My old Presonus Firestudio Mobile has the serial type of S/PDIF connector (AES3 I think?). I've never used it, but is it simply a matter of getting a physical adapter or am I facing a protocol mismatch?
A new interface is in the plan eventually, but trying to put it off if I can.
I think AES3 won't work - SPDIF is a protocol, not a connector type.
It says S/PDIF next to the connector. There are adapters available. I think it used to be the professional version of S/PDIF apparently.
It's no worries - analogue is fine for what I'm doing. 
It says S/PDIF next to the connector. There are adapters available. I think it used to be the professional version of S/PDIF apparently.
It's no worries - analogue is fine for what I'm doing.
Ah, OK. Now I understand. Yeah, then you should be able to get an adapter.
It's no worries - analogue is fine for what I'm doing.
It's fine anyway IMHO, Finally; I'll be using it exclusively, and the TS connectors at that (short cable runs).
I think the fact that KPA only uses 44.1 is limiting. moving up to 48kz will result in significant overall quality of a project considering that there are multi tracks.
What are you basing this on? Why would a higher sample rate equal better quality? 20 - 22000 (ish) Hertz are represented and reproduced perfectly at 44.1 kHz sample rate. 48 kHz sample rate just moves the low-pass filter cut off up to 24000 Hertz. Your statement about there being multi tracks leads me to believe you are mixing up sample rate with bit rate, two very different things.
Yeah, as Sam says, bit depth will affect summing accuracy, but the only advantage I can think of in using the higher SR in this case concerns bell EQ curves at very-high frequencies - the "integrity" of the bell's shape will be better higher on the spectrum in the latter's case.
I've drawn a blank as to the technical term for this, but it's a legitimate concern for some engineers (definitely not consumers 'though). Essentially, bell curves are truncated (LP-filtered) at the limits the SR can faithfully reproduce, meaning that they take on a lop-sided shape. Hardly anything worth worrying about in most real-world situations 'though IMHO.
.
I've drawn a blank as to the technical term for this,
Is it cramping, or aliasing?
I'm talking about other tracks, software instruments and doing any editing on any of these tracks. I realize the KPA is already at 44.1 but any editing can give better results at higher sample rates. Many software instruments sound much better at 48khz than 44.1 and even though it might be incremental for each track, but when you combine many tracks, it can be significant.
I've never experienced that and can't think of any scientific reason why that should be, other than sub-standard converters and filtering is letting through too much high frequency content to create aliasing. I wouldn't have thought that would be a problem with any of today's modern converters though, budget or otherwise. As for editing giving better results, again, I can't see any reason why that would be the case. Can you explain?
As for editing giving better results, again, I can't see any reason why that would be the case. Can you explain?
Time stretching is the only scenario where I can see an effect
.
Is it cramping, or aliasing?
Could be "cramping", but I don't think so, Dean. Sort of rings a bell(!) 'though.
It was discussed at length in one of those epic BeerGutz threads where the utility of using higher rates was debated. The thread's too-long to re-read and I've thrown the link away as I'd gleaned all I needed to from it.
Could have been "blocking", cramping or anything along those lines. At any rate it involved the asymmetrical shaping of bell curves at high frequencies in order to fit them into the available frequency-response window.
Display MoreThere are probably advocates for all sort of advice on what's a good sample rate to use. There are those who advocate an optimum sample rate like Dan Lavry who proposed that 60Khz (which we don't have) would the the optimum to use. Then you have Plugin makers using upsampling internally to get more accurate results (example Cytomic's The Glue) to benefit those who use 44.1KHZ.
https://www.soundonsound.com/s…psampling-within-plug-ins
Then again there are microphones that has a range as high as 50khz such as Sennhiser "MKH 8040"
DVD audio is 48kzh and I think most people will hear a noticeable difference as I do.
Digital hardware synths that I've seen and own are mostly 48KHZ sampling rate or higher.
The KPA internal "algorithm for the tube simulation runs on more than 700 kHz sampling rate" isn't incidental that 44.1khz wasn't used
https://www.guitar-muse.com/kemper-profiling-amp-2949-2949
Also,( I don't know the science behind it ) reverb plugins, based on personal experience, sound better at higher rates. The reverbs alone are enough for me and given all of the above, I don't see a benefit in limiting how you run a digital audio workstation to 44.1KHZ sampling rate.
Well, you've kind of just rallied for NOT running a DAW at higher sample rates. As you pointed out, many plugins (and DAWs themselves) run internally at much higher sample rates to ease calculation of certain processes and to help reduce errors. Slate Digital and Fab Filter are two other software manufacturers that come to mind. As for hardware, yes, even the Kemper reportedly runs at over 700 kHz internally. However, the sample rate of DAWs is mostly relevant to the recorded audio, and I think that anyone would be hard-pressed to hear the difference between a sine wave at 22kHz and 24kHz, when comparing 44.1kHz to 48kHz sampling rates, if they could hear a signal at all (most adult humans can't hear beyond 16kHz). As for 60kHz being the ideal sampling rate, that has more to do with NTSC broadcasting (where video runs at 30FPS) and encoding than pure audio recording, processing and mixing. The subject is quite a deep one. I recommend plenty of research
I recommend plenty of research
I recommend making music instead of plenty research into a (currently) obsolete topic given the fact that nowadays pretty much all music ends up getting highly compressed by MP3 or AAC codecs anyway. 
Just sayin'
On topic:
I recommend to use the S/PDIF output (and input for reamping) if your interface has the option.
Even if you're perfectly fine with the analog main outputs, it's such a helpful thing to have e.g. dry and wet tracks recorded separately as well, plus the DI of course. Make use of the outputs you're given and enjoy the extra flexibility when mixing your recordings.
I recommend making music instead of plenty research into a (currently) obsolete topic given the fact that nowadays pretty much all music ends up getting highly compressed by MP3 or AAC codecs anyway. Just sayin'
Fair enough, but if one wants to express an opinion on the subject of digital audio, sample rates, bit rates and latency, understanding how it all works is pretty useful
I also think that this moment in history will be regarded by future generations as a dark time, with regards to music and audio quality. As internet speeds, server space and streaming bandwidth increase, it'll only be a matter of time before lossy audio formats are made redundant. Imagine then that you wrote and recorded a masterpiece, only to have mastered it as a crappy mp3 for future generations to enjoy 
.
understanding how it all works is pretty useful
Agree 100% on that ... but I'm not sure if every musician or even audio engineer needs to get into the last bit of detail in understanding. As a stupid example, I very much doubt that many of our music heroes from back in the day (musicians and audio engineers) were able to draw a detailed block diagram of a 2" tape, 24 track Studer, Otari or MCI tape machine let alone even more details of the circuit boards.
What I'm trying to say ... basic knowledge and understanding is important but there's little reason to clutter your brain with the complete understanding of every detail. If it sounds good, it is good.
Listen to really old recordings of classical masterpieces like e.g. Mahler symphonies directed by Otto Klemperer. Astounding music and no noise level or in very old recordings even the lack of stereo will diminish the joy to listen. Technical perfection does nothing to musical expression.
Recorded music will never be 100% "true" to the live experience or the sound that had been crafted and heard in the studio, just because of the lack of the same room and speakers. I'm sure you get my point.
At the end of day we want to make music, play our instruments, get something done. Digital Audio has made it so much easier to record and produce decent quality, even in bedroom studios. We should be careful not to loose contact with the actual music over technical nitpicking.
I can't listen to most of the 90s recordings even though there were some good music. They must have record it at 16 bits before they compressed it to hell.
What still puzzles me today is that I hear that many mastering houses use analog gear in the final mastering stage so sample rate doesn't really matter, because it's recorded from analog into the selected sample rate. This makes me wonder whether, even today, digital is still lacking in some areas.
The start of the 90s saw mixing engineers mixing to DAT and then mastering engineers mastering digitally. Most guitar-based music was still recorded to tape, though. For me, the problems started to arise when the loudness wars began, somewhere around '93-'94 and culminating with the Death Magnetic debacle, where masters became louder and louder by decreasing the dynamic range through a combination of high-pass filtering, soft clipping and brick wall limiting. Even at the height of the loudness wars, analogue outboard gear was still commonplace in mastering studios (mainly EQs and compressors), though digital stereo bounces of the mix or sometimes stems were and are still the normal practice. The reason that analogue gear is still so popular and even plugins try to replicate the classic pieces of equipment is that digital is too accurate, too clean. Our ears have become used to the subtle harmonic distortion, channel bleed and yes even the noise and hiss of analogue that once these things are taken away, recordings can sound sterile, or lacking warmth. It's the reason why vinyl is still popular, even though it's a much less accurate representation of the actual recordings than a CD or 44.1kHz 16-bit wav file. The format colours the music in a pleasing, 'musical' way.
Edit : the reason why there were no loudness wars before CDs is due to the accuracy and reliability of the digital medium. Too many or too few dynamics on a record, and the needle will skip. On a cassette, you have to be above the (relatively very high) noise floor, yet not overly push the tape into compression and distortion. With digital, 0 dBFS is 0 and anything up to that can be reproduced perfectly, time after time. 16-bit has a theoretical dynamic range of about 96 dB, though real-world background noise levels will mean you probably will only enjoy about 80 dB before the dog barking two houses away will drown out the quietest sounds. Funny then, that the loudest CDs probably had an average dynamic range of around 3 dB! I'm looking at you, Metallica!
Well-summed-up, Sammy.
Another thing that could be relevant to Dean's observation is that convertors sounded harsher back then. Prior to the loudness wars, LRB released what to my ears is to this day the epitome of ruining a talented band, producer and mix-engineer's efforts - "Playing to Win".
This master is instantly-run-a-mile harsh, and is all-but unlistenable to me. Such a shame. It does, however, provide an extreme example of what I'm talking about.
Dean_R said:
QuoteDVD audio is 48kzh and I think most people will hear a noticeable difference as I do.
DVDA maximum rates without Meridian Lossless Packing is 48 khz for 5.1 However, you can do 4.0 at 96khz even without MLP.
DVDA can be sampled at 44.1, 48, 88.2, 96, (multichannel) and 192 (2 channel) khz rates with MLP.
You can use S/PDIF output on the Kemper with some audio interfaces which have sample rate conversion available on their S/PDIF inputs. My UA Apollo has this functionality. But the KPA must remain the master for that digital connection. SRC is also not available on the interface's input, so re-amping via the S/PDIF connection is not possible. The best solution is to use a separate sample rate converter.
