Super NES Digital Audio Mod

the cable may physically be able to transmit that much data, but even 5.1 lossless is beyond coax/spdif in actual application

we already have spc's. i can see someone doing the mod for fun, or for actual game playing, but you're gonna have a hard time matching an spc (which is ripped directly from the cart)

 

You've got that backwards. SPCs have a hard time matching this, since they're emulations and this is the real deal.

 

Absolutely. Where S/PDIF is used in a professional environment, it'll more than likely be via phono jacks, not optical. Actually, S/PDIF via co-ax is limited to 10m, whereas optical is only meant to be used with cables up to about 6m.

The clue is in the name TOSLINK. Here in the UK, toss means rubbish!

 

spc's are sound information ripped directly from a cart. in fact, they can be played back on real hardware if you have a flash cart because the files are just instructions for the sound chip.

just because something is emulated does not automatically make it inferior to the original. i'm willing to bet in a blind test an spc is either going to be equal or better than a wav form an actual console. remember you're dealing with w/e noise or interference (jitter maybe?) is coming from the console, then it's a question of your sound card's quality in making the recording.

 

Define "better". Better, for me, is more accurate. You can't get more accurate than the real hardware. If I wanted my SNES music to be tarted up there's no shortage of covers and remixes out there.

It's a digital signal, if interference is making any difference you're doing something very wrong. I can't speak to jitter, that's outside of my experience really.

Emulated SPC playback has the same issue, i.e. it's only as good as your playback hardware.

 

x

 

I mean both. At some point you've got to hear it and the hardware that facilitates that is pretty important

 

SPDIF is meant to be expanded as everything but STEREO audio is unsupported. Kludging on 7.1 wouldn't be that different than anything else not on the spec.

OT: the cable isn't just able to transmit vastly oversampled 7.1 (<30 Mbit = 30 MHz uncompressed digital signal), everyday coax is good for ~300 MHz which with QAM256 (52 Mbit/chan) is good for 200+ MiB/s of data.

If you're playing back via hardware the SPDIF rip WILL match the DSP's output. And you don't have to spend hours-days hacking game code or weeks-years writing an emulator for an unemulated/poorly emulated system (such as most arcade games). As for emulated SPC playback, it has a hard time matching the actual DSP output since even with a bit perfect DSP (I'm not sure emulation is there yet) the poor SPC emulation will cause phase errors in the audio. Obviously these errors are inaudible but they create bitstreams far less accurate than randomly cycling a real console. If you were only going by what's audible formats like VGM which do not emulate a CPU at all (they just contain periodic register writes) would be entirely sufficient.

I'm sure you know in theory a SPC and DSP can be for all intents and purposes perfectly emulated (the jitter will be in the order of nanoseconds from propagation delay inherent in all circuits and there will be unpredictable phase differences between two different oscillators as there is in real hardware). In that respect every time you start up the console you could have a slightly different bitstream of the same music track. Also the analog filtering can be emulated to the component's ideal value. All this is not a reality as historically emulators have poor instruction timing and drop extraneous read/write cycles (only one or two emulators emulate a system to the clock cycle much less to the clock edge, much less logic gate propagation on an asynchronous output signal XD).

Thrifty and very technical pros will be using SPDIF, big pros should be using AES (XLR connector or something) since it can do a lot more apparently with sub data and probably all the studio software is for it.

Of course. It won't have unintended analog filtering or sample aliasing and it will have perfect dynamic range.

Jitter sounds like audiophile bullshit. Clock recovery should be very accurate on either interface. Any receiver should be able to handle 32 kHz since it's below the spec.

 

actraiser filmoa

actraiser filmoa

one of these was captured from snes via spdif. the other is an spc.

 

Oh absolutely, AES/EBU is better, which is why I was careful to say WHERE it is used by pros However, there are some items (usually semi-pro or older equipment) that only use S/PDIF.

mr. newbie, your links don't work.

 

I listened to them and neither was spectacular. (1) is clearly the emulated SPC since it's 44.1 kHz which means the DSP output was interpolated, where (2) is 32 kHz. (2) was quieter, but I suspect that's the actual waveform and (1) has inaccurate DSP mixer emulation or a gain stage. Both of them would sound better if they were sent (via SPDIF lol) to a receiver instead of being (FURTHER) interpolated and attenuated/amplified (and possibly mixed with any other faint audio or microphone noise) by the operating system's sound mixer for playback.

 

my bad about the links, they should work now

1. was actually the spdif output from this site

2. was converted from spc to flac via foobar.

my computer is connected to my receiver via spdif and i cannot hear any difference whatsoever. that's actually the reason i posted them.

 

The S/PDIF Actraiser recording was done by me and compared to the emulated recording so does it sound sharper and i don't think it is because the recordings is at 44.1khz... it's more because of that the emulator is emulating/simulating the analog filtering that happens after tha DAC when the sound passes thru caps and resistors.

Maybe a new thread should be created as Zedrein might not be interested in "VS" discussions?

 

So the actual 32 kHz SPDIF rip was interpolated into 44.1 kHz during compression while the emulator output simulates analog filtering and is interpolated is still 32 kHz? XD this is not a very good test.

How about with headphones? I can notice the difference with extremely poor netbook speakers. That wasn't what I meant though about being hooked up to a receiver; unless you have custom drivers for your sound card you aren't actually listening to the raw PCM, there is a single sample rate and bit depth which the OS resamples all audio to. So with a computer it doesn't really make a difference if you use a SPDIF receiver because it's not being clocked at 32 kHz anyway, the only improvement is that it's a digital signal longer.

 

I always assumed toslink was supposed to be pronounced tozz-link.

 

x

 

OS resampling means fuckall? How do you know Win 7 doesn't use nearest-neighbor or linear interpolation? Or that your driver doesn't have an auto-equalizer to restore dynamic range compression / other crap you can't turn off? I'm not being an audiophile, just real; for a good listening test raw 32 kHz emulated SPC output should be compared to the real thing.

And as far as I was aware we WERE talking about nitpicking lossless waveforms, not what is necessary for casual listening. With this example however there is a difference, not just in dynamic range but tonally.

As things are now, both the SPC and DSP were RE'd through black boxing so yes, we know SPC are emulated well, but as emulator DSP output doesn't match the hardware DSP output barring output resampling, amplification and filtering so they aren't emulated as correctly as possible. The difference is for the most part inaudible, but the output is freaking digital, it should be correct*.

*except for any slight nondeterministic differences due to out of phase, unideal clock signals

 

x

 

Now i don't remember exactly how i recorded the Actraiser (S/PDIF) recording, but i think it was at 44.1khz and then i changed it to 32khz using Audacity... but it still says it play at 44.1khz? well, anyway at least the speed was corrected.

 

my concern is really whether one could differentiate an spc vs actual output in a blind test. i understand that we could legitimately have pages of fascinating talk about the intricacies of audio recording and playback equipment. I think the only measurement that matters is how similar the songs sound without being able to look at sample rates or anything else.

is this possible? can we contact anyone who has done the mod?

are you sure the audio re sampling is done in every case? i can set foobar to send my 48khz files to my receiver at 48khz (confirmed on my receiver's screen) and my 96khz files at 96khz etc.

i'm not trying to to question question your ears but are you sure your netbook speakers are the best devices to use? i used an onkyo ht-r340. the only headphones i have that work are koss portapros .