Hey there,
could you me tell if there’s a difference between recorded stuff per USB-Overbridge or over an external audio interface?
Cant imagine that USB has the same quality as my Soundcard.
Greetings, Loom 
had the A4 on 16 kbit (/or whatever the heck mbit kbit kbs…lazy right now=) … had the Sh101 run through ext input A4 AND into focusrite 2i4 at the same time…
could not hear a difference
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I’d be curious about an A/B test with proper monitors as well.
Have not properly A/B’ed but my at least according to my ears, the quality is rather good (ie. comparable to running the outs to an UA Apollo). It’s certainly good enough for serious work IME.
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What’s your sound card?
I’ll do a phase cancelation test with proper spectrum
analisis for you later this week.
Apart from the quality it self, I am getting way lower volume over USB, any way to fix it? or I might be doing something wrong?
Think usb signal is fixed for overide cliping problem if i remember well so you just have to put an utility and gain like you want… But you know if it’s lower it’s not a bad thing you need headroom… Just boost your monitors a bit
As USB is plenty fast to transfer many channels of lossless audio, audio over Overbridge will surely sound identical to the A4 audio post-digital-fx and pre-output-DAC.
By going out to a soundcard, you’re adding two conversion stages, A4 DAC -> Sound Card ADC, no? It’s hard to imagine that this is better than going directly digital into a computer.
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I didn’t like the quality of the synth audio over usb at all. It sounded similar to my experience if using the Virus over usb…(thinner/flatter)
As a soundcard however the usb to analogue sounds great (go figure)
I much prefer the sound of the synth going into my interface (Digidesign 192)
I, for one, am a bit skeptical towards claims of varying (audible!) differences between soundcards.
Granted, I don’t have much experience with high-end soundcards.
But aren’t DACs / ADCs run-of-the-mill commodity components nowadays? They’re in almost any device you can buy…
These parts would be totally useless crap if they couldn’t reliably encode the original signal to digital (or the other way around).
They’re supposed to be transparent, e.g. there’s an analog signal going into the ADC, and the same signal is going out, but in a digital representation, which you can then convert back to the original signal using a DAC. If you compare the original signal with the re-analogisized signal, they should be exactly the same (or same enough anyway to fall within a certain tolerance, and on high-end audio equipment, that tolerance should be such that you definitely cannot hear a difference).
So naturally there is a cap on bandwidth in the conversion. Obviously there’s a cap at the high end, but that’s beyond human-audible frequencies. On audio equipment there’s often a DC removal stage at the low end. But the DC removal cutoff should be below human hearing range as well, e.g. somewhat below 20 Hz.
If there are audible differences in the bass end of the spectrum, then imo something is going really, really wrong. Or there is some other stage in the recording/playback process which colors the signal, such as some EQ to warm-up the sound or something.
Anyway that’s my understanding. if this is incorrect, feel free to educate me…
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Man…you have no idea! Just for starters…
http://www.audiophilleo.com/home/definition/jitter
http://www.el.gunma-u.ac.jp/~kobaweb/news/pdf/sdarticle.pdf
Honestly…studios dont spend 5k on Prism interfaces for no reason!
i have an apogee ensemble TB and there is a noticeable difference. probably not enough to hear in a mix, but A/B, the difference is there.
hm. would be interesting to actually have something quantifiable…
a phase-cancellation recording would be neat.
What are the real-world effects of this?
afaik jitter can be heard as high-frequency noise, if it’s bad jitter.
should be negligible in the real world though?
the paper posted above is about “high-speed DACs”, it doesn’t ever mention the sampling rates used, but probably they’re talking about the GHz range, i.e. an entirely different domain than audio DACs.
the other article is just PR material.
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High End Mastering Studio not working only in the Music productions territory but they working for movies, documentary, a lot of TV advertisement etc… so they need to get a higher audio quality in bit depth and sample rate. (… quality converters for multitrack acoustic instruments recording such as orchestral, band etc…)
But in a regular basis, i mean for Electronic Musicians, Urban Musicians, the truth is at the end the mastering engineer will not working on a Mastering with a file bounced (recorded) at 32bits 192kHz …
(for decades Home Studio is for Musician and no one can afford a 5000 euro sound card interface as a high end studio…)
As 16bits 44100kHz is the rule i really think people can work through Overbridge and release Vinyl’s in a Professional way (or Passion anyway)
It’s probably sufficient for kids who play pirated MP3 160 Kbps anyway
I really think that, no offense, but as myself a label manager since few years i think people need to overpass that kind of technical things and release a lot of good tracks. It’s certainly the key and not the Audiofile excellence challenge … perfectionism is not always your friend 
Edit: but at the end do what is sounding the best at your ears because it’s your tracks and it deserve you to be satisfied at FIRST !
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Right now is not about sounding “perfect”, is about developing your own sound. You can sound “vintage” or “modern” all ITB.
People concentrate too much on quality instead of having a vision.
Use what ever suits you.
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Well said, commonsense.
I’d rather spend 1000-5000 on quality synths than a hyped super dac sound card and record them through a usb synth or an old used apogee $150 sound card
Or record to tape
yes, I am also in the camp that is not entirely convinced of the merits of 192/32 or using the most expensive converters. If the source sounds dope, it will sound dope through a passable converter as well.
many “x clearly not sounding tha same as y” cases are easily fixed with a subtle surgical eq move anyway
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If you have Ableton: better buy logic x instead of a 2000 EUR soundcard 
and I think to hear a difference between a UAD Apollo and an Motu you have to spend another 1k or more for studio monitores and another 1k for room treatment. But If you are looking for really high audio quality, feel free to do so.
My Motu 828 is 10 years old and it sounds not so soft and deep (bass) like my focusrite for example. So there are (tiny) differences between soundcards when it comes to DA conversion.
What would be interesting is, if you can clock the AD converters of the A4/AR to external clock sources like RME fireface or a masterclock.
What gave me better results in mixdown is to work with 96khz / 24bit. The top end sounds a lot smoother and less harsh then working in 44.1khz. But yeah I also run a label and have released music from artists in 44.1 khz and also in 96 khz … and at the end its the music what catches peoples attention.
This is exactly what I was thinking!
Using Digitakt and Digitone as an example multi-track recording in DAW over Overbridge, you are not leaving the digital domain, therefore no conversion stages and no change in the original signal from Digital to DAW.
In my case I A/B’ed TRS cables audio connection vs. Overbridge and OB sounds cleaner, no coloration from preamps or line level ICs.
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