By far most passive acoustic systems are linear and time invariant. As such, they don't create frequencies not present in the original signal.
Passive systems that aren't linear are typically of the snaring/clacking variety. They either create overtones to existing frequencies, or they have their own strong resonance but are triggered by external signals.
There are two answers to your question. Simple answer has been given by ObscureRobot here. I only add to make sure your source already completely summed to mono, otherwise the right channel will be dropped, as explained below.
The right channel on all five stereo jacks are connected to literally nothing (a waste if you ask me), but ...
If you're are used to the Dynaudio BM15a, I'd try the BM6aII or a second hand mkI.
Regarding a converter/audio interface, I'd advice the RME Fireface UCX. I own and
use the UC and it's really nice. The UCX is a recent update and can also be used
on an iPad(2,3,4) (for a mobile setup, it could be handy).
But it largely depends on your budget..
If you have ProTools (and understandably a hardware unit to plug into), why not just feed in the RCAs, create a 96k/24bit session, create an Aux and an Audio Track, bus them so that they feed from the Aux to the Audio Track, record to the track (using the Aux beforehand to set your input level). Do any RTAS EQ or limiting afterward, then bounce to 320kbps ...
The splitter cable you're using combines the 2 stereo inputs into the 3 pin XLR connector, putting the left channel in pin 2 and the right in pin 3, or the other way around. You can only use that type of connection with a system that specifically uses XLR type connectors for stereo signals.
That's not the case with normal performance and PA devices, where ...
ffmpeg -i original-file.wav -f u8 -acodec pcm_u8 8bit-file.pcm
See also here.
It's not comma-separated but binary, but that's definitely the more sensible choice – especially for a board with very limited bandwidth.
Do this in two steps. First generate the RAW file (in audacity, there is an option to export to headerless RAW audio file under Export -> Export Audio). Make sure you export to 8-bit signed or unsigned values.
Then, using a linux tool called xxd, you convert the binary file into a c header file(I suppose you could do this with any hex viewer but I haven't ...
The ADC (Analog-to-digital converter) and the DAC (Digital-to-analog converter are the gateways between the real analog world of electrons and the digital binary world; While it is possible that some converters are controllable via firmware/software and thus conceptually are upgradeable, it fundamentally imperative that some sort of hardware exist that ...
Your title question is different to the main question you ask in the body of your post, so you may want to change it a little! However, in response to your main question:-
DACs, whether external audio interfaces, or the internal audio chip in your cellphone, are just like any other piece of hardware; their performance can be tweaked and (possibly) improved ...
As leftaroundabout said, use ffmpeg (or avconv - the one is forked off the other).
It is command line based, which makes it useful for batch operations.
On my Linux box the batch command would be something similar to:
for FILE in *.mp4
do ffmpeg -i $FILE $FILE.mp3
Use the -c:a copy option if you just want to strip away video and use the audio codec as-...
Monoprice should have everything you need:
1/4" mono to 3.5mm Stereo, ideal for headphones.
1/4" mono to 1/4" stereo, good for headphones if your headphones only have a 1/4" plug
1/4" mono splitter
AJ Henderson brings up a good issue in the comments: impedance issues can't and won't be addressed by splitters alone.
Check out the Prism Sound interfaces. I use the Orpheus unit myself and it sounds amazing. They now make a half sized Orpheus called the Lyra, that would be my pick for you if you're after the best D/A - http://www.prismsound.com/music_recording/products_subs/lyra/lyra_home.php