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Build High-End Audio System w/ Hard Drive Storage?
nganju asks: "Hard Drives have finally reached the size where I can rip down 1000 CDs directly to WAV files, and skip the compression step (read: headache) altogether, ensuring that the audio playback is what the original CD author intended. Now the question is, how do I get that WAV data off the computer and into the amplifier with minimal distortion? Are there D/A PCI cards comparable to high-end CD players? Or is the best solution some direct digital output card (SPDIF) and a standalone D/A converter component? Specific model names would be greatly appreciated."
MOTU, M-Audio, Digidesign, and many other companies make audio interfaces capable of 24 bit 96 Khz audio encoding and decoding, which is well above the 16 bit 44.1 Khz that CDs use. Any of them should do.
You really should consider some type of lossless compression. The "headache" is minimal, and although it isn't the 10x compression of its lossy brethern, 2x is nothing to completely ignore. http://flac.sourceforge.net/
Here, though it does use lossless compression, if you care.
And it's only $129.
GPL Deconstructed
You really should be asking this over at Hydrogenaudio:
http://www.hydrogenaudio.org/forums/index.php
The signal to noise ratio is much better there for this kind of question.
I dub thee... Sir Phobos, Knight of Mars, Beater of Ass.
No, you want compression -- lossless compression. Oh, and just grab an Audigy if it's a Linux box.
Otherwise, be warned that Creative will not give you free Windows driver downloads, only updates.
Don't thank God, thank a doctor!
Seems braindead to me.
Just rip to uncompressed PCM AC3 and pump directly to the receiver via the SPDIF jack.
Or get a receiver with a USB Audio jack, like I do, and your receiver itself becomes the sound card.
http://www.zzounds.com/cat--PCI-Audio-Interfaces-- 2421
t er/search?c=9132
or
http://www.musiciansfriend.com/srs7/g=rec/s=compu
Choices galore.
Unique.
Do an ABX test (http://www.bostonaudiosociety.org/bas_speaker/abx _testing.htm) comparing a MP3 extracted with EAC and encoded with Lame 3.93 --alt preset standard and a wav file.
l oad&name=Elite_DAE&file=painless for an easy guide.
Come back to me with the results.
I think you will be suprised.
99% of the population can't tell a difference.
http://www.chrismyden.com/nuke/modules.php?op=mod
MP3s are not only smaller, they work on portables, and they have great metadata.
Regardless of your decision regarding encoding or not - EAC is a must for a quality extraction!
I dub thee... Sir Phobos, Knight of Mars, Beater of Ass.
Here's one way.... Get a small computer, big harddrive.
Get an M-Audio Audiophile 2496 (~$100) and maybe a right-angle PCI adapter to fit it into your little BTX box or whatever. Load your OS of choice. You've already got plans for the rest - that way should be just fine. Rip your stuff onto the drive (encode with FLAC), hook it up to an amplifier, and you're all set.
The 2496 has already got RCA IN/OUT and Digital connectors (read the specifics on compatibility and what you can and cannot use at the same time) making hookup easy. It will also record at impressive rates and resolutions (playback too if you've got fancy hi-res sources). You can find drivers for most of the following at OSS (these are commercial drivers that run ~$50 for the most common OSs that include free tech support and upgrades for 2 years).
* Linux (x86, Alpha, PowerPC)
* VxWorks (Tornado)
* LynxOS (x86, PowerPC)
* SCO Open Server
* SCO UnixWare
* Solaris (x86, Sparc)
* IBM AIX
* FreeBSD
* BSD/OS
* OpenBSD
* NetBSD
* HP-UX
You could buy a mixer and some mics to do some high quality recordings too. (I've picked up a 10 channel Yamaha mixer [MG10/2] w/ 4 mic inputs (phantom capable) for $99 and a Samson CO2 matched pair of small condensers for ~$120 at Sam Ash to do recordings with a setup very similar to that above and it worked quite well.) No experience with the OSS drivers but they seem to be responsive to email inquirys about specifics and have a free trial available.
I dream of a portable custom BSD based solution that has easy controls (serial keypad and LCD - "real" buttons and switches), could be setup for automated recordings, has a builtin mixer, microphone inputs (phantom powered for my dream large condenser pair), and speaker/headphone driver, AND is powerful enough to run baudline for use in the field. Background processes could compress material as I was recording (incremental, selectively, to be sure you could grab the entire recording - even if your quality had to suffer - but you'd get the highest possible of any given event). The network interface could stream audio at selectable bitrates (.ogg peeling) OR amplify a stream like an internet radio station. AND it could do my laundry for me and fit in a backpack. If anybody else would be interesed in something like this please contact me and I'd love to collaborate. [ bricoleur !AT! 80d !DOT! org ]
Lower frequencies are where MP3 excels - but I will assume you are uninformed, and not a cheap troll.
Besides - there is a HUGE difference between an insanely high bitrate MP3 and a quality MP3.
Encoder and settings make a large difference.
Once again - I challange you to use EAC & Lame with --alt preset standard. ABX test against source and let's talk about your results.
Don't let your bad experience with shit MP3s cloud your judgement. Do the scientific test and THEN talk.
To put as nice a face on this as I can.... you, sir, are an idiot.
Of COURSE the losslessly compressed files are different on disk... they take half as much space! When you uncompress them, you get back exactly what you started with. That's why it's lossless compression. Bits are bits are bits... as long as the bits that go to the DAC are the same, how they're stored doesn't really matter.
THERE IS NOTHING LOST WITH LOSSLESS COMPRESSION. That's why it is 'lossless' compression. The files just take less space. You route the compressed bits through an uncompression program and you get a bit-for-bit identical copy to what you started with.
And I love your 'don't give me the math' line. "Don't confuse me with facts!"
You know that you want to keep the signal digital until it's as close to your amp as possible. Assuming your amplifier has an optical input, simply running fiber from a soundcard's optical output is the best choice. This puts the burden of clarity on the amplifier's internal DAC and power supply. Optical SPDIF seems capable of 15 meters on standard cable with normal drivers. Since the PC end is all digital, component choice is essentially irrelevant. PCI soundcards with optical outputs are common, so let reputation and support be your guide.
If your amp only accepts analog inputs, things get more complicated. A standalone SPDIF-analog converter seems obvious (and leaves a simple amplifier upgrade path in the future) but consider that such gizmos, while overpriced, usually include a heinously noisy wall-wart power supply. Ripple on the DAC's inputs translates to noise in your audio. Careful design can filter this crap, but caveat emptor. Do listening tests.
This can also be a problem with many of the USB audio devices available. Since they're powered from the USB, a bit of digital noise is inevitably coupled to the analog side. Component choice and careful design are essential here. I'd trust any of the big names to get this right. M-Audio and Edirol both make some slick little USB audio dongles with excellent analog stages. A plethora of USB and firewire audio interfaces are avilable.
If your PC is just a few meters from your stereo, then USB is probably the way to go. My first question would be about ground potential differences, between the USB signal and the amplifier's idea of analog signal ground. Feeding the whole mess from the same branch circuit is an easy way to sidestep the question, but I'm sure someone has tackled it. (Clueful? Please reply!)
If you're dealing with a longer distance, real networking may be the way to go. The idea here is to let your PC in the next room serve the files, but put enough intelligence in the hifi rack to do the decoding as well as the DAC step locally. This usually includes a display and interface of some sort, so you don't need to mess with wireless keyboards or whatever. Various network music players are available, with varying levels of software sophistication and hardware quality. I don't believe any of them include audiophile-quality components in the outputs, and power supply noise is usually an issue in these cheapie designs done by digital engineers without an analog bone in their bodies. If you can find one that supports raw WAV file input, give it a try and see if the audio quality suits you.
Most such players rely heavily on the ID3 tag info for database and display purposes, so tagless WAVs might be awkward at best. Alternately, "tune" the network player to an "internet radio station" which is really a stream running from your desktop's player software. The stream server can then stuff tag information into the stream's metadata, which will appear on the display.
Someone mentioned using the Airport Express as an output device that iTunes could throw digitized audio at. Cute, but I'd be skeptical of any analog components sitting so close to a power supply. Anyone done SNR measurements on this sucker? If it worked with software besides iTunes, it wouldn't suck so hard.
You sir a idiot.
"The very notion of "lossless compression" is faulty." No it really is not. Take a 200k text file and compress it with Zip. Rename the original. Unzip the compressed file and now compare them. Wow it just be magic. They are the same.
" If you compress, you lose. I don't get lossless compression. "
Then how do explain the text file?
"How can you substitute one thing for another and then get it back the same way?" Gee I do not know lets try. How about I write five. And then I write it 5. Gee I just got a five to one compression! The number 5 and the word five have the same meaning but one is smaller than the other.
"Don't give me the math". Of course not since you would never understand it to start with. If you take a digital file and compress it using lossless compression when the file is decompressed it will be identical to the compressed version. This has been used since the days of ARC and Zoo and is still true with FLAC, ZIP, and gz today.
See my blog http://ilovecookes.blogspot.com/ for light hearted technical information.
Basically, driving a pair of Totem System 1's with a Kenwood integrated is going to sound better than cheap speakers driven by an Atmasphere.
Electronics is just so ridiculously easier to get working right. Speaker design and acoustics is much harder to get right. It's subjective, for one, and the measurement equipment and physics knowledge required just isn't that common.
Expectation bias and placebo effect are also big factors.