16 February 2021, 14:54 | #241 |
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Maybe relevant, several times you have:
Code:
lsl.l #7,d1 addq #1,d7 cmp.l d6,d7 blt .sk4_left Code:
eor.l d6,d6 |
16 February 2021, 15:05 | #242 |
Defendit numerus
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16 February 2021, 15:08 | #243 |
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16 February 2021, 15:11 | #244 |
Defendit numerus
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16 February 2021, 15:21 | #245 | |
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Quote:
I kinda like eor tho. |
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16 February 2021, 15:42 | #246 | |
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Use clr on 68020+
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Last edited by Thorham; 16 February 2021 at 15:59. |
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16 February 2021, 16:03 | #247 |
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Incorrect! There's a 23.453443466666% chance that programmer will forget to specify the size and summon a guru. And with moveq there's a 100% chance that programmer will specify the .L size although it's completely pointless.
Interesting... I like Lea, with a single L. |
16 February 2021, 17:02 | #248 |
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16 February 2021, 17:19 | #249 |
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16 February 2021, 17:21 | #250 |
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Oh ok, I see. I was indeed using it on memory in my benchmark test ( clr.l (a0)+ ).
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16 February 2021, 17:34 | #251 |
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Looks like I am going to have to revert to the previous version. Some people are reporting noisy playback on this one, and I think I know what is causing it.
It would appear it isn't possible to fully sync the channels. I can sync them no problem for lower frequencies, for example audxper = 2048 will be synced without any measurable impact. However, somewhere below 256 the syncing doesn't work anymore. It would appear that starting an audio DMA event at cycle 4 or cycle 80 yields the exact same result. I am guessing the same issue persist in larger audxper values, but it isn't audible because the margin of error is higher. So the only thing I can see as a solution is to use the previous method of having the channels unsynchronized and then compensating for that as that method gave the best results. Which is a pity because that code is way more CPU intensive. Apparently the: cycle 0: channel 0 cycle 4: channel 1 cycle 8: channel 2 cycle 12: channel 3 Dynamic is somehow set in hardware. I still don't know why, when I look at the audio diagram, manual writes to audxdat will reset the volume counter and the percounter is always 0 while dma is off and the audio chip has done it's work. So manually setting the audxdat values and then starting the dma before they expire should work. Even just starting the DMA channels should work because even though they don't have a built in volume counter reset, they should reset the volume counters when the next DMA fetch occurs. Yet they somehow don't. |
16 February 2021, 17:50 | #252 | |
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Perhaps there's only one adder multiplexed between the four period counters, so they're updated in turn? (Perhaps in the down cycles between channel 3 and 0 it's also handles SERPER, and maybe a fixed period for the disk subsystem too?) |
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16 February 2021, 18:46 | #253 |
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Perhaps i have one idea for sync channels, which perhaps can works. I dont know if you tested this idea before. You can start/set up to play empty sample (128KB, maybe less) with different periods and maybe different start address and sample length. And when leading channel is looped set only periods for 2,3,4 channels.
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19 February 2021, 10:14 | #254 |
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Well, I think I reached as far as objectivity, goes, I need subjective input on these:
http://s000.tinyupload.com/index.php...57391871838150 I will add detailed explanation about each later, but I really need people who would just listen to it and report which one sounds cleanest. Thanks. |
20 February 2021, 00:47 | #255 |
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True to my tradition, I managed to assemble and post examples with a huge bug in them: the right channel was playing the same thing the left one was, which made a huge noise in all of the samples.
Here are the fixed versions and also the old method for reference: http://s000.tinyupload.com/index.php...88271423456522 The difference between the new and the old method is that the old one had 32bit float as a source in the PC converter, which gave it a headstart, but it also used calculations that might be too complex for a realtime conversion. The new ones can be easily converted on amiga, if they pass, I will write the assembler code for those. Here is what each does: PAL-piano-none.exe -- only compensates for the noise that is added in the phase shift and is playing the audio in a constant slightly above 9bit bitrate PAL-piano-full.exe -- shapes the phase shift noise to be equal and opposite of the 16 bit to 9 bit conversion noise in the hopes that the 2 noise signals cancel each other out and we get to 11 bit quality PAL-piano-thrd.exe -- has a threshold on the added phase shift noise so that the output is 10 bit quality, but in case the phaseshift noise is too audible, this will help with that PAL-piano-half.exe -- reduces the added phase shift noise by 50%, which should also result in 10 bit quality but in a different way compared to the threshold one PAL-piano-old.exe -- using the old method with 64/63 volumes PAL-song-full.exe -- converted the song with full noise shaping just because the piano is boring (at the end of this, random stuff will be played because I didn't quit on time, just ignore that, thx) None of these methods are too computationally complex - except the old one. Last edited by orangespider; 20 February 2021 at 05:16. |
20 February 2021, 11:30 | #256 |
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Sadly they're all still more noisy than traditional calibrated 14 bit.
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20 February 2021, 11:50 | #257 | |
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Also, can you test with a freshly restarted system? If I use different audio sources it messes up the audio sync and I still didn't figure out how to resync it. Once the sync is completely messed up you get 10x the noise you get while the channels are in default sync. |
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20 February 2021, 12:12 | #258 |
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Yes. I actually have two versions of the piano music. One 28 khz and one 44 khz and using traditional calibrated 14 bit they sound equally noisy.
Tried it. Did a reset after each executable and it made no difference. |
20 February 2021, 12:15 | #259 |
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20 February 2021, 12:17 | #260 | |
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If it's not same on all systems, then I won't get away without figuring out a way to sync the channels. |
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