Should I upgrade?

[Kane1972]

I produce music as a hobby, but a serious hobby and I make a little cash from my recordings now and again.

I recently bought a Topping E30 for DAC duties when mixing, but I'm still using my aging Firewire TC Impact Twin for recording (ADC). I use external preamps going into the line inputs, although the mic preamps in it are decent.

I keep asking myself if there are any inexpensive converters/interfaces out there that will have a notable improvement over it in terms of ADC predominantly but also DAC, as I also use the DAC section when I send audio out from it for processing through analogue compressors and EQ's (a round trip).

I'd like to send it to Amir for testing, but I can't be without it. Not sure if anyone can tell from the specs if it's worth an upgrade but the last page shows the quoted specs. I know there are definitely cheap interfaces with better specs (Focusrite etc) but will this translate to better sounding recordings? I wish TC still made interfaces.

Thanks

https://www.manualslib.com/manual/532622/Tc-Electronic-Impact-Twin.html

 

[sergeauckland]

I don't see better specs translating to better recordings, as the technical specs aren't the limiting factors. Your current interface's microphone amps are a bit noisier than the better modern interfaces, but the 6 or so dB difference in noise will be swamped by the self-noise of the microphones which itself will be swamped by the ambient noise of the recording venue.

Frequency response and distortion of your current interface are already well below any possibility of audible improvement, so my view is that I wouldn't bother to 'upgrade' if your current interface does what you want, as any technical upgrade won't be audible.

I still use a well over 20 year old Digigram audio card and/or a now fairly elderly Lexicon interface for my recordings, and the limitation is much more my skill and the venue than anything technical.

S.

 

[Kane1972]

Thanks for that.

I don't use the mic preamps very often, as I have Buzz Audio MA2.2 as well as others, but they are perfectly decent.

 

[AnalogSteph]

That's a nice and comprehensive set of specs for once. Yes, D/A and A/D dynamic range specs aren't state of the art any more, but still perfectly decent nonetheless. (Looks like they are approaching datasheet performance for the AK4620's ADC, while the DAC side is a few dB noisier and with a hair more distortion, all well within the inaudible range though. Might be moddable.)

This interface won't shave off the last millisecond in latency, but periodic passband ripple is low in return - many interfaces these days default to low latency filters with much more ripple instead.

This interface has a fixed +13 dBu line level input only. It doesn't have any other options like e.g. the various RMEs for further optimization (+4, +13, +19, some +24 dBu). If your studio has no problems accommodating this, not a problem.

Output level is more flexible in level but limited to +12 dBu max. That's a typical prosumer interface level. Again, if your studio has no problems accommodating this, fine.

If you do have to upgrade, I'd go from prosumer straight to pro, maybe an RME ADI-2 fs.

When I hear "analog compressors and EQs", I think "noise". A lot more than what the interface has, potentially, making audio interface performance a moot point. I would make very sure that input levels for those are correct, and research what of these could be replaced by equivalent VST plugins. There is no reason not to do a plain parametric EQ digitally, and in the days of internal oversampling, compressors should have gotten substantially better as well.

I'd take a look at the workflow in general:
Make sure that the DAW / version you're using is not known problematic in terms of processing (there has been the odd bad apple over the years). See how its resampler fares. (If very mediocre, there are some perfectly fine free options. The one in Audacity at highest quality is textbook level, just avoid the somewhat insufficient rect dither and use one of the other options for 16-bit output. The SoX resampler for Foobar2000 is pretty much the same thing and should also be trustworthy.)
Apply compression to individual tracks only unless you very specifically want something else.
If you are mastering yourself, use a nice modern-day oversampling limiter. When picking a mastering level, compare apples to apples - apply volume normalization to all versions (either Foobar2000 ReplayGain / R.128 gain, or same average level). I wouldn't mind seeing more releases with something like -14 dBFS average levels and peak ~0.85 again - in the days of shaped dither, we have plenty of dynamic range to spare on the bottom end that is hardly ever being used. (16/44 audio can hit a little over 100 dB(A) of dynamic range without undue effort. Seems alright for a distribution format to me.)

 

[Kane1972]

That's a nice and comprehensive set of specs for once. Yes, D/A and A/D dynamic range specs aren't state of the art any more, but still perfectly decent nonetheless. (Looks like they are approaching datasheet performance for the AK4620's ADC, while the DAC side is a few dB noisier and with a hair more distortion, all well within the inaudible range though. Might be moddable.)

This interface won't shave off the last millisecond in latency, but periodic passband ripple is low in return - many interfaces these days default to low latency filters with much more ripple instead.

This interface has a fixed +13 dBu line level input only. It doesn't have any other options like e.g. the various RMEs for further optimization (+4, +13, +19, some +24 dBu). If your studio has no problems accommodating this, not a problem.

Output level is more flexible in level but limited to +12 dBu max. That's a typical prosumer interface level. Again, if your studio has no problems accommodating this, fine.

If you do have to upgrade, I'd go from prosumer straight to pro, maybe an RME ADI-2 fs.

When I hear "analog compressors and EQs", I think "noise". A lot more than what the interface has, potentially, making audio interface performance a moot point. I would make very sure that input levels for those are correct, and research what of these could be replaced by equivalent VST plugins. There is no reason not to do a plain parametric EQ digitally, and in the days of internal oversampling, compressors should have gotten substantially better as well.

I'd take a look at the workflow in general:
Make sure that the DAW / version you're using is not known problematic in terms of processing (there has been the odd bad apple over the years). See how its resampler fares. (If very mediocre, there are some perfectly fine free options. The one in Audacity at highest quality is textbook level, just avoid the somewhat insufficient rect dither and use one of the other options for 16-bit output. The SoX resampler for Foobar2000 is pretty much the same thing and should also be trustworthy.)
Apply compression to individual tracks only unless you very specifically want something else.
If you are mastering yourself, use a nice modern-day oversampling limiter. When picking a mastering level, compare apples to apples - apply volume normalization to all versions (either Foobar2000 ReplayGain / R.128 gain, or same average level). I wouldn't mind seeing more releases with something like -14 dBFS average levels and peak ~0.85 again - in the days of shaped dither, we have plenty of dynamic range to spare on the bottom end that is hardly ever being used. (16/44 audio can hit a little over 100 dB(A) of dynamic range without undue effort. Seems alright for a distribution format to me.)

Thanks for taking the time to give all that insight and advice. I mainly use VSTs (AU's) in the box, but as I have some nice outboard, my intention is to simply process a few busses through my Urie 1178 or ADR Compex and some tube compressors etc. I make music influenced by the 70's to early 80's, so a little extra noise doesn't worry me to get some authentic colouration.

I will look into the input and output level aspect. I've not noticed any issues when I record using my outboard preamps into a compressor, but I'd not really given that much thought.

 

[dfuller]

If it was me, I'd upgrade just because with a device that old unless it's RME the chances of drivers still being updated are so beyond slim.

 

[Kane1972]

If it was me, I'd upgrade just because with a device that old unless it's RME the chances of drivers still being updated are so beyond slim.

I use it with a Mac, so no drivers necessary.

 

[AnalogSteph]

I make music influenced by the 70's to early 80's, so a little extra noise doesn't worry me to get some authentic colouration.

I get what you mean.

Compared to genuine 1980s gear, your interface is miles better. Just take a look at the specifications for an early '80s Sony PCM1630:
Sampling frequency: 44.1kHz or 44.056kHz
Frequency response: 20HZ to 20 kHZ (+0.5/-1.0 dB)
Dynamic range: More than 90dB
Harmonic Distortion: less than 0.05% (at reference input level = +4 dBu)
Maximum input level: +24 dBu

A number of CDs were mastered on this guy at the time, including e.g. Peter Gabriel's eponymous #4 (Security) if memory serves. Those still sound pretty good.

For an illustration of how fast things were moving, just over a decade later, performance of the best audio ADC chips was approaching those in the AK4620 (see e.g. CS5389 and CS5390 datasheet, dated 1993 - apparently Deutsche Grammophon used the CS5390 in a composite ADC arrangement while coaxing 96 kHz out of it as well, that must have been bleeding edge in '94). Another decade later, we had gained another 13 dB in dynamic range and 10 dB less distortion, along with quadrupled samplerates (see AK5394A).
Since then we've gained another 7 dB in dynamic range (when combining up to 8 channels) and maybe 2 dB less distortion (e.g. AK5578), and sample rates have gone from ample to ludicrous. The main advancement of recent years has been improved efficiency - you can now make an upper midrange DAC similar to yours with about 1/3 the power consumption compared to types from 15 years ago.