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I'm using a Sony SDP EP9ES for capture. I also have a small Yamaha demodulator but I ran into the same issue, so I just used that one when watching AC-3 discs, rather than capturing them.
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I have a Pioneer RFD-1, and I also have to re-start the capture after a side change, pretty annoying. Maybe your Sony has some additional circuitry to prevent the signal from dropping (which would also be annoying on some receivers, I know my SPDIF-modded SNES causes the audio to drop out on my Onkyo receiver when the sound chip isn't producing audio), not sure.
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Does this mean you don't even need an AC3-demodulator anymore and you can simply capture the AC3 as WAV and decode it later, just like it has to be done for DTS?
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Yeap, for offline processing at least. The original author of the code was looking to use it in an FPGA device for real-time decoding/playback, but looks like he may have moved on. The VHDL files are there however for someone else implementing a hardware solution someday.
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So an analog capture of the noise and then convert that to digital data? That’s wild. You’d need a non AC3 player to do that as the analog out would be muted on an AC3 unit.
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Oh wow, I think I already have all my AC3 LDs captured but that's awesome.
I hope that real-time playback is possible someday as well.
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Yes the AC-3 RF output is the raw FM signal from the player which the demodulator filters and converts to a bitstream signal. I think that using the right channel analogue audio output (assuming it isn't muted) would result in the signal being FM demodulated which could corrupt the data, which is QPSK modulated, but couldn't say for sure. It's certainly an interesting development!
Borisanddoris, going back to your initial problem, could you elaborate? It sounds like you're saying when you run a LD through your demodulator into your A/V receiver it locks instantly whereas when running into a capture device there's a delay?