| |
Martin Piper
Registered: Nov 2007
Posts: 698 |
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. |
|
... 36 posts hidden. Click here to view all posts....
|
| |
Martin Piper
Registered: Nov 2007
Posts: 698 |
Improved sample rate. Same instructions and links as post #16 above...
Download this PRG: https://github.com/martinpiper/C64Public/raw/master/IRQTape/Tap..
Download this TAP file: https://github.com/martinpiper/C64Public/raw/master/IRQTape/vic..
You should see narrow colour bars indicating the volume this time. This time it's using "Tom's Diner". |
| |
tlr
Registered: Sep 2003
Posts: 1762 |
Quoting KrillQuoting tlrI would think that higher frequency content has lower amplitude in general, but feel free to experiment for optimum results. I wouldn't. :)
Gutsfeeling says there's some kind of optimum to be achieved by having a special switch-token pulse length to flip between the two (apparently concurrent) goals, though.
My assumption is based on the fact that the spectrum of typical songs falls off towards high frequencies, thus the changes of higher frequency would statistically have smaller steps.
This can of course be tested with some actual data.
Quoting KrillAnd on another thought... differential encoding alone would have to rely on lossless/error-free encoding, lest it degrade quickly (without some intermediate absolute literals).
yes, errors would be handle badly. |
| |
tlr
Registered: Sep 2003
Posts: 1762 |
Quote: I might try, delta pulse lengths for +1 and -1, and no change could be any longer pulse, which would allow consecutive unchanged samples to just use one pulse of a long length instead of having to encode "0" often.
Then every 16/32/64 samples read the full 4 bit token to reset any delta read errors.
My idea was to not only encode +1 and -1, but 0, +1, -1, +2, -2 and perhaps more as single pulses. The reasoning behind this is that the difference between the pulses need not be a multiple of the shortest one. |
| |
Hoogo
Registered: Jun 2002
Posts: 103 |
Quote: Quoting Martin PiperAccuracy would be improved by using an IRQ instead of polling bit 4 on $dc0d. Main thread running over a thousand NOPs, interrupt kicks in, does stuff, and jumps back to start of NOP desert.
Did any tape loaders use this to improve accuracy (= increase bit rate)? =)
At least to test jitter. Iirc, 95% Jitter is surprisingly small, within a 6 cycle window. But the other 5% are far off, so you have to increase your pulse lengths. And if you want to write with a PC, you will stick to 44100 anyways.
But why is there so much noise in this sample?? |
| |
Oswald
Registered: Apr 2002
Posts: 5074 |
Quote: Improved sample rate. Same instructions and links as post #16 above...
Download this PRG: https://github.com/martinpiper/C64Public/raw/master/IRQTape/Tap..
Download this TAP file: https://github.com/martinpiper/C64Public/raw/master/IRQTape/vic..
You should see narrow colour bars indicating the volume this time. This time it's using "Tom's Diner".
that sounds very good |
| |
Martin Piper
Registered: Nov 2007
Posts: 698 |
Quote: At least to test jitter. Iirc, 95% Jitter is surprisingly small, within a 6 cycle window. But the other 5% are far off, so you have to increase your pulse lengths. And if you want to write with a PC, you will stick to 44100 anyways.
But why is there so much noise in this sample??
Downloaded from YouTube. Volume increased to increase the range of volume bits used. Also the quantization is accentuating rumble and hiss. |
| |
tlr
Registered: Sep 2003
Posts: 1762 |
Quote: Downloaded from YouTube. Volume increased to increase the range of volume bits used. Also the quantization is accentuating rumble and hiss.
With this variable sample rate, how do you do your downsampling/alias filtering of the source material? |
| |
Martin Piper
Registered: Nov 2007
Posts: 698 |
Quote: With this variable sample rate, how do you do your downsampling/alias filtering of the source material?
I don't, which obviously doesn't help at all. :)
New build using IRQ (and a NOP fill) is available at the same links as post #16. The sound quality is improved because there is less jitter in the pulse detection. I was also able to add timer underflow detection which clamps the volume appropriately.
Anyone recognise the music? :) |
| |
Martin Piper
Registered: Nov 2007
Posts: 698 |
Hmm, so at the moment the 16 volume values are separated by 16 cycles, which is 256 cycles in total. A base value or 56 cycles is added. This means an early pulse or longer pulse (due to tape speed or wobble) has the high timer bit set.
The timer is setup to count cycles to measure the pulse length... The read of the high timer value and the associated branch adds precious cycles to the IRQ, which reduces the overall maximum volume value rate from the tape.
However... what if the one timer was setup to count not cycles but "cycles divided by 2" instead. This would allow the middle 128 value range of the timer to represent the desired pulses and anything outside that range to represent undesired values and clamp the volume appropriately.
This would mean the timer is doing the underflow check, not the CPU, which would save precious cycles in the IRQ and allow a higher sample rate. |
| |
Hoogo
Registered: Jun 2002
Posts: 103 |
You can't have less than 2 cycles of jitter, so a long column of INX can do the counting instead. Just add a Jmp * at the end. |
| Previous - 1 | 2 | 3 | 4 | 5 - Next |
