SPDIF vs. Balanced, Which = Better Quality?

They are two very different ways of transmitting audio signals.

SPDIF is digital. 'Balanced' is analog.

If you're going to be doing any external processing of the Fantom signal -- compression, EQ, etc. -- or put it thru a patch bay then you're best choice would be analog or 'balanced'.

If not, SPDIF will give you a pure digital signal warts and all.


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all i am doing is recording my keyboard into my computer software to make songs, i dont run any analog hardware. I just want to know which one will provide the absolute best quality. what do you mean warts and all?
 
If there are anomalies in the patch like tuning or envelope problems they will go in to your sound card unmodified. That's what 'warts and all' means.

If you want to hear your Fantom just as it is, take the SPDIF output into the SPDIF input of your soundcard.


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if you had the ability to transfer the sound in any way you wanted, digital, analog, whatever. which one would you choose for optimal end result?
 
darkecho said:
if you had the ability to transfer the sound in any way you wanted, digital, analog, whatever. which one would you choose for optimal end result?
Click to expand...
It's not really possible to say definitively, but, in general, I would choose digital. The only counter case I can think of off hand is where the D/A conversion of the source imparted a colour that was integral to the sound of the synth, or one that I liked.
 
Firstly check the Roland is sampling at the rate you require. If it only does 16/44 & you want 24/96, go analog into the emu or 44.

The second thing is the quality of the convertors - are the Roland's better or worse than the cards?
 
In your situation, it's probably best to use the S/PDIF output, but I'd suggest that you record some of your favorite patches through both interfaces and carefully A/B them. Specs are one thing, but at the end of the day.........
 
ssscientist said:
If you want to hear your Fantom just as it is, take the SPDIF output into the SPDIF input of your soundcard.
Click to expand...

That sums it up well.

Staying digital will eliminate a complete DA to AD conversion cycle. So you'll hear exactly what's in the synth, and you won't hear any converters.

I should add that the converters are actually pretty decent, and there won't be a huge difference between the analog and the SPDIF signals. However, for the cleanest and most accurate reproduction of what's in the synth, use the digital output.
 
How Do I find out the bitrate/sample rate?

It says in the owners manual it says "Data Format: 16 bit linear (Filetype Wav. Aiff.)"
and it says "Sampling Frequency: 44.1kHz Fixed" but these are under the "Sample Section" and because this keyboard is a sampler I have a feeling this is just the max that the keybaord will store samples you put onto it...

I cant find anywhere what its internal bitrate/sampling rate is...
 
Ok i found out what the sampling rate is via a forum about my keyboard but i dont quite understand the post here it is:

"Yes, like phran said, the S/PDIF output from the Fantom-X is


24-bit word size
44.1 kHz sample rate
2 channel (stereo)

It should be noted that the data from the S/PDIF comes directly from the Fantom-X's WX DSP so, assuming your external D/A hardware is capable, you could get the theoretical maximum S/N for a 24-bit signal of 144 dB. Of course, you wouldn't get this S/N since all the samples in ROM are probably not 24-bit samples and the internal sampler only samples at 16-bit resolution, but you'd still get as good or better S/N from the S/PDIF compared to analog. "

so hes saying that i wnt nget any better quality than 24/44.1 out of my keyboard? and even at that heis saying that the keyboard has 16bit samples.

but how does this matter between analog versus digital, the same 16 bit sample will be used via analog channels too right?

should I have my bitrate/sampling rate set to whatever the highest is from my gear?

cause currently im recording at 32bit/44.1 but my keyboard doesnt output higher than 24 (and really 16 i guess) and my Delta44 does a max of 24/96

so does that mean that my computer is only capable of taking in a signal maximum of 24/96? and because of my interface being the ultimate bottleneck, should I have my software processing in 32bit? or is that just causing uneccessary work for the computer.

thank you anyone who can decipher my questions!!!!!
 
Using the maximum sample rate is not exactly a 'bottleneck'.

24 bit/96kHz is substantially over the CD sample rate which is 16bit/44.1kHz, so any music you produce at the higher rate will need to be dithered down to the CD rate to be reproduced.

Some people claim there's a huge difference in using 96kHz sample rates, but I use 24 bit 44.1kHz for most of the music I do. I can hear the extra depth, extra bass, extra crunch and extra all around fidelity those crucial 8 bits bring and I find that to be MUCH more important than the fidelity improvements you get from using a 96kHz sample rate.

Try analog, then the various digital configurations available to you and see if you can actually hear the difference. Because at the end of the day, that's what counts.

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ok i got a digital card..

heres what i conclude for anyone reading this...

Digital picks up much less interference noise over analog, balanced or unbalanced...

other than that it seems to be pretty much the same sound, maybe the digital sounds a little brighter/clearer but that could be my imagination..

so i would say, Digital offers higher quality, if you have the ability, go digital, otherwise, just go balanced.
 
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