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5th March 2011, 08:59 | #13121 | Link | ||
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Rounding 32fp -> 16int is not just "not optimal", it's a straight forward violation of digital processing laws. Yes, ffmpeg is like that. And that is reason enough to not remove possibly better alternatives from ffdshow. My opinion: Make libav default, if you want, but don't remove libdts/liba52, until the libav devs get their act together. Just to balance my (negatively sounding) comment, let me say here that IMHO libav is a *wonderful* open source project. Last edited by madshi; 5th March 2011 at 09:05. |
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5th March 2011, 12:33 | #13122 | Link |
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Agreed, but remember that was you who started it...
Just keep the other decoders, it's as simple as that. Or is there any problem of having them inside ffdshow? Agreed. I also think that the problem should be handled by libavcodec authors, and not patched in ffdshow. |
5th March 2011, 19:18 | #13124 | Link | ||||
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And that's exactly the point. "Fix" it in ffmpeg, and all projects using ffmpeg will benefit for it. Quote:
EDIT: I'll make it output "N/A" instead of 0 though. Last edited by STaRGaZeR; 5th March 2011 at 19:26. |
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5th March 2011, 22:00 | #13126 | Link |
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Regarding the "FLV4 decoding bug", reimar says it's a ffplay bug, not ffmpeg.. and now I wonder: ffplay = ffdshow?
http://roundup.ffmpeg.org/issue2620
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5th March 2011, 23:07 | #13127 | Link | ||
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I will not continue this discussion too. Besides, I'm not one of the devs, so my opinion doesn't really counts. Do what you want, and then I will decide what to use. |
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5th March 2011, 23:13 | #13128 | Link | |
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From what I can see, up until now 4 people have voted against your suggestion to remove liba52/libdts, at least 2 of them being devs, and only 1 person (non-dev) so far seems to support your suggestion. So you should accept that you've been outvoted, at least so far. That said, I very much appreciate you working on ffdshow. Thank you. Last edited by madshi; 5th March 2011 at 23:16. |
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6th March 2011, 01:14 | #13129 | Link | ||
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I'm having trouble with that sentence. 4 people: Qak, yesgrey, madshi, Gleb Egorych. You say 2 are devs. I see no devs in there. I see 2 placebo guys I know very well, and 2 guys that don't give any reason for anything, just like with the recent encoder removal. This is not a public poll, just in case you missed it. Also it doesn't matter where an opinion comes from, a dev or an idiot, the content is what matters. And after I said I didn't want tl;dr posts, I'm finishing one of them for the exact same reason as always. Sigh. I see no reason to continue this conversation unless you provide something to backup your desire of not removing liba52/dts/faad/mad. Output resolution isn't one of them. Put up or shut up, as they say. Last edited by STaRGaZeR; 6th March 2011 at 01:18. |
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6th March 2011, 02:57 | #13131 | Link |
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Overall agreeing with STaRGaZeR here.
People should care if a decoder follows spec or not decoding-wise, instead of looking at some random int/float output setting. In case of at least AC3 and AAC my opinion is that the libav decoders would be at the very least on the level of the current outside decoder libraries if not better, as both have been worked on during the last year+ (The AAC one at the very least should be better, looking at its added and fixed features, which lead to the removal of libfaad from the ffmpeg supported outside libraries). The DTS decoder has also been under at least some kind of development, although I haven't taken as deep look at it as with f.ex. AAC. And if there are bugs, ffmpeg is actually an active project, and looking at how Jumpyshoes as well as elenril got their patches in I'd say it's no longer as impossible as it used to be to get your patches in if they make sense. Of course, just taking a look at the spec, making sure that the decoder actually fails at it, and taking contact with the current maintainer of the given part of libavcodec is never a bad idea and isn't exactly impossible either, given the fact that even I have posted something on the ffmpeg mailing lists -- and gotten a response.
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6th March 2011, 08:43 | #13132 | Link | ||
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Actually no. I do not use liba52. I use libavcodec, patched to floating point output.
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Output resolution was never the problem. The problem is raping the audio data by skipping the required dithering and thus introducing measurable and (with good equipment) audible quantization noise. Which is exactly what (unpatched) libavcodec is doing. I just did. Which would have been your job, btw, before making decisions on which decoders you remove. libavcodec violates fundamental audio processing laws, liba52 does not. Last edited by madshi; 6th March 2011 at 09:15. |
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6th March 2011, 12:05 | #13133 | Link |
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Personally, I don't really care, I already have what I want. I use bitstream under Win7 and under XP I build a chain like this: ffdshow decoder (32fp) > ffdshow output (32fp) > ReClock processing: resampling (32fp AFAIR), volume attenuation (53fp), final stage rounding (24 padded to 32int) > Kernel Streaming. Do I see any sense in 32fp > 16int > 32fp? No. It's stupid and should be fixed by someone who is in right mind. Too bad we lost albain. He is never used to play a *boss* or something.
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6th March 2011, 12:58 | #13134 | Link | |
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As long as any advanced processing is still in 32fp, this conversion costs rather little in quality, and in performance only on systems that can easily afford it. If it's an either-or decision the performance advantage on systems that need it is what makes the current solution win. That said, I do not think it would be that had to make libavcodec support both, it just needs someone who considers it worth the effort to do it. |
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6th March 2011, 13:32 | #13135 | Link |
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A couple of years ago I tried getting patches in to allow floating point output via #define. But my patches were declined with the argument that the audio pipeline would "soon" be rewritten, anyway. Well, maybe I should try again now...
What did happen to albain, btw? |
6th March 2011, 13:54 | #13136 | Link | |||
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6th March 2011, 14:11 | #13137 | Link |
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liba52/libdts/libfaad will only be removed once libavcodec becomes superior. That includes having 32fp support.
@madshi, Can you send me your ffmpeg patches for 32fp ac3/dts (and possibly other formats)?
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6th March 2011, 14:17 | #13139 | Link | |
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Sorry if put words on you that weren't exact. I have no hidden agenda, just gave my opinion. Please do. Maybe you have better luck now. |
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6th March 2011, 14:50 | #13140 | Link | |
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Also, personally I would feel that run-time code path selection would be the better alternative than #defining stuff in the source files before building... Although I guess that would add some "unneeded" fluff into the whole thing. Also, ever since I saw this audiophile herp derp on these threads I've been thinking, don't the specifications for audio decoders specify what is right and what is wrong to do with an encoded audio stream? Or am I one of those happy fellows who has gotten used to standards like H.264 that standardize the decoder to be bit-exact? All this "You should have X instead of Y to have better output" stuff just doesn't make muchos sense, coming from such a background. And if it's something like dithering post-decoding, I don't really get why it can't be decoded with int to get the exact output that was meant to be gotten (given if the specification specifies this -- and I would guess it actually might specify it given the fact that float math's results depend highly on the system/architecture etc.), and then converted to float with dithering for output/filtering/whatever your cat wants to do to it. But maybe lossy audio codecs just make less sense than I thought.
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ffdshow, ffdshow tryouts, ffdshow-mt, ffplay, icl |
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