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15th May 2011, 14:24 | #3001 | Link |
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That version doesn't help either, the build I did 2 days ago also had this issue. I don't recall which revision it was, but it was maybe d1a6c61.
I am not sure how to roll back changes to compile an older build in git, if you could give me a tip of this, I can do some testing to try find which one broke the sync. This issue isn't to all DTS in MKV though it seems to be somewhat to this file only and maybe others, though 0.25 worked perfectly for this file. Decoded DTS doesn't have the sync issue with this file either. I mostly found bitstreaming to have better sync than decoding, especially so for the build 2 days ago.
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15th May 2011, 14:24 | #3002 | Link |
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I have no issues whatsoever, so i claim your file is broken.
With the latest version, and the A/V sync thing enabled, it'll always match against the timestamps of the incoming data, and make sure it fits. Edit: Try with this older version; http://files.1f0.de/lavf/LAVFilters-...2-g9f7db6b.zip
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LAV Filters - open source ffmpeg based media splitter and decoders Last edited by nevcairiel; 15th May 2011 at 14:38. |
15th May 2011, 14:42 | #3004 | Link |
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You're out of luck then, get rid of the broken file.
Or give me a sample to reproduce the problem. I could revert some of those changes, and when bitstreaming, just trust whatever timestamps the source gives me (like 0.25 did), so if they are messed up, its clearly the files fault. Except, the audio jitter is horrible then, and the graph of EVR-CP looks like crap because the audio timings jitter so much, so its clearly not good. However, with A/V sync thing enabled, i really do not see how there could be sync issues, ever. It'll compare all timings to the input time of the data, and if there is a too bad drift, it'll adjust to that. Its just impossible that my calculated timings cause a 2-3 second desync. Its just not possible. It uses the same timing the decoding would use, it reads the data, and figures out how long it would be - and then increments the clock by that. Thats a perfect match to the source timings, minus the jitter of DTS in MKV, of course.
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LAV Filters - open source ffmpeg based media splitter and decoders Last edited by nevcairiel; 15th May 2011 at 15:06. |
15th May 2011, 15:09 | #3005 | Link |
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fyi, you can go to the github for LAVFsplitter, choose "History" link on the right, just above the file listing,then you can browse the commits and select a commit, then with the "Download" option you can extract that "branch" and build it.
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15th May 2011, 15:13 | #3006 | Link |
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So i have a new theory, maybe its the ffmpeg parser that somehow causes something funny.
Please try this: http://files.1f0.de/lavf/LAVAudio-0.25-15-g19442af.zip This version already includes the ffmpeg parsers, but still the old timing logic.
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15th May 2011, 15:20 | #3007 | Link | ||
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Quote:
Quote:
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15th May 2011, 15:21 | #3008 | Link |
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The latest one i posted earlier should work, maybe all other 0.25 builds too.
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15th May 2011, 15:24 | #3009 | Link |
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Yep, LAVFilters-0.25-55-g31d0dd6 with the AX from LAVAudio-0.25-15-g19442af fixes the issue
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15th May 2011, 15:30 | #3010 | Link | |
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Quote:
If you support IAMStreamSelect in one of the filters involved shouldn't that make stream switching work with most players? Maybe you'll have to make LAV Splitter support IAMStreamSelect and then forward that to the switching filter, cause it might be possible that players expect IAMStreamSelect only from the splitter. |
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15th May 2011, 15:36 | #3011 | Link | ||
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Quote:
I'll draft up a full concept, and post it for commenting somewhere. Quote:
The calculation for the DTS durations was off for that file, not sure why. But i changed it, and it should work better now. I would've kinda hoped that my A/V sync code would've taken care of this, but the timings were drifting apart so fast, that it couldn't keep up. http://files.1f0.de/lavf/LAVFilters-...6-g41ca5a7.zip
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LAV Filters - open source ffmpeg based media splitter and decoders Last edited by nevcairiel; 15th May 2011 at 15:40. |
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15th May 2011, 15:46 | #3012 | Link | |
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Quote:
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15th May 2011, 15:55 | #3013 | Link |
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madFlac uses the official FLAC library / source code and has been working for years with perfect A/V sync, no stability problems and bit perfect output, with all kinds of channel configurations, bitdepths and samplerates. AFAIK, the LAV FLAC decoder uses a different code base (not the "official"/original one) and has not been extensively tested yet. Why would you even consider going away from a pretty much perfectly working decoder? I don't really understand that. There's nothing to gain, from what I can see. From my experience, everytime a different FLAC decoder was (accidently) activated on my PC, I ran into all kinds of playback problems. That also includes LAV Audio Decoder, although it was an older version, the problem might be fixed in the latest version.
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15th May 2011, 16:16 | #3014 | Link |
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Yea, ffmpeg implemented their own FLAC decoder, and i'm not sure if its working perfectly.
There was a problem that caused sync issues in earlier LAV Audio versions, but that was fixed a long while ago, and everything is working good now. Since its a lossless codec, we could compare the PCM output, couldn't we? Anyhow, i don't use FLAC that much, and i have other things to do.
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15th May 2011, 17:15 | #3015 | Link |
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LAV Filters 0.26
Code:
General - The VC2010 runtime is no longer required to be installed LAV Splitter - Further VC-1 in MKV improvements - Added a separate "mpegvideo" (m2v files) option to the Format section - Added a flag for secondary audio streams - Changed the default DTS media type (better compatibility) LAV Audio - Added the ability to use the ArcSoft DTS decoder directly through LAV Audio - Added an automatic algorithm to keep A/V sync stable - Improved Bitstreaming with splitters other then LAV Splitter - Improved timestamp generation when bitstreaming - Fixed LPCM 24-bit Media Types Using the ArcSoft DTS decoder LAV Audio can now directly interface with the ArcSoft DTS decoder. You do no longer need any complicated setup or installation. To use the ArcSoft decoder in LAV Audio, simply get the "dtsdecoderdll.dll" from your TMT3/5 Codecs directory, and put it in the same directory as LAV Audio. Thats all you need to do, no configuration, no hacked checkactivate.dll, no extra filter to be used. When playing a file, you can see if its being used on the "Status" page of the LAV Audio propertys sheet. "dca" is the ffmpeg based decoder, "dts" (and any variations of that) is the ArcSoft decoder. This will allow you full decoding of DTS-HD MA and all other variations of DTS. One thing, though. Do NOT ask where to get the ArcSoft decoder. If you own TMT3 or TMT5, it will be in the TMT Codecs folder. If you do not own it, you should buy it. I respect their work, and so should you! Bitstreaming Improvements Bitstreaming should now work much better with other splitters. Note that there is still one limitation however, TrueHD only works properly if its actually flagged as TrueHD, and the interleaved AC3 was removed. Therefor, it will most likely not work with Haali and the MPC-HC/Gabest splitters. As always, please report any regressions or still persistent bugs directly here, with all the relevant details, and if possible a sample file.
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15th May 2011, 17:24 | #3016 | Link |
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Just an FYI...I tested last night's code with MC16 build 160090 and Star Trek 2009 off the disc. I have LAV Audio decoding everything to PCM. I'm using ReClock at the moment for exclusive mode (plan on playing with MC16's exclusive mode at some point, but, in testing I had sync issues). Star Trek was rock solid. That's a good thing as that's always been one of the "trouble" discs for TrueHD. I'm very pleased! Building 0.26 now and will test it out at some point. Thanks, Nev!!
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15th May 2011, 18:35 | #3018 | Link |
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Yes, but you'd have to test all bitdepths, channel configurations and samplerates. Furthermore you'd have to test whether the decoder works with all containers (that support FLAC) and all splitters and produces perfect A/V sync with all splitters, including seeking etc. Then, there are also extra tricks to support non-standard channel configurations which I'd bet ffmpeg does not implement by itself. Finally, since ffmpeg is constantly being worked on, with FLAC only being one of the less important small parts of ffmpeg, there's always a chance that some future ffmpeg revision might accidently break something without being noticed. FATE tests don't cover everything.
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15th May 2011, 19:31 | #3019 | Link |
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Crazy we can do DTS-HD MA transcoding live now, i guess madshi helped you here with where to hook into thx
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15th May 2011, 19:32 | #3020 | Link |
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Nope, thats all my doing. It actually wasn't that hard to figure out. Just some leg-work, testing and debugging to figure out the arguments.
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