27 October 2015, 14:48 | #61 |
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Read here: https://en.wikipedia.org/wiki/Original_Chip_Set#Audio.
Paula has 4 independent 8 bit DACs where each DAC has a 6 bit analog volume control which can also be modulated. When the 8 bit in the DAC is combined with 6 bit from the volume control, you get something like 14 bit depending on how quickly the volume control can be modulated. I would also reckon that you could combine the two channels on each side so instead of 4x14 bit, you could get something close to 2x16 bit. |
27 October 2015, 14:58 | #62 |
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For playing OctaMed Soundstudio sound you can see the source from Aminet though the rights to use it directly rather than in the original .library form may be dubious now that the rights to OMed SS have been bought out.
Also, the way that 14-bit sounds work is by playing the low 6 bits through a second voice at a volume of 1. Stereo is accomplished by doing this through the other two voices. This means that you can play one voice of 14-bit stereo sound without CPU mixing at all. Anything beyond this is either CPU or added hardware. |
27 October 2015, 15:38 | #63 |
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27 October 2015, 15:46 | #64 |
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I was talking about a 14-bit sample but if I had been talking about a 16-bit sample you'd have been right with your correction.
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27 October 2015, 20:11 | #65 | |
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Quote:
14bit is done by using two channels, one at maximum volume (64) and the other at minimum nonzero volume (1). Internally (in memory and CPU) it's actually 16bit that's used. The Amiga DAC do the volume by being muted (turned off) during (64-n) clock ticks out of 64. Then the filtering hardware removes the high frequencies and you get proper volume (i hope my explanation was clear enough ?). |
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27 October 2015, 20:15 | #66 | ||
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Quote:
Quote:
Volume 64, not 63 |
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27 October 2015, 23:33 | #67 | |
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Indeed Code:
Agnus/ Read/ Denise/ Register Address Write Paula Function -------- ------- ----- ------- -------- AUDxVOL 0A8 W P Audio channel x volume This register contains the volume setting for audio channel x. Bits 6,5,4,3,2,1,0 specify 65 linear volume levels as shown below. Bit# Use ---- -------------------------- 15-07 Not used 06 Forces volume to max (64 ones, no zeros) 05-00 Sets one of 64 levels (000000=no output (111111=63 1s, one 0) Shows what I know |
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30 December 2015, 14:29 | #68 | |
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Jochen Hippel original routine uses 268 cycles per mixed byte. Optimised code uses 254 (or 246 in non osfriendly way) Digital Mugician 2 original playing routine uses 272cycles per mixed byte, but optimised code only 211 (or 203 in non os friendly way)!!! You can find their assembly code here http://wt.exotica.org.uk/players.html Overally we have to have more mixed bytes per frame to obtain decent quality, so this limits the possible usefulness of this technology. On C64 we could see a progres in music quality all the time, but Amiga go stuck in Protracker era. Protracker is great, but maybe someone could find a way to enhance number of channels with very limited cost of CPU. |
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30 December 2015, 16:50 | #69 | |
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03 January 2016, 17:10 | #70 | |
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This seems like a surprising choice given that the instruments usually selected for mixing are usually percussions and the likes which have mostly constant volume and frequencies. It seems sensible to constrain the trackers to use constant frequencies for mixed voices which should greatly reduce the complexity and time needed for each byte: for n voices, only n adds + one divide (simple table). Is there such a thing on the market? |
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06 January 2016, 17:42 | #71 |
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That's a terrible idea, even with percussion you want a bit of pitch control. Apart from the obvious things like toms and cymbals getting a bit of variance in your drums needs pitch effects.
Even using this to play long sample loops wouldn't work because they all need individual tuning to the master bpm. |
06 January 2016, 20:16 | #72 | |
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If Paula perform sample interleaving then it is possible to increase sample rate by using 2 DAC's and without reprogramming horizontal scan rate, only by proper sample interleaving in memory (accordingly to Your claim - odd samples in first channel, even on second). I can imagine that cost of silicone may justify to have 4 DAC's (multiplier 8*6 even single, working in sample interleaving mode, probably will occupy maybe even more area on die than whole CPU). |
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06 January 2016, 21:18 | #73 | |
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But i'd have thought one of the most popular uses of more channels would be chords. |
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07 January 2016, 00:00 | #74 | |
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PC audio world is sampling fixed so one sampling rate is not a problem - similar approach can be applied on Amiga - some PC audio hw support only one native sampling frequency... Probably only GUS was somehow comparable to Amiga audio HW (similar principle - variable sampling rate). |
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07 January 2016, 10:41 | #75 | ||
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I'm not a musician but from what i've heard, drums don't use many channels in comparison to tones. So the fixed pitch mixing wouldn't be useful. A fixed per-channel volume can bring a few cpu cycles but i don't think it's worth. |
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07 January 2016, 13:26 | #76 | |
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56kHz * 64=3.58MHz - DAC must be updated with this frequency (unless DAC_Vref is controlled by PWM but then DAC must be multiplying type which itself affecting significantly DAC bandwith).
My point is - until someone not map Paula silicone or Paula schematics will be available we can only speculate how many DAC's is on Paula. Block schematics which are available suggest 4 independent DAC's and all channels are combined in analog domain. Quote:
Stacking to 8 bit DAC's in parallel you should be able to receive 9 bit resolution (if there is no sample interleaving), if there is sample interleaving you should be able double sample rate by using 2 channels where one channel play odd samples and second even samples (maybe with 1 byte offset). If some one is able to accept some limitations (2 virtual channels must have same sampling rate + sample rate reduced by half i.e. relatively lofi sound) then 8 channels can be created by interleaving odd samples and interleaving even samples in one channel buffer. |
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07 January 2016, 13:58 | #77 | |||
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If DACs occupy a large die size, using 2 instead of 4 makes sense. If i recall correctly - which is dubious - i've read somewhere that the mixing was done this way - which might have been wrong as well. Quote:
Quote:
Anyway, for 8 channels or more, i'll always prefer using the cpu. And even an unexpanded A1200 can play many channels with acceptable quality ; for example the menu music in superstardust is a 12-channel S3M. |
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07 January 2016, 16:00 | #78 | |||
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So from one side silicone designer is focus on reducing die size but on second he is limited by technology available and expected parameters. Seem that on picture: two large areas are very well visible - one area consist 4 blocks (audio digital part?, audio digital + analog?) second are consist 2 large block utilising roughly 40% of die (flopy + other logic?, audio DAC's?) Based on this picture still nothing is clear... And you can imagine even different approach - analog switches controlled by PWM where they switch between DAC output and analog ground - as such PWM is applied in analog domain... Nothing is clear and after longer picture observation i almost convinced that audio part occupy approx 60% and that there are 4 DAC's and that second large block is probably chipset bus with address decoder + other bus logic. Picture from http://obligement.free.fr/articles/p...microscope.php but real source is here http://www.tinytransistors.net/index...d=31&Itemid=32 Quote:
Quote:
Last edited by pandy71; 07 January 2016 at 16:41. |
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07 January 2016, 16:22 | #79 |
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US Patent 4,829,473 says ""Each of the data registers 29,31 transfers its data into a plurality of audio control counters and registers 33 and 35, which in turn each drive an individual one of the digital to analog (D/A) converters 37,39 of which there are four in all." and FIG 4 shows 4 identical channel blocks and each has single analog out line.
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07 January 2016, 16:24 | #80 | |
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A good routine can do 4ch on bare 68000 and 32ch with interpolation on 68030 - not what i'd call slow. |
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