Slashdot Mirror

Er, but they could've done that already. Lightning audio output already works, doesn't it?

Yes, it does, since iOS 5.1.1: http://www.apogeedigital.com/p... (and many others)

Anyways, Apple's built-in DACs are widely known for being better than almost anyone else's. I'm not an audiophile, and I never had to worry about whether the random headphones or stereo system or speakers I had on hand had a quality DAC, but now I do - and it'll cost more to boot (especially for a mediocre one, let alone one as good as the one Apple used to have).

I live in the pro audio world, almost at the bleeding edge of it I might add. What the other poster stated about sub 5ms latency is true. Get yourself an HDX system and test it out. Also, there's this, this, the entire Apollo line, and even an entry level interface that's Thunderbolt based.

IMHO, nothing will compare to using a mixer, but that's for the functionality and quality. I would never pitch a Mackie as a decent recording system, live sound, sure, but, unless you've got an Onyx, it's crap for recording (and even then, just decent). A mixer to me is something like this. My interface is a Fireface 800 and I have 0 perceived latency (so long as I'm not sending anything into the DAW for processing on my cue mix, or merely playing back from the DAW and recording new tracks). To claim something as 0 latency would be incorrect for everything, there will be latency. To claim that the conversion process adds less than X ms of latency is what we're talking about, and the Thunderbolt stuff from MOTU (not even a stellar name in the industry) is leaps and bounds beyond my Fireface (it should be, my interface is 8 years old now), clocking in at sub 1ms at the hardware itself (seen in info note in the link above), with the Fireface being a respectable 5ms at the lowest.

Long story short, if you're looking for the lowest latency and a professional setup, Thunderbolt or PCIe is king. If, as it seems from the photo you posted, you're working in a prosumer or entry level situation, than USB will suffice.

(Other sources: I work in pro audio. A number of years of experience behind the board in both live and recording environments on everything from small projects to working with the likes of Bob Mintzer. Much of my knowledge comes from the real world.)

Its only/main real use will be DisplayPort.

Wow, for a "geek" site, Slashdot seems inordinately populated with techno-IDIOTS, who don't bother to KEEP UP on IN THE PIPELINE THUNDERBOLT PRODUCTS. And there is beginning to be interest shown by other companies, like Canon, AJA, Apogee, Sonnet, and others.

Does anybody have any quantitive measurements of the Apri 2010 Mac book pro As i'm interested in doing some recording with that wondered how good a quality I'm likely to get.

Good enough!

I started buying Macs because you never know what you're getting with other vendors. With a typical PC, 1:1 audio transformers solve any grounding issues but the power supply is usually unshielded and spewing RF noise into the analogue stage. I've yet to encounter an intel based Mac that exhibits either of these problems.

I very much doubt anybody could differentiate (blind listening test) between a recording done using the onboard MacBook chipset and high end external A/D. I've never used the onboard microphone preamplifier and wouldn't expect much anyway. Grab a duet if that's a requirement.

USB and Firewire both hook to real audio interface units well enough, although, if you want to go high end, you might still want to use a PCI interface card.

Apogee Rosetta 200 + Firewire expansion

Apogee Rosetta 200 + Firewire expansion

if by "sound card" you allow extrapolation to the more general term "audio interface" there are plenty - M-Audio, http://www.m-audio.com/ Echo Audio http://www.echoaudio.com/ Mark of the Unicorn, http://www.motu.com/ Digidesign, http://www.digidesign.com/ RME, http://www.rme-audio.com/ Apogee, http://www.apogeedigital.com/ Edirol, http://www.edirol.com/ etc.

Here be Apogee they make reasonable convertors. I know people doing voice work with the audiophile but let's not pretend it's professional kit, it isn't.

It's hard to make a recommendation without knowing at a granular level what you want to do. How many inputs? How many outputs? Is latency an issue? What about frequency/bitrate? Digital inputs? Analog? MADI? Lightpipe? Some light reading... On the ULTRA high end, you would go with Apogee- http://www.apogeedigital.com/ - these are some of the industry's best da/ad converters; and with something like a big ben+rosetta on firewire, you'd be in good hands. Another contender could be rme http://www.rme-audio.com/ Then there's motu's line of products - http://www.motu.com/ - I've personally owned several of their interfaces and can tell you right out of the gate they're great. Good bang for the buck... Then you've got m-audio http://www.m-audio.com/ edirol http://www.edirol.com/ presonus http://www.presonus.com/

I don't recall what OS you wanted to run, but the Apogee Ensemble is supposed to be "the stuff" I've heard.

Of course there are differences between good and bad SPDIF outputs (good and bad systems with SPDIF outputs to be precise). The impedance, the connectors, the regularity of the data output, the jitter... suposing the system won't resample the datas.
Concerning your computer, I don't think it would have any problem in forwarding data at the right rate.
Avoid too much cpu-intensive tasks when listening your music.

People talk about jitter and it's interesting because it mainly affects only the end segment, the DAC, because before it you can do all the crap you want.
If you remember to have a good DAC that will just put the data back to normal (I mean jitter-less).

An amazing analysis can be found here (not so far from your setup) Apple AirPort Express Wi-Fi Hub-D/A processor.
This guy uses a "bad" setup ending in 0% (yes, 0) data lost. And with a reclocking DAC with less jitter than a multi-thousands $ system.

I may recommend you the Mini-Dac if it's in your budget (not so expensive for high-end systems, but expensive compared to $99 stuff).

Would Pro Tools count as "damn good"?

Yes Pro Tools would certainly count as "damn good", and also happens to be in the price-range that I said "costs more than many computers". Other "damn good" interfaces come from Apogee, MOTU, and plenty of others.

There's a lot more to quality CD playback than the number of bits in the D/A converter. The analog section of most sound cards is so cheap that you get quite a bit of hiss and distortion in a D/A/D conversion such as you describe.

Secondly, there is the matter of the quality of the A/D conversion. Again, you have some cheap-ass 30-cent analog parts in there upstream from the A/Ds. And I doubt the A/D converters themselves are all that great, even if they're labeled "24-bit".

There are reasons that professional studio time is so expensive. Mostly it is the recording engineers' time, but the equipment they use costs a helluva lot more than a Sound Blaster Audigy or whatever you got in your white-box PC.

Even lower-grade professional studios use things like this to make digital masters - stuff costs many thousands of dollars, and is built with several kg of high-purity copper in the "analog" parts of the box.

Try putting one of these between the digital out of your computer/cd player/what-have-you and your amplification system. Aha!

And yes, I'm a tube guy when it comes to instrument amps. My crackling, hissing 1961 blackface Fender Showman-Amp makes any speaker sound like a whole 'nother ball game. No master volume on this so it won't distort without making your ears bleed.

Tell that to these guys:

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

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

Apogee makes gold CD-Rs that are used in high-end digital audio. Might be worth a look...

D->A->D works fine with the combination of my MOTU 2408 and my Apogee PSX-100.

I believe Nyquist's Theorem states that to reproduce a frequency N, you need to take *AT LEAST* 2N samples/sec.

No, you need exactly 2N samples/sec, and any more are superfluous. This is what makes Nyquist's Theorem so useful.

Say I make a low frequency wave, say 1hz to keep things simple. Make it a nice gentle sine wave, with peak values of +/- 1. Say I'm lucky, and using a 2hz sampling frequency, I sample right at the peaks (and troughs). The sampled stream is 1,-1,1,-1,1,-1... which, if played back, represents a *triangle* wave [...]

What you are doing here is (incorrectly) interpolating values between samples. If you don't do that, and assume that it is a sine wave, you don't have this problem, and the signal is exactly reproduced with no errors. Re-read Fourier :-)

Theoretically, the "best" digital sound would require an infinite sample rate, which would cause any wave to be reproduced exactly.

Only if you are interested in inaudible frequencies. I'm all for overkill in this regard, but even 50-60kHz is overkill. 96kHz was probably chosen because it is an even multiple of 48kHz (the DAT sampling frequency).

(generally, it's better to take more bits than necessary and chop off the least significant, since the ADC process isn't perfect, and there's an error of 1/2 LSB).

Bear in mind that a very high end ADC might give you a 120dB range, which is almost 20 bits. Also, simply "chopping" the bits isn't a very good idea either. Apogee's web site have some quite good info on this; check it out for more information.

ATRAC encoding (used on MiniDiscs) sounds a helluva lot better than any other lossy audio compression I've heard, I wonder why they didn't review that?

In fact, on my system at home (Cyrus Amp Pair, Apogee DA-1000E, obscenely thick cabling, and home-assembled ear-tuned speakers), I find it hard to discern between the MiniDisc and the original! (Ok, this is definitely flamebait in the audiophile crowd but I can probably get away with it on /.)