0

AND.....once set up, don't need to modify settings to get:

play WMALossless bitperfect 44.1Hz within MCE
play DVDs ripped to HD with AC3 or DTS surround at 48Khz within MCE

Confirmed bitperfect with the German "surround test" DTS file.

Don't have any details handy, but the realtec HD audio driver is not the version referenced above for the MSI motherboard.

I don't do TV, High Def, or gaming so wont be able to comment on that.

Just happy to have bitperfect audio and digital surround for me and the non-technical savvy family!

I think this uses the Nvidia 6150 chipset. Not sure about the audio codec, but sounds like an ALC880 chip may be there for you to get this behavior.

Thanks,
Mike

I haven't heard much about this since your post. Has this solution been replicated by other transit users? I also use Purevideo with my setup, so I am very interested in having DD/DTS passthrough working with Purevideo.

Thanks,
Mike

I find it strange too, that no more people have replied. I thought there would be more interest in this.

About Purevideo as mentioned a few post above:
I use Purevideo too (v.223) but I use the v.172 version of nvtrpdemux.ax. That is the last version of nvidia which allowed attachment from an audio codec other then nvidia's own. So than you can use DD/DTS passthrough with AC3filter and use nvidia for video.

I find it strange too, that no more people have replied. I thought there would be more interest in this.

About Purevideo as mentioned a few post above:
I use Purevideo too (v.223) but I use the v.172 version of nvtrpdemux.ax. That is the last version of nvidia which allowed attachment from an audio codec other then nvidia's own. So than you can use DD/DTS passthrough with AC3filter and use nvidia for video.


I am sorry I am just reporting, but yes I am able to get DD/DTS passthrough with AC3filter and use nvidia for video. Even with MCE 2005. I am using the M-Audio Transit.

BTW, it appears at least from the manual that the new AM2 K9NGM2 motherboard equipped with the ALC883 audio chipset has the same "autolock" functionality on SPDIF output that the K8NGM2 did, at least from the settings that are shown for the audio driver configuration in the manual. If this works the same way, this will be good news, as this in an AM2 board and S939 production is being phased out.

Does anyone have experience with this board?

thanks,
Mike

MikeSM - so then - are you saying that the K9NGM2 has the ability to play bit perfect playback as well (or should I say the ALC883 chipset?)

Also, has anyone been able to confirm or deny the ability of the ALC882 chip set to be able to play bit perfect?

I find it strange too, that no more people have replied. I thought there would be more interest in this.

About Purevideo as mentioned a few post above:
I use Purevideo too (v.223) but I use the v.172 version of nvtrpdemux.ax. That is the last version of nvidia which allowed attachment from an audio codec other then nvidia's own. So than you can use DD/DTS passthrough with AC3filter and use nvidia for video.

Can you offer it to me?? I'm interested in it but just can't find 172 nvtrpdemux.ax, no longer available now it's too old.

Can you offer it to me?? I'm interested in it but just can't find 172 nvtrpdemux.ax, no longer available now it's too old.

Sorry I meant 177, but you can still find it here at nvidia's ftp:

ftp://download.nvidia.com/Windows/dvd_decoder/1.02-177/

Sorry I meant 177

lol, no wonder.....

Can you offer it to me?? I'm interested in it but just can't find 172 nvtrpdemux.ax, no longer available now it's too old.

Can someone send me just the 172 nvtrpdemux.ax file? I'd like to replace it in my current purevideo installation and see if that enables full passthrough.

thanks,
mike

Well, I did some more tonight on my mce system with the usb transit. It seems to talk fine to my reciever, but the reciever is telling me it's sending 96 khz PCM for music playback. I thought it would not be saying that if it was bitperfect? Am I missing something?

Also, I can't seem to get it to output 5.1 from DVD playback. I do have purevideo, but have ac3filter installed, and have applied the registry hack after each reboot. Still no go.

I am running the latest XP transit drivers (from 2005).

thanks,
Mike

Don't use that one use the older drivers! The 2005 drivers are WDM drivers and don't deliver bit perfect.

Don't use that one use the older drivers! The 2005 drivers are WDM drivers and don't deliver bit perfect.

Now that does sound like a problem!

Where do you find the new drivers? they don't seem to be available on the m-audio website.

thanks,
mike

Now that does sound like a problem!

Where do you find the new drivers? they don't seem to be available on the m-audio website.

thanks,
mike


OLD driver, you need the 2003 one not the one from 2005:
http://www.m-audio.com/index.php?do=support.drivers&k=driver&s=6&p=8165223d4cfae86989745b38a55cdfcb&o=15&f=242

I got my mb to use it's spdif out to FINALLY do bit-perfect. Thanks to whoever said that the karaoke plug-in screws up the audio, I removed it and WHAM! All sounds work now (DD/DTS, PCM) And I'm not uses any add-ons, just the driver.

My motherboard uses a SoundMax chip, so it's different than the Riteck's mentioned earlier. The key is to install only the driver, not the app that comes with it.

"Soundmax chip"? Is it AD1986a?

I got my mb to use it's spdif out to FINALLY do bit-perfect. Thanks to whoever said that the karaoke plug-in screws up the audio, I removed it and WHAM! All sounds work now (DD/DTS, PCM) And I'm not uses any add-ons, just the driver.

My motherboard uses a SoundMax chip, so it's different than the Riteck's mentioned earlier. The key is to install only the driver, not the app that comes with it.

What is the exact model of the motherboard?

Foxconn 760GXK8MC-RS Socket 754 SiS 760 GX Micro ATX AMD Motherboard

It's cheap, I just wanted something that would go into s3 sleep mode well, and it does that perfectly.

It's an AD1988 soundchip.

OLD driver, you need the 2003 one not the one from 2005:
http://www.m-audio.com/index.php?do=support.drivers&k=driver&s=6&p=8165223d4cfae86989745b38a55cdfcb&o=15&f=242


Ok, I loaded these, but now I don't get any audio reing recognized by my reciever. I cleaned out the old ones, rebooted, loaded the 2003 version, connected the transit and went through the popups. No go.

All volumes set to max, no other audio devices in the system. Any ideas as to what's wrong? The driver version is right from device properties. It won't play a thing, audio, DD/DTS, not even a system beep.

Thanks,
mike

Make sure you set the device to be the default audio device in Control Panel. Otherwise Windows won't use it (by default).

Does anyone here gets sound from using an OTA HD tuner card in MCE 2005 w/ a M-Audio Transit. All I seem to get is static.

Make sure you set the device to be the default audio device in Control Panel. Otherwise Windows won't use it (by default).

It is. Still no go.

Well, I finally reached the point of frustration with this approach and for a couple other reasons decided upgrade my hardware, just a bit. :-) It won't be here for this weekend, but probably next week I'll rebuild the system.

I just ordered a new ABit AT832X motherboard, which has the alc882d (d = dolby digital encoding built into the codec). From other reports and the specs on the codec, this should do autolock nicely (both bitperfect audio and autoswitching for DVD surround in MCE), and may even let me run gaming audio over SP/DIF to my Marantz reciever (since it encodes Dolby digital).

I also have ordered a dual core opteron 165 (which should overclock really really well) I got for less than $150, as well as a new 7900 GTO video card. This should be a beast of a machine that should feed HD nicely to my Sony 1080p 60A2000 via HDMI, and even let me run flightsim occasionally, which should look great on this monitor.

Given memory prices are really high right now, I wanted to stay with a S939 system, since DDR2 doesn't buy you much, and I have a ton of good DDR DRAM. The downside is you kind of have to use a RD580 based motherboard to find HD-audio support. All my other systems have Nforce 4 chips in them, and no HD audio support is available.

Will let you know how this all works.

Thanks,
Mike