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Martin Piper
Registered: Nov 2007
Posts: 739 |
Sample streaming from tape
Has anyone done four bit sample streaming from tape before? I'm thinking the time between pulses can be directly translated to four bit volume values as they are read from tape.
Practically useless, except for pure geekyness. But I wanted to know if it's been done, before I do it myself. |
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chatGPZ
Registered: Dec 2001
Posts: 11523 |
I think i have seen something like that back in the days. No idea if that worked with special prepared data though
Why limit it to 4 bit though? |
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Repose
Account closed
Registered: Oct 2010
Posts: 227 |
Interesting idea. I've done something related; just play a tape and listen to the read bit. You can hear 1-bit music this way. |
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chatGPZ
Registered: Dec 2001
Posts: 11523 |
THAT was all over the mags back in the days :) |
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Martin Piper
Registered: Nov 2007
Posts: 739 |
Quote: Interesting idea. I've done something related; just play a tape and listen to the read bit. You can hear 1-bit music this way.
1 bit is different. :)
I'm thinking 4-bit to use the full SID volume range.
Instead of trying to binary encode the 4 bits used for the SID volume, I'm thinking use pulse length to fill the number range from 0 to 15 (4 bits) volume.
This means creating a tape with 16 different pulse lengths basically. |
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tlr
Registered: Sep 2003
Posts: 1814 |
I think ”Pulse” by pixel for the vic-20 uses this technique for music playback somewhere in the loading sequence. |
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Repose
Account closed
Registered: Oct 2010
Posts: 227 |
So the question is, what's the max frequency of pulses you can read? Based on the music I heard, it seems at 1kHz. Your average pulse length would be 8, so your sample rate would only be 128Hz.
Or look at it this way, to get phone quality sample rate (8kHz), you'd need pulses at 64KHz. I'm pretty sure that's not possible.
In fact, your encoding method is not efficient. You face the same tradeoffs as disk encoding. It's only that pulse length can give the wrong value with a slightly different speed, but that's hidden due to the nature of audio.
If you write 1/0 directly with occasional sync, that's most efficient, then the factor is only 4 instead of 8.
This is still a very low sample rate. If you get fancy with ADPCM, you could do at best 2x ratio.
This concept could only play bass/drums at best.
The best system I could imagine is using some DSP to find a combo of SID waveforms as a basis function to recreate the psychoacoustically most relevant frequencies, then the sample rate needs to be more related to how quickly the sound changes, and you could get something like the "mp3" demo.
But for normal coders, basically just d'n'b. |
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Repose
Account closed
Registered: Oct 2010
Posts: 227 |
On 2nd thought, I estimated pulse speed in a very poor way. Looking at typical tape response, a cheap tape deck would do 10kHz, so somewhere below that. That's still pretty poor sampling rate. |
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MagerValp
Registered: Dec 2001
Posts: 1082 |
Use Macbeth's delta encoding instead (0/+1/-1), it only requires three different pulses. |
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Frantic
Registered: Mar 2003
Posts: 1661 |
Vaguely related. Video streaming from tape:
Deep Throat |
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Martin Piper
Registered: Nov 2007
Posts: 739 |
Quote: So the question is, what's the max frequency of pulses you can read? Based on the music I heard, it seems at 1kHz. Your average pulse length would be 8, so your sample rate would only be 128Hz.
Or look at it this way, to get phone quality sample rate (8kHz), you'd need pulses at 64KHz. I'm pretty sure that's not possible.
In fact, your encoding method is not efficient. You face the same tradeoffs as disk encoding. It's only that pulse length can give the wrong value with a slightly different speed, but that's hidden due to the nature of audio.
If you write 1/0 directly with occasional sync, that's most efficient, then the factor is only 4 instead of 8.
This is still a very low sample rate. If you get fancy with ADPCM, you could do at best 2x ratio.
This concept could only play bass/drums at best.
The best system I could imagine is using some DSP to find a combo of SID waveforms as a basis function to recreate the psychoacoustically most relevant frequencies, then the sample rate needs to be more related to how quickly the sound changes, and you could get something like the "mp3" demo.
But for normal coders, basically just d'n'b.
Eh? On PAL, for 8KHz I'd need a pulse every ~123 cycles, right? Assuming one tape pulse encodes one SID volume store of 4 bits.
The TAP file format has a resolution of 8 cycles, so each of the 16 SID volume levels would be, in the TAP file, a pulse length of X to X+15. Where X is a lower boundary constant to allow enough cycles to trigger on tape pulse, read the pulse length with a CIA timer, reset the timer, store to SID, loop...
So in reality the tape pulses in cycle terms could be 64 to 192 cycles to map directly to volume levels 0 to 15. Even an upper limit of 192 cycles would, in PAL, give 5131 Hz, right? |
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