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Lynx Studio's LynxTwo
It's a VERY high-performance card. That said, it is a professional card, and thus rather expensive. Compatible with windows, mac os, Linux, and FreeBSD. Possibly other BSDs as well.

That may be necessary in order to synchronise the audio. The 48Khz input source might not be exactly synchronous with the output and hence has to be resampled.

Take for example: you want to mix together a digital input at 48KHz and the microphone input. The 48KHz input is driven by a clock in whatever is providing the signal. The microphone is sample using a clock generated by the soundcard. These two clocks will be fractionally different, which will cause problems with under/over flow of data across the clock domains, which would result in some distortion every few seconds. The ideal solution is to make all the system clocks synchronous - effectively genlocking, so that all sampling and output is driven by the same clock source. This is not an option on most consumer devices, and hence the _only_ solution that will lead to audio playback with no distortion is to sample rate convert, even if it's between two 48KHz sources that are running at marginally different rates. If the clocks differ by 1 ppm then you still have an extra second of samples every 20.8 seconds.

I agree it's not the best option, but it is probably the most sensible given their market. You want to able to do fully synchronous digital audio you need some professional kit that lets you drive its internal clocks from a single source. Something like a: http://www.lynxstudio.com/lynxl22.html should do the job.

This is some serious sampling hardware [no affiliation]:

http://www.lynxstudio.com/products.html

Would love to be able to run a modern OS on Harvard Architecture hardware - something like OpenBSD on a Data General AViiON with lots and lots of Motorolla 88K CPUs.

More recently, a big honkin hypertransport backplane with a mix of quad-core Opterons and FPGAs.

Bonus points if there's an onboard analog processing unit.

Soundcard would have to be the LynxTWO-C: six input channels of 200K samples per second at 24-bits per sample.

$50k is not required for very high quality sound. Do something like this:

~$700 for Lynx L22 sound card (24 bit/192kHz fully balanced analog IO):

http://www.lynxstudio.com/lynxl22.html

~$1300 for a pair of Mackie HR824 powered monitors (flat response, amazing low end for "small" speakers):

http://www.mackie.com/products/hr824/index.html

So there you have a $2000 very high quality audio setup. (Assumes you already have a PC to plug the sound card into.) I run WinAMP with the MAD full precision MP3 decoder plugin and the low-latency ASIO output plugin with the above noted sound card and powered speakers.

Recently I have been experimenting with LAME encoding and the Foobar2000 player. I like the fact that this combination will preserve the segues on album oriented music (so called "gapless" playback)

I have been doing the encoding on my Powerbook using the Blacktree LAME encoder plugin for iTunes (I swapped out the provided LAME executable for one of the new beta builds so I could try out the "VBR new" settings.)

I'm not that familiar with the SACD standard, so forgive me if I'm missing something obvious... but can a normal DVD-ROM drive play back the audio from an SACD?

If so, what's to stop me from using any normal audio recording application along with an appropriate soundcard (Like, say, a Lynx AES16 with the output looped back to the input? Sure, there'll be a very, very slight drop in quality (that no human should ever notice) since you're going from a full rate DSD stream down to a 24-bit/192khz PCM signal, but you'd have to drop down to at least that to burn your backup copy to a DVD-A disk anyhow, so it shouldn't really matter, should it?

Tell that to these guys:

http://www.apogeedigital.com/products/rosetta800.p hp

http://www.lynxstudio.com/lynxtwo.html

There is no physical possibility of having *good* onboard audio. Even with all the above construction techniques, it's damn near impossible to completely isolate the prodigious amounts of digital noise that a typical computer produces.

A much better idea is to run a digital link to an outboard DAC that has its own power supply and is outside the computer. That would actually give you extremely high quality audio, assuming the DAC box is properly designed.


As far as the "onboard" audio, I don't know. But Lynx TWO is a multi-channel professional sound card that fits in a PCI slot (and I mean real professional, not fake-spec professional) Check out the freaking specs!

. It's like somebody trying to buy a backhoe and you suggesting "Why not use a shovel? They're smaller." Because it doesn't do the same thing! If you want to record the exact soundwave produced by a singer on a high note, and play exactly that back into your ear, then lossless compression is your only choice.

No, using your analogy, it's like saying, "hey, I need to dig a hole in the garden to plant the seeds for my corn. I think that I'll use my backhoe." When the hole that you need to make could be just as easily made with a shovel, and in probably a much more effecient manner.

You have to use the right tool for the job. There is little need for playback of lossless audio if your sources aren't good enough to hear the difference. Particularly when they are grossly inadequate. "Hey, I'm listening to lossless audio while I jog." Whoopdido... who cares. Does losslessly encoded audio improve that experience? Probably not.

Now, lossless audio through a proper system (meaning a real sound card, none of this Creative resampling stuff). With a really good set of speakers and/or head phones. Now that can make a difference.

For really good sound cards, look at Lynx Studio, RME Audio, or M-Audio to name a few off the top of my head.

Pure digital is too big, Flac is 2 to 3 times smaller, and therefore this device fills an essential niche: it gets your Flac files to your receiver without requiring a compact disc

2 to 3 times smaller? Not really. 1.5-2 times smaller is more like it. Audio has too much information to compress more than that. You get about 55-65% compression. And most of that is due to the fact that left and right channels of audio tend to be relatively similar.

I'll give you that it would allow you to get your FLAC files to a receiver, just that you aren't really benefitting greatly from the files, as the devices analog output will not be as good as it needs to be.

If you're looking to get sound, then MB sound is just fine. I use it for filler/background noise all the time and love it.

If you're looking for music, they still have miles and miles to go before they will compete. Check out products by Lynx,M-Audio,,RME and Digital Audio Labs

Also check out this thread in a forum for a list of just some of the cards that are worth looking at.
HiFi Sound Cards

And don't be fooled by statistics and numbers, even the best DAC in the world can get messed up by some 2bit clown laying it down with the wrong analog circuitry to support it.

I'm not saying that the people who lay out all these cards are 2bit clowns, just that people look at the numbers and don't use their ears all too often.

The most important thing is do you like the sound that comes out of the system. If yes, then who cares what else is out there. Be happy with it.

That's a great link you posted. Even better, it does appear at least 1 is supported under Linux. The ALSA Project's Soundcard Matrix lists support for the Midiman Delta 66, and there's also OSS/Linux beta support for the LynxONE (though binary non-free).

Of course, they'll both set you back 400-500 bucks... Fortunately, the still well-performing Turtle Beach cards have long enjoyed good Linux support.