The Pono Player and Promises Fulfilled(innerfidelity.com) |
The Pono Player and Promises Fulfilled(innerfidelity.com) |
I listen to MP3s, I listen to AACs from the iTunes store, I listen to FLAC (concert bootlegs), and I listen to vinyl with a tube amp. I cannot tell the difference, even on quality studio headphones, between a v0 MP3, an iTunes 256kbps AAC, and a FLAC file. I can tell a difference with vinyl and the amp, but while it's different, I would hesitate to say it's _better_, let alone assume the reason why.
He postures as a scientist but he doesn't do his diligence and post any studies on whether or not people can tell the difference between music played back through $5 and $50 DAC chips. That actually pisses me off, if you want to take the "science" side in an argument you should do your homework.
And if you're going to get all that gear together, you're going to want at least decent conditions for listening (quiet, mainly) or those details you paid $400 to get out of the music will be drowned out.
I guess the point I'm making is - who is this DAC meant for? It seems like its attempting to fill a space (or create one) that is already filled.
U2 gave away a new album last year, it has song great U2 tracks in it, it's a strong album from them. The entire discussion about it revolved around not the music but the delivery. Not about them getting old or how they've only made crap since War or any of the other slags on them, it was about the delivery. Maybe it's always been this way but it seems like the appreciation is going down. Seems like in a few years, people won't buy dedicated audio equipment, not generally at least.
Fun fact: Some producers that use digital recording systems will mix in analog noise because the noise itself can be very pleasing in the right dosage. In this way it is used more like an effect rather than an inherent limitation of the recording process.
To quote Daniel Rutter: "[B]etter-mastered music will sound better on cassette than badly-mastered music would at a zillion bits per second."
Here are a few possible differences:
- RIAA EQ curve on the vinyl compared to the mastering on digital music
- Vinyl is not exactly a precision format, it's very noisy and you're getting a lot of audio that isn't even part of the recording
- Tube amps inherently distort sound. This is the "tube warmth" that people talk about; it's just the amp not being very accurate at all.
These aren't "bad" things, but they're also not exactly something to point to for superiority of a medium.
What type of music are you listening to? Certain instruments are handled very poorly by LAME. A v0 MP3 is going to be pretty bad on cymbals, woodwind instruments, and harpsichords, for example.
AAC is far better about this, and I've never been able to blind ABX a 320kbps AAC against a FLAC successfully, even using a headphone setup worth over 2k. I can blind ABX 320kbps mp3s vs flac 100% of the time on those underperforming instruments, and a good portion of the time on music in general - assuming I have listened to the FLAC version of the song extensively.
That being said even on expensive mobile setups I have a hard time even on the 'bad' mp3s. Granted, my mobile setup is a fiio x5 and midrange CIEMs, so it's not nearly as nice as what I have at home. But on the go is where I'm space constrained - I see no reason to keep FLAC files on a portable player. An MP3 or AAC file should be all you need on the go.
But, there is a MAJOR problem with MP3s, particularly in today's mastering climate - converting from WAC/FLAC to MP3 usually adds .5DB to the audio. With the 'Loudness War', there is a significant amount of audio that is mastered to -0.1, and this conversion will push the audio to 0.0, resulting in clipping.
As for vinyl, the difference in sound very well might be the loudness war. A portion of vinyl releases use the same mastering as the CDs, but a lot get a special "vinyl master" where they haven't squashed the dynamic range. Part of this is due to physical limitations of the format - you can compress all of the audio to have the same dynamic range like they do in the loudness war, but this compression is a casualty on CDs - they just want it to be louder. On vinyl, because of the way it works, if you want all of the DR crushed, the average loudness has to actually go down, which defeats the purpose. If you're comparing against a vinyl master with non-compressed dynamic range vs something that came from a cd loudness war casualty, you're getting a very different audio experience. I'd say it's better, and as such have spent a significant amount of money on vinyl, despite it being an inferior format on a purely technical level.
All that being said, with storage space being so cheap, I choose to keep FLACs of all of the CDs I rip, because it isn't expensive for me to do so, and I don't have to worry about losing information whenever some new compressed format comes out. I keep AACs and MP3s on my DAP, and this way I won't have to re-rip everything when some new format comes out. It's inexpensive to keep archival quality audio around, so why not do it?
" This test pitted the Pono Player against a Galaxy Note 4 and an Apple iPad Air. This was actually the hardest test to get a reliable grip on identifying the various players... Because differences were so small I had to rely more on a subjective approach—just relaxing and sensing how the music was affecting me. Usually it would take me 5-15 minutes of listening and slowly switching back and forth between sources before I could determine which was the Pono Player."
One of these things is not like the other...
Yes, I'm not clear on how his blinding setup works, but this sounds like the sort of activity that could easily unblind oneself.
Search for "Blind Testing" on the first page: "I use a passive three-way switch to route the outputs from the three sources to a pair of headphones. ... without looking, I scramble the cables in each hand."
TL;DR - Don't you bring your SCIENCE in here, I like being conned with audiophool nonsense!
Yes, a 192k playback sampling rate is a joke and maybe even superstitious. There's not much of a reason for anything to play back at over 96k and many argue 41k is sufficient for mixed/mastered material. Point granted.
However good engineering is most definitely not a joke. Using higher quality DAC's (digital-to-analog converters) and better designed circuitry, spending the extra few pennies and dollars here and there to get the parts with lower tolerances, designing the shape of the product around the circuits instead of cramming circuits into the shape of the product, these are the exact things I would expect to see in a better music player.
Sums up the subject really well.
Yes, there's lots of crappy audio kit out there, but it's not hard to get superfluously good stuff either.
I think people just liked the fact that in the 60's and 70's you could make a hobby out of actively pursuing a better sound in amplifiers, and are disappointed that some time in the 80's it became possible to buy gear that was indistinguishable from perfection, and these days it's not even expensive.
Neil Young should have focused on headphones, or speakers - that's an area where there's still detectable amounts of distortion. But that would require something more difficult that attaching a big branding effort to a solved problem.
If you look at actual studies of the performance of smartphone audio you see it's not a trivial task to get right[0], if we're seeing problems on a large company's flagship model like Samsung's Galaxy S5 then this isn't a solved problem.
Playback at home while plugged into a 120V power grid with equipment that only needs to fit into a shoebox is a bit different from playback from a device that's simultaneously a computer and a phone which also happens to have severe space and power constraints (and a giant color touch-screen to boot.)
[0] - http://www.anandtech.com/show/8078/smartphone-audio-testing-...
Please don't interpret what I'm about to say as audiophoolery. I'm an EE. I haven't worked in audio a whole lot, but I've looked at things enough that I feel like I could do a reasonable job at designing a DAC.
The one thing that a higher sampling rate does is that it makes good engineering easier. Designing an antialiasing filter that both prevents aliasing and still sounds good at a 41k sample rate is hard. There's just not a whole lot of room between your passband and stopband, so you need a really sharp filter. Add in component tolerances, and your sharp filter is probably going to be a set of mediocre tradeoffs.
I suspect when the author talks about "smoother treble", they're probably referring to the fact that the anti-aliasing filter doesn't need to have a weird phase response to be able work properly.
It's possible there are counterpoints to this that I don't know about.
The outer grooves have relatively more fidelity (needle moving faster) than the inner grooves, so the master may be progressively tweaked to add more high end as the content gets closer to the inner groove. The closer you get to the inner groove, the lower the playback fidelity.
12" Singles have larger groove spacing so you can slam them way harder than an LP. There is a greater margin of error and less chance of the cutting lathe skipping or hopping during cutting the master.
The process for creating stereo vinyl recordings is pretty much a compromise that allows stereo content while maintaining compatibility with mono playback equipment. This results in another series of compromises/limitations in the mix, like severely limiting the amount of truly stereo content. Extremely loud or bass heavy content that is panned wide would cause the cutting lathe to skip, etc.
For robustness sake, I really really love keeping as many filters as possible in the "gentle" range of design specs. Sometimes that's not possible, but if it's possible to do it start to finish through the signal chain, that makes me very happy.
I'm not saying that we can hear stuff on the recording at 30kHz, but the stuff right at the edge of our hearing around 20kHz will get less messed up.
My gut feeling is that it's way easier to make a well-engineered audio system when there aren't any sample frequencies that cause any of the stopbands to be close to audible frequencies.
In terms of it being easier to filter an orginal signal than to filter a downsampled->upsampled one... my gut goes the other way, but it's something I'd definitely have to look into before I'd comment either way. Interesting question though.