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Playing in real-time through digital

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After many ramblings where I complain about (theoretically solid) limitations of digital audio, here's a short description of what I did this week that actually worked for me, without ifs and buts besides some obvious considerations (for me). So. Here's what I did that might interest some people, others may click away, I'm not suggesting full attention of everybody should be in order here!


The main test, which is well valid for synthesizers/keyboards/workstations/digital-pianos after it doesn't fail, is to use a microphone (I use an Audio Technica AT2020, the most sold studio mic in N.A., with an improvised slight spit-filter, a quality mic-cable and a totally cheap boom), send it through the best pre-amp and AD convertor I have at my disposal (a balanced input of the Lexicon MX400), drive the AD convertor clock as stable as I have it at the moment, copy the AD data into a Linux computer with a good (Lexicon Omega) USB interface, up-sample it to a stable 192kHz (192,000 Samples per Second) clock domain with the best medium quality I currently can use, apply slight but correct microphone correction in the 192kS/s domain (with Linux "jack-rack" + Ladspa "plugins" of various kinds, just a couple of dBs here and there according to the microphone manufacturers frequency response measurement graph), then an up-sampler module to the highest sampling frequency of the high quality USB DA-convertor board, being 384 kS/s, which is ground-lifted connected directly (with short wire) to the input of my quality pre-amp+seperation filter and power amps.


So like this:


AT2020 mic -> Lexicon Omega/DBX preamp ("insert output") ->

Lexicon MX400 multi effect input -> digital input of Omega ->

USB computer input (a I7 Linux system running Fedora 17/64 bits) ->

upsampler module -> Jack audio at 192kS/s -> upsampler 384 kS/s ->

USB 2.0 output to "XMOS" digital signal convertor making a 384 I2C signal for the DA convertor ->

full ground lift (with chip, see here for pdf), Quality DA convertor with low noise supply and DC coupled line-output ( PCM5102A based) ->

my DIY preamp based on TI/Burr Brown OPA627 very high quality OpAmps.


The Lexicon digital audio is 24 bits, but only 48kHz clock frequency, so I use these Linux commands to up-sample (with a filter that takes a little delay, and quite some CPU power):



alsa_out -d hw:2 -c 2 -r 384000 -q 2 -p 512 -n 32

alsa_in -d hw:3 -c 2 -r 48000 -q 4 -n 3 &



First the input is upsampled neatly (-q 4 i the highest quality that makes on core pretty busy), connected with the jack-rack (not shown) and the output of that is resampled to 384 with mild "sinc function" related digital filter to connect to the 24 bits output USB.


This is the DIY (it isn't very expensive, but you have to solder some connectors, order a few connection wires, and improvise some wall wart (for instance) to get 5V for the ground-lifted side, and connect that properly:


(see this thread )


The result is mostly devoid of the well known (for me and I'm sure others) sampling artifacts that come through just about any digital system, and are usually easily audible on my big (analog) monitoring system.


In fact, even the feedback echo after a few times sounds natural, and not (much) distorted, up to little under "blasting" output levels, and also when using strong Lexicon effects like studio delay and various types of reverb or DBX compression simulation.


And the result of feeding the left balanced output of a CP4 to the other Lexicon MX400 balanced input (CP4 volume at "one o-clock", MX400 gain control idem), using my latest acoustic piano sound edit, one iteration after the one I've made available for download, and playing and listening through the AD-DA chain was great: mic feedback in order, sound great. That's a first for me, but admitted, I'm critical.




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