Has anyone found a demodulator for CD4 (Quadraphonic) Vinyl

[Robert J. H.]

Cool.
You lost me here:
Where did this equation come from?

Code: Select all

  (atan (+ (* s-1 c0) (* c-1 s0))
        (- (* c-1 c0)  (* s-1 s0)))
(psetq c-1 c0 s-1 (* -1 s0))

This would probably be handled OO.

oo?
Well, the atan function works either with real or complex numbers (unfortunately the only nyquist function that can do this).
It returns a angle in radiants (from -pi to pi). the argument is essentially "real divided by imaginary". The phase is not taken from the actual complex number but rather from the past number multiplied by the current one. All inside atan is a multiplication of two complex numbers.
The last line saves the current rex and imx and multiplies imx (the sine term) by -1, this gives the conjugate.
For example: You can take an FFT, multiply all imaginary numbers by -1 and you'll get the conjugate. If you transform this back to the time domain, the whole sound will be reversed (in time).
I think I've confused the terms (although the output is essentially the same with a 90 deg phase shift).
It needs a lot simplification anyway.

[steve]

OO > Object orientated.
Not pretty in Nyquist, but good for processing sounds sample by sample.

[Robert J. H.]

That was my intention anyway, at least if I have to use the 'atan' function.
However, later a lookup table could be great to perform several tasks at once.
I've also implemented the PLL variant - all object-oriented. However, the results are not yet convincing due to the lack of information about the fine tuning.
The only advantage, that I can see so far is the automatic carrier detection. But this could probably also included in the code above as a pre-processing step.
I forgot: There's always a nasty click in the beginning. This is perhaps because of the high-order filters, which need some time to react. Hence it is better to put some dummy sound in front and cut it away in the end.

[steve]

There's always a nasty click in the beginning.

Yes I noticed

The only advantage, that I can see so far is the automatic carrier detection.

I wonder how accurate the specified frequency is? I guess it will be highly dependent on the quality and condition of the turntable. Also, how much the speed/frequency will drift over the course of a record.

[Robert J. H.]

I wonder how accurate the specified frequency is? I guess it will be highly dependent on the quality and condition of the turntable. Also, how much the speed/frequency will drift over the course of a record.

The carrier frequency should be very accurate. The recording has been cut at 11.1 rpm and the carrier is therefore 10 kHz.
I am actually surprised how good the new algorithm extracts the upper signal. From your example code and from my earliest tries, I've expected less more than trash.
One has to keep in mind that the carrier has only -30 to -40 dB and could be therefore very susceptible to noise.
The original stylus is flat and has 4 times more surface contact than an elliptic one. That's where most of the quality gets lost when using commonplace equipement, I fancy.
Without additional audio samples from Tim or whomever, the CD-4 demodulator project is at a dead end, I'm afraid.

[steve]

I'd be interested to see your PLL code if you're happy to post it.

Without additional audio samples from Tim or whomever, the CD-4 demodulator project is at a dead end

That would be a shame, but you've come up with some interesting stuff on FM decoding and we may find other applications for that.

[TimDog73]

Hi Robert,

I can provide more samples from the same equipment and different discs if that's what your requesting?

Unfortunately I don't have access to a CD-4 decoder at this time to provide samples of it's output.

Let me know what you require and I'll do my best.

Best regards
Tim

[Robert J. H.]

That would be great Tim.
There's for one part the delay I am interested in. The base band is normally delayed by about 40 ms. This time is needed to demodulate the 30 kHz FM signal. The difference might best be seen at the very beginning of a recording.
Then, there's the RIAA equalization that must be averaged from a longer piece of music.
I guess that the RIAA inversion is made for the lower band by your turntable automatically but not for the encoded signal of course.
The bad thing is that the expander section of the compander - which is essentially the noise reduction part - is the least known thing to me.
It seems to work in two defined bands, but threshold, ratio attack and release are mere guesswork for the time being.
Once the front and back are aligned, we will probably hear some "pumping" due to this and it should give us some clues.
In conclusion, we need some longer Audio (or a lot of short clips) for the spectrum averaging, some track transitions for the delay alignment, some longer lasting notes for the phase shift profile, in short all that is available.
I've last used a turntable 20 years ago, otherwise I'd purchased a CD-4 test disc. It would make things easier because some tracks have the exact same information stored in the front and back channel.
Maybe there are others out there who want to share some samples too (@Waxcylinder: Peter, wasn't there never a quadrophonic phase in your life? ).
Since this is a educational project, clips of up to 30 sec shouldn't violate any copyright (you know, I am not far from Berne...).
You can upload the samples here, use a public file sharing site or we (Tim, Steve and I) share my CD-4 drop box folder, whatever is the most convenient.
Best regards
RJ

[waxcylinder]

(@Waxcylinder: Peter, wasn't there never a quadrophonic phase in your life? )

I considered it (briefly) but decided I was quite happy with my pair of electrostatic QUADS

Peter.

[TimDog73]

Robert,

Sorry for the delay getting you more samples; I have had a few technical problems but on the plus side I have now managed to capture at 192kHz.

I've uploaded to Dropbox six clips. They include the lead in at the beginning, lead out at the end and a transition between tracks around the middle point of two different disc's.

The FLAC files are 192khz 24bit.
The WAV files are 192kHz 16bit.

Unfortunately I haven't as yet heard from the chap in the USA who said he would provide clips from the calibration disc, needless to say I'll upload them to Drop box as soon as I get them.

Hopefully this will enable you to make a bit more progress?

Regards,
Tim