Has anyone found a demodulator for CD4 (Quadraphonic) Vinyl

[steve]

That's pretty complex.
The difficult parts are getting the demodulation to work correctly and then handling the dynamic expansion correctly.

As a proof of concept (but dreadful sound quality) we can demodulate the 30 kHz fm component with something like:

Code: Select all

(setf s
  (mult
    (highpass8
      (lowpass8 s 30000) 
      30000) 
    (hzosc 30000)))

(defun normalize (sig)
  (mult 0.8 sig (/ (peak sig ny:all))))

(multichan-expand #'normalize s)

The result from your demo file is:

CD4-Test.flac

(214.75 KiB) Downloaded 189 times

(I said it was awful )

I'm sure that the demodulation could be improved (though I'm not sure off the top of my head how we would emulate a phase locked loop digitally), but then the dynamic expansion will need to be exactly right in order to achieve correct sum and differencing. I've not read all of the pdf files that you posted, but to get that right there's a lot of parameters that we need to know (both the compression and expansion thresholds, the attack and release times, and the knee radii), and if that was not enough, we also need to maintain the correct phase relationships across the full audio frequency spectrum. If we don't get all of this right the result will (sadly) be garbage.

In short, it is easy to demonstrate the principle, but to produce a CD4 decoder that sounds anything other than complete rubbish is likely to be very difficult.

[Robert J. H.]

Hi Tim,
Steve was so kind to introduce us presently. I'll try to support you in the plug-in development.
Firstly, I must familiarize myself with the cd4 stuff. Unfortunately is none of your provided material accessible for screen readers. Hence, I must do some OCR with them.
It's certainly a interesting project and other topics could profit from the outcome as well (Dolby decoding for example).
Best regards
Robert

[Robert J. H.]

Some more documentation; about a new demodulator 2007/2008.
http://www.johana.com/~johana/dorren/

[steve]

At least the "sum and difference" part is easy (it's about the only part that is)

[Robert J. H.]

At least the "sum and difference" part is easy (it's about the only part that is)

Yeah, but only if the direct signal (with about 40 ms delay) and the demodulated signal can be properly synchronized.
The low-pass and band-pass filters, along with the equalization and compander section produce serious phase shifts. Even on the original device are 3 to 4 knobs to set up the turntable after each individual cartridge change. It is like solving an equation with 10 unknown variables. Ideally, we had both, a record from tim's device and the digitally transferred channels from a quadrophonic turntable, e.g. a test disc.

[Robert J. H.]

Big question, why does Steve's code snippet produce such a horrible sound?
Answer: Better ask, why does it produce sound at all?
The code does an AM demodulation, i.e. the message is coded in the instant amplitude of the 30000 Hz carrier.
CD-4 is modulated in "FM PM and SSB-FM" which is very nasty to demodulate.
Frequency modulation only squeezes and expands the wave-form horizontally, that's why we shouldn't hear anything after an AM demodulation.
The developers have now decided to adapt the FM-signal to the Main (what you hear on a stereo TT) signal's amplitude, always about 19 dB below it (so called CLC - Carrier Level Control).
There's additional noise and the groove in the other wall that modulates the carrier's amplitude further.
FM-demodulation hinges on a signal of equal amplitude throughout. Hence, the first steps to retrieve the original message are:
- Highpass filter at 15500 Hz to get the carrier + side bands.
- Limit the carrier by hard-clipping.

Code: Select all

(clip (mult 10000 carrier) 1)

- Bandpass the result between 20000 and 45000 Hz to remove the square wave harmonics.
- Actual demodulation with phase lock loop

The PLL from the last step is rather hard to implement because it uses a recursive approach.
I've now employed another method that firstly doesn't need a limiter and secondly needs no lowpass update recursively.
It splits the signal into sine and cosine and keeps tracking the instant phase.
It works at the moment only for short periods because the result is gathered in a fixed array.
That's the sound of the left channel from the sample Audio above:

cd4_(l_f-l_b).mp3

After inverse RIAA

(66.54 KiB) Downloaded 399 times

I've also used an inverse RIAA to bring up the bass a bit. Unfortunately, CD-4 uses an own equalization curve + a expander for the noise reduction. I can't read the response curves myself, thus...
By the way, the FM signal has only -31 to 36 dB, instead of -19 dB. I therefore assume that a normal stylus has been used.

[steve]

Big question, why does Steve's code snippet produce such a horrible sound?
Answer: Better ask, why does it produce sound at all?

The snippet first applies a narrow band pass filter at the carrier frequency.
The FM signal frequency has an average frequency equal to the carrier, but is modulated (by the message signal) so that the instantaneous frequency oscillates either side of the carrier frequency.
As the FM signal frequency moves through the narrow filter range, the amplitude will change such that maximum signal occurs when the FM signal frequency matches the filter's centre frequency. In effect, the narrow pass band filter converts from FM to AM.
The resulting AM signal is then demodulated.
It's primitive, but simple code and works as a proof of concept.

[steve]

I've now employed another method that firstly doesn't need a limiter and secondly needs no lowpass update recursively.
It splits the signal into sine and cosine and keeps tracking the instant phase.

Excellent - are you doing that with FFT ?

That's the sound of the left channel from the sample Audio above:
cd4_(l_f-l_b).mp3

Terrific, great progress

[Robert J. H.]

I've now employed another method that firstly doesn't need a limiter and secondly needs no lowpass update recursively.
It splits the signal into sine and cosine and keeps tracking the instant phase.

Excellent - are you doing that with FFT ?

That's the sound of the left channel from the sample Audio above:
cd4_(l_f-l_b).mp3

Terrific, great progress

No, it doesn't work with FFT, but it is related to it.
It is a Discreet Fourier transformation that returns for each sample a complex vector, tuned to the carrier frequency. It is basically what I've written in my PM.

Code: Select all

(setf sig (highpass8 s 15500))
(defun dem (s cf)
  (let* ((ref-osc (osc (hz-to-step cf)))
         (ref-quad  (osc (hz-to-step cf) 1 *table* 90))
         (sn (lowpass8 (mult s ref-osc) 15500))
         (cs (lowpass8 (mult s ref-quad) 15500 )) 
         (out (make-array (truncate len)))
         (c-1 0) (s-1 1))
   (dotimes (i (truncate len)) 
      (let* ((c0 (snd-fetch cs)) (s0 (snd-fetch sn)))
             (setf (aref out i) 
                   (atan (+ (* s-1 c0) (* c-1 s0))
                         (- (* c-1 c0)  (* s-1 s0))))
                   (psetq c-1 c0 s-1 (* -1 s0)))) 
             (setf (aref out 0) 0)
   (snd-from-array 0 *sound-srate* out)))
(setf sig (highpass8 (dem sig 30000) 50))
(mult (/ (peak sig ny:all)) sig)

C0/s0 is the current complex pair and c-1/s-1 is the conjugate of the past one. The atan function returns the instant phase for the pair.
This is normally very slow and therefore replaced with a lookup-table or a curve fitting function.
The code is not yet very refined, it should actually be implemented as object/class.
Please note: it works only with a mono track, stereo gives a nice visual C++ runtime error.

[steve]

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.