LAV Filters - DirectShow Media Splitter and Decoders

[VipZ]

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.

[nevcairiel]

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-0.25-22-g9f7db6b.zip

[VipZ]

Nope, that is also out of sync :(

[nevcairiel]

You're out of luck then, get rid of the broken file. :)
Or give me a sample to reproduce the problem. :p

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.

[noee]

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.


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.

[nevcairiel]

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.

[VipZ]

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.

Thanks

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.

Which build should I drop this AX in, or it doesn't matter?

[nevcairiel]

The latest one i posted earlier should work, maybe all other 0.25 builds too.

[VipZ]

The latest one i posted earlier should work, maybe all other 0.25 builds too.

Yep, LAVFilters-0.25-55-g31d0dd6 with the AX from LAVAudio-0.25-15-g19442af fixes the issue :)

[madshi]

Thats where that extra stream switching filter comes in, basically a multiplexer that takes all decoders, and forwards only the active one to the renderer.
Ah, I see, I missed that part, makes sense to me now.

But yeah, its not a viable solution, unless a player fully supports this mode.
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.

[nevcairiel]

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.

That would be possible. Maybe i'll build such a multiplexer in due time. Of course it'll look a bit weird in the graph list, say you have 6 audio tracks, you suddenly have 6 decoders in there...

I'll draft up a full concept, and post it for commenting somewhere.

Yep, LAVFilters-0.25-55-g31d0dd6 with the AX from LAVAudio-0.25-15-g19442af fixes the issue :)

I properly fixed the issue now. Thanks for the sample.
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-0.25-56-g41ca5a7.zip

[VipZ]

I properly fixed the issue now. Thanks for the sample.
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-0.25-56-g41ca5a7.zip

No problem. Thanks, its working good now :)

[madshi]

Is there any reason *not* to use LAV Audio for FLAC?
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.

[nevcairiel]

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. :)

[nevcairiel]

LAV Filters 0.26

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


Download: Installer (both x86/x64) (http://files.1f0.de/lavf/LAVFilters-0.26.exe) -- Zips: 32-bit (http://files.1f0.de/lavf/LAVFilters-0.26.zip) & 64-bit (http://files.1f0.de/lavf/LAVFilters-0.26-x64.zip)

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.

[SamuriHL]

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!!

[Gleb Egorych]

How about that avi crash?
This testing took a bit more time because the bug appeared pretty random. Seems it is fixed :). Tested 0.25-53 and slightly 0.26.

I'll also watch if player hang bug is also fixed with those patches.

Thanks, nevcairiel!

[madshi]

Since its a lossless codec, we could compare the PCM output, couldn't we?
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.

[CruNcher]

Crazy we can do DTS-HD MA transcoding live now, i guess madshi helped you here with where to hook into ;) :D thx

[nevcairiel]

i guess madshi helped you here with where to hook into

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.

[joeydrunk]

Nev, in what scenario would it be better to use arcsofts dts decoder? Anything specific, or is completely a matter of opinion on wich you prefer?

[SamuriHL]

If you want to decode dts-hd ma, it's your only option. Period. Otherwise bit stream.

[Carpo]

glad i just use DTS on my backups, seems dts-hd ma is a pain in the ass :/

[noee]

glad i just use DTS on my backups, seems dts-hd ma is a pain in the ass :/

Au contraire, it's a piece of cake, now. ;)

So smooth, like butter.

[nevcairiel]

Indeed, now its as easy as any other format, you just need a copy of TMT, and drop that dll into the LAV Audio directory. Easy! :)

[Carpo]

well it would be a pita for me to re-rip them to add dts-hd just in case in the future i get a receiver capable of decoding it :p, although it is tempting ;)

[SamuriHL]

Yea I agree...no problem at all now. Drop in a DLL, enjoy fully decoded dts-hd ma.

[SamuriHL]

Decode it on the PC...that's what we're talking about. :-)

[Carpo]

i do, but i only have a set of z5500, just wondering if there is any point

[SamuriHL]

Going forward I would just do the HD audio. Is there any point in your current setup? Probably not. But, if you do upgrade, do you want to have to go back and rerip everything? (Even just anything you rip after this point, I mean). You might as well just throw the HD audio in there and call it good.

[Carpo]

Thats what im thinking about atm, that and i mainly use linux as an htpc, will do a test rip of a bluray and see how it fairs, guessing it should be fine as it should just drop back to plain dts

[SamuriHL]

Right, DTS-HD MA contains a DTS core so even if your decoder doesn't support DTS-HD MA, it'll decode the core audio track and you'll lose nothing by keeping the HD.

[Carpo]

now all i have to decide is whether to re-rip and convert 20 blurays using crf --18 (used 2 pass before i saw the light) or just remuxing the ones i have with the audio......i think i need a stiff drink first :D

[SamuriHL]

HEHE. Good luck. :) Personally, I'm SOOOOO happy with how things are progressing with LAVF Splitter and LAV Audio. The ability to watch movie only blu-rays in my player of choice is awesome. The ability to decode HD audio in full bitdepth, full quality is awesome. Add madVR to the mix and wow...just wow. :)

[Carpo]

Yes i have to admit that since i switched to madVR+LAV CUVID+Lav Audio, my image and sound quallity have gone through the roof, now if i could just replace mpc-hc (hint hint nevcairiel) ;)

[nevcairiel]

Its fine (for now) :p

The main problem with replacing MPC-HC is that the subtitle quality would go significantly down by using ffdshow or DirectVobSub, so that would have to be replaced first, or alot of people will be unhappy. Personally, i dont usually watch with subtitles, but when i do (mostly only forced subs), i won't accept blurred and washed out subs. :p

[SamuriHL]

I already have. :D I'm using J River's MC16. Admittedly not free, but, just a joy to use. Nonetheless, I still custom build my own MPC-HC builds that don't include ANY internal filters. :D

[nevcairiel]

I have all my content neatly organized into folders, most library-like players will always try to force some stupid organization scheme of their own..
That said, i did kinda like the library of XBMC, too bad its not DirectShow (and just using it for the front-end, using MPC-HC for playback, is kinda crappy).

[SamuriHL]

I was surprised to learn that XBMC didn't use directshow, as well. MC16, once I forced it to do things *MY* way in terms of library handling, works very well for me. I agree, that I've never really been one for library players, either, but, I have it setup quite well now.

[nevcairiel]

I wasn't surprised, its a multi-platform player, sparked back on the XBox, i would've been surprised if it used DirectShow. :P
Does MC16 support scrapping series infos and automatic tagging from like thetvdb? That was really neat in XBMC, i had to rename maybe a handful of folders, and it imported my whole TV series library without a single hiccup.

[SamuriHL]

Well, I guess you do have a point there. :)

[nevcairiel]

I wish there was another player with a neat 10ft UI, which i could just point at my TV series library, tell it "these are TV series", and it would do all the magic.
I've really only seen this in XBMC. Using regex'es on folder and file names to separate stuff by show, and season. Downloading coverart by itself, tagging based on video/audio codecs used, and everything in a neatly organized interface.

I just want a menu structure like this, in a usable 10ft UI:


Movies
- Foo
- Bar
- ...
TV Series
- Show1
- Season 1
- Episode 1
- Episode 2
-...
- Season 2
-...
- Show2
-...


That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?

Oh well, can't have it all, i guess.

[HeadlessCow]

I wish there was another player with a neat 10ft UI, which i could just point at my TV series library, tell it "these are TV series", and it would do all the magic.
I've really only seen this in XBMC. Using regex'es on folder and file names to separate stuff by show, and season. Downloading coverart by itself, tagging based on video/audio codecs used, and everything in a neatly organized interface.

I just want a menu structure like this, in a usable 10ft UI:


Movies
- Foo
- Bar
- ...
TV Series
- Show1
- Season 1
- Episode 1
- Episode 2
-...
- Season 2
-...
- Show2
-...


That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?

Oh well, can't have it all, i guess.

http://www.mediabrowser.tv/ perhaps?

[SamuriHL]

That's a good question if it can do that or not. If not, we can always request it. :) I think they'd be interested in that idea. (Referring to MC16)

[nevcairiel]

The Media Browser UI is ok, but then i'm stuck with WMC playing the stuff, which is ugh. :P

[hoborg]

Try this:
http://files.1f0.de/lavf/LAVFilters-0.19-audio-jitter-test.zip

@nevcairiel:
Using "dtsdecoderdll.dll" will reintroduce this DTS stuttering issue :/

[nevcairiel]

Using "dtsdecoderdll.dll" will reintroduce this DTS stuttering issue :/

No it doesn't.
There are no differences in which decoder is used, unless your system is too slow to real-time decode DTS-HD MA. :p

If there is any stuttering again, first try to turn off the new A/V sync option, as DTS-HD in MKV can sometimes be so f'ed up, that it might confuse it and cause a resync.

[SamuriHL]

The Media Browser UI is ok, but then i'm stuck with WMC playing the stuff, which is ugh. :P

Yea, NO. Just NO. I gave up on 7MC. It just doesn't handle everything I want. And since MC16 allows me to play WTV files I can have 7MC record and use MC16 to play them back. Works for me! In any case, I need to play around with the library features more in MC16 as i've had to manually setup a lot of what I have.

[hoborg]

No it doesn't.

You are right. I forgot to disable MPC-HC internal decoders...
Sorry for that. It is working just fine.

[nevcairiel]

Yea, NO. Just NO. I gave up on 7MC. It just doesn't handle everything I want. And since MC16 allows me to play WTV files I can have 7MC record and use MC16 to play them back. Works for me! In any case, I need to play around with the library features more in MC16 as i've had to manually setup a lot of what I have.

If nothing else helps, there is always the ability to use XBMC with an external player configured (MPC-HC), not sure how that plays with madVR however, as XBMC is already a full screen D3D application .. maybe i should test that one day.

[pankov]

nev,
you can configure XBMC to not work in D3D exclusive mode and this way it doesn't conflict with the player (ZoomPlayer in my case).
Sadly the interaction is not as smooth as I'd like, but sadly ZoomPlayer's Media Library is nowhere near as nice as the one in XBMC.
I'm really hoping somebody can combine the UI of XBMC and the versatility of ZoomPlayer in one.

[SamuriHL]

If nothing else helps, there is always the ability to use XBMC with an external player configured (MPC-HC), not sure how that plays with madVR however, as XBMC is already a full screen D3D application .. maybe i should test that one day.

I'm not too concerned. So far I'm really pleased with MC16. Watching Star Trek last night was a joy. I stuck the disc in, let AnyDVD scan it, MC16 popped up a "Hey, a disc!" message and I was off and running. Very very happy. As for the library, I manually groomed it yesterday so it's setup the way I want. It's not perfect, though, and I think I may think about it architecturally and see if I can make some recommendations to the J River team on how to improve it. Or at least allowing us to manage it the way we want. Like you, I wanna group shows together, and it really doesn't seem to allow you to do that. Yet. ;)

[Mercury_22]

The Media Browser UI is ok, but then i'm stuck with WMC playing the stuff, which is ugh. :P
Not really ! :)
I'm using MPC-HC as default player in WMC for everything using Media Center Open With (http://mikinho.com/wmc/open-with/) and for ISO Media Center Mount Image (http://mikinho.com/wmc/mount-image/):p

[nevcairiel]

Well can as well do the same with XBMC then. :p

[Mercury_22]

Well can as well do the same with XBMC then. :p

Yeah but I like to have a minimum of programs/filters... on my system and since WMC it's already there ... :)
I've made an exception for LAV but I can't wait for the MPC-HC-LAV branch (hint, hint...maybe the time has come ?):p

[jmone]

On MC, you can change any of the view to suit by using meta data on the files (of which a directory stucture/name can be one). While I also store my TV Series in a similar stucture the nevcairiel, my new views tend to use populated meta data as it is easer. There is also a thread on "Sickbeard" integration is that seems to be an auto TVdb scraper and meta data importor for TV Series (I've not been folloiwng it or trying it out however).

[CruNcher]

I wish there was another player with a neat 10ft UI, which i could just point at my TV series library, tell it "these are TV series", and it would do all the magic.
I've really only seen this in XBMC. Using regex'es on folder and file names to separate stuff by show, and season. Downloading coverart by itself, tagging based on video/audio codecs used, and everything in a neatly organized interface.

I just want a menu structure like this, in a usable 10ft UI:


Movies
- Foo
- Bar
- ...
TV Series
- Show1
- Season 1
- Episode 1
- Episode 2
-...
- Season 2
-...
- Show2
-...


That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?

Oh well, can't have it all, i guess.

yep and additionally this to command it :)

http://www.youtube.com/watch?v=2kR2C6IeaPw
http://www.youtube.com/watch?v=zBNyUBRwmSQ
http://www.youtube.com/watch?v=T6K-2spC_P4
http://voxcommando.com/

Though as it uses the internal Vista/7 Speech Recognition API it wont work with XP ;)

[SamuriHL]

On MC, you can change any of the view to suit by using meta data on the files (of which a directory stucture/name can be one). While I also store my TV Series in a similar stucture the nevcairiel, my new views tend to use populated meta data as it is easer. There is also a thread on "Sickbeard" integration is that seems to be an auto TVdb scraper and meta data importor for TV Series (I've not been folloiwng it or trying it out however).

Neat. I'm gonna have to look into how to do those customizations.

[dbone1026]

On MC, you can change any of the view to suit by using meta data on the files (of which a directory stucture/name can be one). While I also store my TV Series in a similar stucture the nevcairiel, my new views tend to use populated meta data as it is easer. There is also a thread on "Sickbeard" integration is that seems to be an auto TVdb scraper and meta data importor for TV Series (I've not been folloiwng it or trying it out however).

Sickbeard works well with SABNzbd for an "internet pvr". You can also use 3rd party programs such as metabrowser to automatically rename and scrape tv shows.

I use Media Browser + MPC HC (for mkvs) and TMT3 (for BluRay folder rips). This gets around the need to have to hack up WMC to get any sort of proper player. The big drawback though is once you start relying on 3rd party players to launch it always leads to the possibilities of instabilities (not good for the all important WAF). The ideal front end is one such as Media Browser or XBMC but also where the internal player can handle proper playback of all files.

[Mark_A_W]

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.

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. :)


Basically I'm forced to use LAV Audio with LAV Splitter.

Despite my settings in Zoom Player Smart Play, LAV Audio is somehow being forced as the FLAC decoder whenever LAV Splitter is used.

Until I sort that out, it seems to be either the LAV Splitter, or madFLAC, but not both :confused:

[mkanet]

Is there any way to tell if audio decoders are decoding "real" DTS-MA and True-HD; and, not just secretly using their respective DTS/AC3 cores?

It is easy to tell if there is DTS-MA/DTS core or TrueHD/AC3 core using mediainfo, but it doesn't show what's being actually decoded... the core or the full blown format.

Reclock seems to just say bit rate/depth/bitexact.. but, I dont see anywhere where it shows if it's ultimately playing back an ac3 stream/dts stream; instead of parent DTS-MA/TrueHD formats.

Thanks,
Michael

[SamuriHL]

TrueHD doesn't have a core. It's NOT AC3.

EDIT: P.S. If you want to know for SURE if you're getting HD and not "core", try a 7.1 track and see how many channels you're getting.

[noee]

That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?

That's the beauty of MC, IMO. It can do almost anything view-wise that you want, there's no reason to be locked into the "stock" views, and the new Theater View is bringing some good stuff, presentation-wise. It doesn't have all of the import "bells and whistles", but even with a standard folder structure, using it's "Fill Properties from Filenames" feature, you can get a solid start if you're stuck on explicit folder structures for organization.

[mkanet]

Thanks, but I would still like to know ultimately what's being played back in realtime; as some filtergraph components are able to somehow get DTS/AC3 from the original DTS-MA/TrueHD audio stream. I dont think there's any real-time re-encoding/playback of TrueHD to AC3 and DTS-MA to DTS in any of the filtergraph components I have. I dont know how that gets done, but that's really not the point... I just wanted a way to see what kind of stream is actually being played back by the DS audio renderer if at all possible.

EDIT: I currently have arcsoft set to 5.1 speaker settings since thats all I have. I ultimately get that.

TrueHD doesn't have a core. It's NOT AC3.

[dbone1026]

Thanks, but I would still like to know ultimately what's being played back in realtime; as some filtergraph components are able to somehow get DTS/AC3 from the original DTS-MA/TrueHD audio stream. I dont think there's any real-time re-encoding/playback of TrueHD to AC3 and DTS-MA to DTS in any of the filtergraph components I have. I dont know how that gets done, but that's really not the point... I just wanted a way to see what kind of stream is actually being played back by the DS audio renderer if at all possible.

For truehd that is easy, play back an mkv with a truehd track as there is no embedded ac3 (as opposed to m2ts), so only the truehd track is present to decode

[SamuriHL]

TrueHD on a Blu-ray is stored in an interleaved track. So there is an AC3 track and a TrueHD track. They are NOT the same track. dbone speaks the truth (depending on how you make it, that is...there IS a way to store an interleaved THD+AC3 track in an MKV he just doesn't know it :D) in that most MKV's that contain TrueHD aren't interleaved.

[mkanet]

Thanks... isn't there a tool that just tells us what's being played back by the audio renderer? It seems like it would be a nice tool for quick reference. Maybe there isn't anything like that available yet?

Edit, I have downloaded all the sample files that many people seem to use (ie, Surround_TrueHD_7.1.m2ts, Surround_TrueHD_7.1.mkv, Surround_DTS-HD_MA_7.1.m2ts, Surround_DTS-HD_MA_7.1.mkv). How can I tell if the DTS-MA is really DTS MA; not just DTS? Maybe at least the truehd mkv doesnt have an interleved ac3 audio track?

For truehd that is easy, play back an mkv with a truehd track as there is no embedded ac3 (as opposed to m2ts), so only the truehd track is present to decode

[dbone1026]

TrueHD on a Blu-ray is stored in an interleaved track. So there is an AC3 track and a TrueHD track. They are NOT the same track. dbone speaks the truth (depending on how you make it, that is...there IS a way to store an interleaved THD+AC3 track in an MKV he just doesn't know it :D) in that most MKV's that contain TrueHD aren't interleaved.

Hmmm...never heard of that. I thought the matroska spec would not allow this since the ac3 track is completely separate from the truehd track. At least none of common tools used right now allow for this so highly unlikely anyone would run in to currently.

[SamuriHL]

ReClock will tell you the number of channels the audio renderer is receiving. So, again, if it's getting 8 channels (regardless of how you output to the receiver) then you know it's decoding the HD track.

[SamuriHL]

Hmmm...never heard of that. I thought the matroska spec would not allow this since the ac3 track is completely separate from the truehd track. At least none of common tools used right now allow for this so highly unlikely anyone would run in to currently.

Solveig allows it to be done by request. :) I've tested it. It works. A real THD+AC3 track in an MKV. Crazy right? :D

[mkanet]

I just tried every single type of container I have that has an HD audio track (the samples I mentioned above with plane flying in a circle):

http://i67.photobucket.com/albums/h283/mkanet/Screenshot-1509.jpg

It looks like that answers my question :) Thanks SamuriHL.

ReClock will tell you the number of channels the audio renderer is receiving. So, again, if it's getting 8 channels (regardless of how you output to the receiver) then you know it's decoding the HD track.

[SamuriHL]

I just tried every single type of container I have that has an HD audio track (the samples I mentioned above with plane flying in a circle):

http://i67.photobucket.com/albums/h283/mkanet/Screenshot-1509.jpg

It looks like that answers my question :) Thanks SamuriHL.

Now you know you can trust that LAV Audio is doing what we all say it is. :D Nev has done an absolutely FANTASTIC job.

[mkanet]

Actually, thats with arcsoft audio decoder HD (which can handle all HD audio formats). However, LAV audio decoder does the same too! I'm currently using LAV splitter for everything except bluray in sageTV; which requires sageTVdemux.

Now you know you can trust that LAV Audio is doing what we all say it is. :D Nev has done an absolutely FANTASTIC job.

[SamuriHL]

Actually, thats with arcsoft audio decoder HD (which can handle all HD audio formats). However, LAV audio decoder does the same too! I'm currently using LAV splitter for everything except bluray in sageTV; which requires sageTVdemux.

Well, as we know, no open source decoder can decode DTS-HD MA. So, when I say LAV Audio, it's implied that you're using the ArcSoft dtsdecoderdll with it.

[JustinChase]

I wish there was another player with a neat 10ft UI, which i could just point at my TV series library, tell it "these are TV series", and it would do all the magic.

I just want a menu structure like this, in a usable 10ft UI:


Movies
- Foo
- Bar
- ...
TV Series
- Show1
- Season 1
- Episode 1
- Episode 2
-...
- Season 2
-...
- Show2
-...


That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?



Yes, MC16 can do that; at least the 10ft UI display part, and I think I've read that they have or are working on getting TV series metadata automatically also, but I have not tried it yet.

I had some trouble getting the Theater View to display the way i wanted, but in this thread, some good feedback got me all squared away. i basically wanted to do the same thing, but with Formula 1 seasons, not TV seasons, but the method and results are the same.

http://yabb.jriver.com/interact/index.php?topic=64038.0

I think you'll really like it after a little bit of setup. After selecting video, you get these options (all changeable)

http://www.pix01.com/gallery/B6318F5F-4769-4385-B3CA-EE27B9BD15F9/theater_view/39975387_orig0.jpg

selecting Formula 1 above goes to this, with seasons shown (i use the year as the season), and the ability to select other fields to display instead if you prefer/need.

http://www.pix01.com/gallery/B6318F5F-4769-4385-B3CA-EE27B9BD15F9/theater_view/39975387_orig2.jpg

and here is what you get if you select Places instead of season (all of which is customizable/changeable)

http://www.pix01.com/gallery/B6318F5F-4769-4385-B3CA-EE27B9BD15F9/theater_view/39975387_orig1.jpg

then if you pick a place/track, you get whatever choice you setup to show next, season in this case, like this...

http://www.pix01.com/gallery/B6318F5F-4769-4385-B3CA-EE27B9BD15F9/Theater_View_2/626910971_orig2.jpg

It can have more layers if you want, or it can play the file you select from here; however you like.

and MC is working perfectly with LAV splitter, LAV Audio, ffdshow video decoder and madVR

Thanks again for all your hard work!!

[mkanet]

Okay, thanks for pointing that out. But, it's not implied I was using LAV audio for TrueHD. :) Im just using Arcsoft for decoding all audio formats it can support.

I definitely love LAV splitter; and use it for everything except sageTV bluray. BTW: Nev thanks for all your hard work!

Well, as we know, no open source decoder can decode DTS-HD MA. So, when I say LAV Audio, it's implied that you're using the ArcSoft dtsdecoderdll with it.

[mkanet]

Hi Andy,

I have great news. I know we've had our differences in the past; however, I was hoping we can call it truce and try to repair our relationship by extending an olive branch... offering a pre-release MpegDemux.ax which sounds like you may be very interested in. This build is only hours old. I was given permission to share this demux for people who needed it. I've been working directly with the developer at sageTV who appears to be responsible for all code changes made to MpegDemux.ax. This demux should hopefully be included in the next sageTV beta.

This pre-release version of sageTV demux is now fullly compatible with Arcsoft Audio Decode HD (full support for DTS-MA/TrueHD decoding without a hitch). In addition to that, it now also supports LAV Audio Decoder's TrueHD. If you have the arcsoft audio decoder package installed, all you have to do it tell sageTV to enable HD audio support and specify a single audio decoder: Arcsoft Audio HD. If you dont have arcsoft, you can at least have TrueHD audio decoding via LAV Audio decoder. My sageTV client has it's internal demuxing off; everything but bluray audio formats (and I think TV).

You might have noticed an earlier post from me today playing back a sample 7.1/96khz m2ts file directly in sageTV.


I got the same result no matter which container I used . I also had the same results with DTS-MA. Of course, it also works just as great with blu-ray discs/iso's as well on on my extender.

Here's the file:
http://hotfile.com/dl/117954203/045f397/MpegDeMux.zip

I've only tested this with TMT3 standalone decoder package (doesn't require to install the entire player just to be able to have access to the audio filter... only a small change in the Windows win.ini file:

1) Open up C:\Windows\win.ini in notepad.
2) Add the following text to the end:
Code:

[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=6
AudioDynamicRange=1

3) Change the numbers to match the settings you prefer.

Code:

AudioDynamicRange:
0: Light
1: Normal
2: Threaded

AudioOutputMode:
0,1,2,5: Stereo
6,13: 5.1 Channels
17: 7.1 Channels
16: S/PDIF

Sincerely,
MKANET


TBH, I'm really only interested in making it work with the Sage demux; it uses the right subtype ({EB27CEC4-163E-4CA3-8B74-8E25F91B517E}) :)

[nevcairiel]

Using the ArcSoft Audio Decoder and its complicated static configuration is not required anymore. :p
You should read the LAV Audio 0.26 release notes.

Also, please don't advertise other demuxers in my thread, thats really not a nice thing to do.

Solveig allows it to be done by request. :) I've tested it. It works. A real THD+AC3 track in an MKV. Crazy right? :D

Just because it can be done doesn't mean the resulting file is valid. Having this kind of track in a MKV is *NOT* a valid file. MKV does *NOT* allow for interleaving two audio streams into one mkv stream. mkvmerge cannot do it, and Mosu specifically denied the feature some time back because it would go against any MKV design, and it is basically the reference muxer.

Besides, i don't get the advantage of that hack. You will not be able to play the AC3 stream with most players, if even with any player. Some tools might even get confused and break on that stream (like LAV Audio). So, why?

[Gleb Egorych]

There (http://forum.doom9.org/showthread.php?p=1500788#post1500788) I wrote about re-sync issues in AVI files when using MS AVI splitter with LAV Audio. I've finally found a small sample that demostrates the problem. "A/V sync" option makes situation even worse.

Here is the sample (4MB): http://www.mediafire.com/?iduygkxulmejmky

[madshi]

@nevcairiel, good work on the ArcSoft decoder!

To be honest, I'm afraid of violating license stuff, so that's why I made eac3to require the decoder to work via DirectShow before I talk to it directly.

One question: How are you handling the channel configuration stuff? Basically you can ask the decoder to output whatever channel configuration you like. Some of which it will convert to, some of which it won't. Are you asking the decoder to output the *native* channel configuration? Or are you asking it to output the channel configuration that matches the speaker configuration?

[nevcairiel]

One question: How are you handling the channel configuration stuff? Basically you can ask the decoder to output whatever channel configuration you like. Some of which it will convert it, some of which it won't. Are you asking the decoder to output the *native* channel configuration? Or are you asking it to output the channel configuration that matches the speaker configuration?

I ask it to output 8 channels. It doesn't really treat this as a "convert to this layout" setting, more like a "maximum channels" setting.
Maybe one of the parameters i couldn't figure out yet controls a "minimum" number of channels, or something like that.
How would you tell it to give you "native" channels?

With my current settings it works like this, on the content i threw at it:

- Stereo, 5.1 and 7.1 gets output as-is (2, 6, 8 channels respectively)
- Mono gets output as Stereo (only 1.1.0.8 of the dll seems to work at all with Mono)
- 6.1 gets converted to 7.1 by doubling back-center to back left/right.

I have not really got my hands on any other, non-standard layouts.
The two conversions seem "ok" to me, its not perfectly reproducing the source layout, but for playback they are acceptable conversions, imho.

I've only tested with 1.1.0.7 and 1.1.0.8 (recent TMT3 and TMT5 versions, respectively), and i cannot say if older versions treat anything differently.

[madshi]

You really do get 6 channels output even if you ask the decoder to output 8? Are you sure? That's not what I'm getting here. If I ask for 8 channels then I do get 8 channels. Maybe we're using different parameters? Maybe we should move this discussion to PM?

[jmone]

The two conversions seem "ok" to me, its not perfectly reproducing the source layout, but for playback they are acceptable conversions, imho.


I agree - any other mixing I want to do will by via a DSP later in the filterchain so outputing the souce number of tracks from the audio decoder is my preference (and I too can certainly "live" with 1.0 --> 2.0 and 6.1 to 7.1 being done in the renderer). Acording to http://www.blu-raystats.com/Stats/Stats.php of the 1,761 Blu-ray discs it is currently tracking with DTS-HD MA only:
* 10 have 6.1
* 15 have 1.0

Then again, apparantly:
* 21 have 5.0
* 3 have 4.0
* 4 have 3.0

[Gleb Egorych]

Basically I'm forced to use LAV Audio with LAV Splitter.

Despite my settings in Zoom Player Smart Play, LAV Audio is somehow being forced as the FLAC decoder whenever LAV Splitter is used.

Until I sort that out, it seems to be either the LAV Splitter, or madFLAC, but not both :confused:
In ZP turn on showing Smart Play errors, I guess in your case Smart Play fails. I tested flac files, ffdshow and LAV Audio successfully connected to LAV Source while madFLAC refused to connect. So I suppose for FLAC in MKV there is a similar problem.

[nevcairiel]

Then again, apparantly:
* 21 have 5.0
* 3 have 4.0
* 4 have 3.0

Would be great to get my hands on DTS samples with these channel counts.
But then, i don't have much influence on what the ArcSoft decoder does..

[pankov]

nev,
I have a theoretical question about the best configuration for playback (theoretical, because I'm still waiting for my new Receiver that will have HDMI inputs - Integra 40.3).
I am/will be using 5.1 speaker setup and I'm kind of worried about 6/7.1 sources. How many speakers should I configure in Windows? Will the receiver do the conversion from 6/7.1 to 5.1 if I feed it 7.1 PCM via HDMI? Is it better to do it at the PC - at the decoding stage so the decoder can use the meta data in the audio track (wildly guessing here)? If the latter is the preferred way how should I configure the LAV Audio / dtsdecoderdll.dll so it can output up to 5.1 channels?

[nevcairiel]

LAV Audio does not support requesting a special channel count (yet), or any post-processing.

Your best bet is probably using 7.1 output from windows, and hope that your receiver does properly add the 2 back channels into the side channels.
Otherwise, you could use something like ffdshow or any other post-processor to matrix your audio properly into 5.1. I might eventually add support for requesting 6 channels out of the 8 channels, but i cannot be sure how well that works.

What i would try to avoid is setting Windows to 5.1 and using the Windows mixer to downmix, as that'll most likely end up being not so good.

[pankov]

10x for the info
I'll report my findings when I get the receiver.

[joeydrunk]

I wish there was another player with a neat 10ft UI, which i could just point at my TV series library, tell it "these are TV series", and it would do all the magic.
I've really only seen this in XBMC. Using regex'es on folder and file names to separate stuff by show, and season. Downloading coverart by itself, tagging based on video/audio codecs used, and everything in a neatly organized interface.

I just want a menu structure like this, in a usable 10ft UI:


Movies
- Foo
- Bar
- ...
TV Series
- Show1
- Season 1
- Episode 1
- Episode 2
-...
- Season 2
-...
- Show2
-...


That can't be too hard, or?
Can i get MC16 to do that, even if i have to manually organize my data in there?

Oh well, can't have it all, i guess.


Why don't you just use xbmc and use mpchc. You can set it up to where it acts exactly like an internal player. You just add the playercorefactory.xml file to the xbmc user data file and that's it. You can add any player to launch. Didn't you say you worked on dsplayer a bit? Will your lav filters work with that? I couldn't handle watching my movies strictly from a player.:,logfile:

[nevcairiel]

Using an external player from XBMC works, but its not perfect. The integration is never really 100% smooth.

DSPlayer does not support madVR, and i'm not going back. :p
Also, its dead. :)

[Mark_A_W]

In ZP turn on showing Smart Play errors, I guess in your case Smart Play fails. I tested flac files, ffdshow and LAV Audio successfully connected to LAV Source while madFLAC refused to connect. So I suppose for FLAC in MKV there is a similar problem.

That's the thing - it IS turned on. I get errors for other stuff if I have it set wrong. Usually I have "fall back on directshow" disabled, but sometimes I use that for clues on what will connect to what.

I would expect an error in this case.


It's very strange, I have 10 years or so of playing with ZP media graphs/Smart play, and only a couple of times have I seen it do things it shouldn't.


I'm currently using File Source Async, then LAV Splitter, then the Audio decoder. I'll try with LAV Source (Splitter ticked, or using the LAV Splitter). Same result each time, it doesn't load madFLAC.


Anyway, so you can't get LAV Source/Splitter to connect to madFLAC either? You implied that you didn't test an MKV file, but something else?

Mark

[Mark_A_W]

I ask it to output 8 channels. It doesn't really treat this as a "convert to this layout" setting, more like a "maximum channels" setting.
Maybe one of the parameters i couldn't figure out yet controls a "minimum" number of channels, or something like that.
How would you tell it to give you "native" channels?

With my current settings it works like this, on the content i threw at it:

- Stereo, 5.1 and 7.1 gets output as-is (2, 6, 8 channels respectively)
- Mono gets output as Stereo (only 1.1.0.8 of the dll seems to work at all with Mono)
- 6.1 gets converted to 7.1 by doubling back-center to back left/right.

I have not really got my hands on any other, non-standard layouts.
The two conversions seem "ok" to me, its not perfectly reproducing the source layout, but for playback they are acceptable conversions, imho.

I've only tested with 1.1.0.7 and 1.1.0.8 (recent TMT3 and TMT5 versions, respectively), and i cannot say if older versions treat anything differently.

This behaviour (mono to stereo, 5.1 unchange, 6.1 to 7.1, 7.1 unchanged), is perfect if you ask me :)

[hawkson]

"Can LAV Audio decode DTS-HD?
A: No. DTS-HD is not supported by the ffmpeg decoder libraries, so you'll only get the DTS "core", which is limited to 5.1 channels and 48kHz."


In the AVS forums, you said you can do DTS-HD. Reading your second post, it says it can't.

Also, can you provide a how to use guide? I installed it, now what? :)

[jmone]

Have a read of the first post in this thread - it explains how you can enable DTS-HD.

[Carpo]

so lav audio will decode true-hd before i send it to the receiver in bitstream? as it wont do anything other than DD

[jmone]

Would be great to get my hands on DTS samples with these channel counts.
But then, i don't have much influence on what the ArcSoft decoder does..

It's a bit old but here is a list of disks http://forum.blu-ray.com/blu-ray-movies-north-america/158996-list-bds-dts-hd-ma-1-0-2-0-3-0-4-0-mixes.html

[hoborg]

Thanks for .26!

BTW, any chance to teach LAVF audio decoder RealMedia COOK format?

[Gleb Egorych]

Anyway, so you can't get LAV Source/Splitter to connect to madFLAC either? You implied that you didn't test an MKV file, but something else?
That time I tested pure .flac file, madFLAC didn't connect to LAV Source and ZP showed Smart Play error. ffdshow Audio Decoder and LAV Audio Decoder connected without a problem.

Now I muxed that file into MKA using mkvmerge and played it using ZP. Tried LAV Source and File Source Async+LAV Splitter. madFLAC connected and played audio successfully.

http://img829.imageshack.us/img829/7391/zplavmadflac.png

[nevcairiel]

That time I tested pure .flac file, madFLAC didn't connect to LAV Source and ZP showed Smart Play error. ffdshow Audio Decoder and LAV Audio Decoder connected without a problem.

Now I muxed that file into MKA using mkvmerge and played it using ZP. Tried LAV Source and File Source Async+LAV Splitter. madFLAC connected and played audio successfully.

I'll test that later today. I don't remember if i did before..

[Mark_A_W]

That time I tested pure .flac file, madFLAC didn't connect to LAV Source and ZP showed Smart Play error. ffdshow Audio Decoder and LAV Audio Decoder connected without a problem.

Now I muxed that file into MKA using mkvmerge and played it using ZP. Tried LAV Source and File Source Async+LAV Splitter. madFLAC connected and played audio successfully.

http://img829.imageshack.us/img829/7391/zplavmadflac.png


Thanks.

Looks like there's something fishy in my system. I'll have to dig deeper.

Cheers

Mark

[pankov]

Mark_A_W,
sorry to join so late in the discussion about the madFlac problem but could you post a sample, so I can try and confirm/deny if actually is a problem only on your system?

[Andy o]

Nev, DTS-ES discrete has wrong channel mappings on LAV decoder (not Arcsoft). See this test (http://www.megaupload.com/?d=NQ36GF0I).


With my current settings it works like this, on the content i threw at it:

- Stereo, 5.1 and 7.1 gets output as-is (2, 6, 8 channels respectively)
- Mono gets output as Stereo (only 1.1.0.8 of the dll seems to work at all with Mono)
- 6.1 gets converted to 7.1 by doubling back-center to back left/right.

With 6.1 on a 7.1 system there's one thing... L and R side surrounds get leaked to L and R back surr. respectively. Center back surround plays in both back surrounds as you say. It's not a big problem, but if you're actually running a 6.1 system, you'll see your L side and R side leaked into the same Back Surround speaker.


EDIT: The DTS-ES discrete issue with LAV audio only happens with WASAPI exclusive, where the decoder is outputting actually 6.1, and making Windows output 6.1 as well. If Windows outputs 7.1 (not exclusive mode) then mapping is OK.

[SamuriHL]

Just because it can be done doesn't mean the resulting file is valid. Having this kind of track in a MKV is *NOT* a valid file. MKV does *NOT* allow for interleaving two audio streams into one mkv stream. mkvmerge cannot do it, and Mosu specifically denied the feature some time back because it would go against any MKV design, and it is basically the reference muxer.

Besides, i don't get the advantage of that hack. You will not be able to play the AC3 stream with most players, if even with any player. Some tools might even get confused and break on that stream (like LAV Audio). So, why?

The original reason I was looking into it was when i first started doing mkv's. I wanted to be able to quickly convert back to BD without having to rerip the disc. Tsmuxer won't touch a non-interleaved THD track. Hence the interest in it. I wasn't suggesting it SHOULD be done, only that it could be. I personally just use makemkv to make them and no longer worry about the BD format.

[nevcairiel]

Nev, DTS-ES discrete has wrong channel mappings on LAV decoder (not Arcsoft). See this test (http://www.megaupload.com/?d=NQ36GF0I).

With that you mean when ffmpeg is doing the decoding, the "dca" decoder, right?

It could just be that outputting 6.1 through WASAPI is not supported. Once PCM leaves through HDMI or S/PDIF, the channel layout is gone. The layout is *only* used by the DirectSound mixer to know which channel is what when upmixing. Of course its also used when mapping PCM to analog speaker outputs. But HDMI does not carry speaker assignments.

I guess thats why the ArcSoft decoder upmixes to 7.1
If playing with the ArcSoft decoder, is that surround leak into the back channels also present?

[Mark_A_W]

Mark_A_W,
sorry to join so late in the discussion about the madFlac problem but could you post a sample, so I can try and confirm/deny if actually is a problem only on your system?


Thanks.

Firstly, there almost certainly isn't a madFLAC problem, or LAV Filter problem, just something funny with my system.

Any bluray sourced MKV file with FLAC audio (created with eac3to and mkvmerge) will show the problem for me.

I just can't get madFLAC and the LAV Splitter to go together.

If you can test any file, that would be appreciated. If you don't have an MKV file, I'll find a link to a sample.

Mark

[joeydrunk]

Using an external player from XBMC works, but its not perfect. The integration is never really 100% smooth.

DSPlayer does not support madVR, and i'm not going back. :p
Also, its dead. :)

It's pretty much smooth, its not perfect but its a lot better than just using a player to watch all your movies. Also I know its dead bit I'm using a pre Eden build compiled with dsplayer to bitstream seamlessly in xbmc because Im having trouble with the playercorefactory.xml file in my htpc. Just saying. Is jriver the only actual media center then where you can use your filters on an internal player, why not use that then? Why no madvr in dsplayer?

[Andy o]

With that you mean when ffmpeg is doing the decoding, the "dca" decoder, right?

It could just be that outputting 6.1 through WASAPI is not supported. Once PCM leaves through HDMI or S/PDIF, the channel layout is gone. The layout is *only* used by the DirectSound mixer to know which channel is what when upmixing. Of course its also used when mapping PCM to analog speaker outputs. But HDMI does not carry speaker assignments.Yeah, I meant with the dca decoder. So then HDMI only carries the number of channels, and the receiver has to know what to do with them? I'm not sure what you meant exactly by 6.1 not supported through WASAPI, but my receiver is getting 6.1 LPCM (with surround back center lit up) with that decoder. Is that what you meant?

I guess thats why the ArcSoft decoder upmixes to 7.1
If playing with the ArcSoft decoder, is that surround leak into the back channels also present?Actually I meant that it happens with the Arcsoft decoder. I suspect it's actually the DTS spec for it, when mapping 6.1 to 7.1 systems, and not a bug. My receiver does it with DTS-HD MA 6.1, but not with DTS-ES. The only difference is that Arcsoft does it with DTS-ES and DTS-MA 6.1. You can tell this with that sample as well.

[nevcairiel]

Yes, HDMI can only know the number of channels, any other metadata cannot be carried over the link. So what exactly do you mean with wrong channel mappings?

The only thing possible is that the channels are ordered wrong.

[Andy o]

When playing with WASAPI exclusive, the ATI 5770 outputs 6.1 channels PCM with the dca decoder, including the discrete rear center. L, C and R are fine, but then SR goes to SB, SB to SL and SL to SR, then LFE is OK too. The Nvidia doesn't output 6.1 via WASAPI?

When bitstreaming though, the receiver also gets the 6.1 channels, and then channels are OK.

[nevcairiel]

The default PCM layout for 6.1 would be like this:

L R C LFE BC SL SR

From what you're describing, your receiver expects

L R C LFE SL SR BC (which is actually L R C LFE BL BR BC)

I will test on mine, and see if its the same.
Can you test other 6.1 sources, if you have any?

PS:
MS defines the channels like this:

http://msdn.microsoft.com/en-us/library/ff537083(v=vs.85).aspx

The order of the flags also defines the order in the bitstream (thats why its BL BR BC, and not SL SR BC)

Edit:
My receiver does not seem to like 6.1 at all, it claims to receive Stereo, and only the two front channels play. I cannot say if thats a Onkyo thing, or maybe NVIDIA. Bitstreaming works, however, and properly doubles the back center into the back speakers.
It would be great if you could test other 6.1 sources, so i can switch the matrix around.

[Andy o]

If bitstreaming works OK it's probably not the file that has incorrect mapping, right? As far as I can tell the DTS-HD 6.1 content also carries a DTS-ES discrete core, but not the 7.1 content. I think I got one DTS-HD MA 6.1 movie, I'll see if I can tell the channel switches without a discrete test.

It would be great if you could test other 6.1 sources, so i can switch the matrix around.
But when not using WASAPI exclusive and setting Windows to 7.1, mappings are correct. If you switch the matrix, won't this be affected?

Just in case, it's not that I'm complaining, cause I have to use Arcsoft anyway for DTS, but just thought you should know. I'll keep testing, lemme know if there's anything in particular you'd like me to test.

[nevcairiel]

It wouldn't be affected as it would also change the channel mask, so the mixer knows.

[Andy o]

Once PCM leaves through HDMI or S/PDIF, the channel layout is gone.

Hmm so it just dawned on me what this means. I've done some more reading, and I think the DTS-HD MA "2.0" with surround "flag" that some Pioneer receivers like mine are having trouble decoding, don't have a "flag" at all and the mono surround is not matrixed. I think they're 3.0 channels. So, I also know that there are DTS-HD MA 3.0 movies (http://forum.blu-ray.com/blu-ray-movies-north-america/158996-list-bds-dts-hd-ma-1-0-2-0-3-0-4-0-mixes.html), whose 3rd channel is center front. So, then if the decoding was done in the PC, HDMI would not be able to tell the receiver if that 3rd channel is center or mono surround. I know that my receiver when bitstreaming detects mono surround, from the movie I'm having trouble with (Delicatessen), and also detects center properly (from You Will Meet a Tall Dark Stranger). The way the Arcsoft decoder deals with the latter, is to put the 3.0 in a 5.1 PCM stream. With the former, it does not decode the mono surround at all, and just sends 2.0 out.

So, that's one advantage of bitstreaming, I guess. Content with same number of channels, but different configuration.

[nevcairiel]

Yeah, with PCM you need to wrap it into a bigger container. I'm working on extending that support now, so that such a file might be properly wrapped into 5.1. Of course when using that mode, you would lose the ability to use full upmixing capabilities of your decoder (however it handles that), as i would already send some extra PCM channels with just silence.

[Andy o]

I'm pretty sure that Neo:6 could not be enabled with that 3.0 LCR stream, so it's probably the same. You can use it with 2.0 and I think 5.1 content, but nothing weird like 3.0.

[DrNein]

It's pretty much smooth, its not perfect but its a lot better than just using a player to watch all your movies. Also I know its dead bit I'm using a pre Eden build compiled with dsplayer to bitstream seamlessly in xbmc because Im having trouble with the playercorefactory.xml file in my htpc.

There is only a brief flash (desktop and/or wallpaper if applicable) between the end of MPC-HC playback and resumption of XBMC browser. If blank or the same backround image then it is even less noticeable.

I run XBMC with the switch "-p" for portable so the user data is in a subfolder of the application and therefore the playcorefactory.xml for an external player. See below and edit monitor number and file types as necessary.

C:\Program Files (x86)\XBMC\portable_data\userdata\playercorefactory.xml

<playercorefactory>
<players>
<player name="MPC-HC" type="ExternalPlayer" audio="false" video="true">
<filename>C:\Program Files (x86)\MPC-HC\mpc-hc.exe</filename>
<args>"{1}" /monitor 2 /fullscreen /close</args>
<hidexbmc>false</hidexbmc>
<hideconsole>false</hideconsole>
<warpcursor>none</warpcursor>
</player>
</players>
<rules action="prepend">
<rule filetypes="avi|flv|mkv|mov|mp4|m2ts|ts|vob|wmv" player="MPC-HC"/>
</rules>
</playercorefactory>

Note: if internet streaming is desired then do not add certain filetypes (since MPC-HC is incapable of streaming) and rather leave them to XBMC internal player. These include at least flv, mp4, and wmv.


MPC-HC + madVR + LAV + ArcSoft or CyberLink + XBMC = :)

:thanks: to gabest, madshi, nevcairiel, and other developers.

[Plutotype]

Hi folks, Nev,
Need a help to explain some things which are related to audio time shift parameter in MPC-HC. Recently, I have noticed that I need to set almost 200ms audio delay for all of my videos which are being played back on my SONY KDL-40EX500 to have (near) perfect lipsync. When I play the same movies on my PC Monitor ( DELL U2311H ), audio time shift is not needed. So I have used input input lag tool to determine the approximate input lag of the SONY LCD TV. As reference I have used the DELL LCD, which has according to following test 0-30ms input lag ( 10.6ms in average ). http://www.tftcentral.co.uk/reviews/content/dell_u2311h.htm#input_lag
Using high shutter speed camera I have managed to capture the input lag of the SONY LCD TV - 90ms http://imageshack.us/f/813/inputlagtestdell2311hvs.png/
As I can not have input lag tool "always on top" and madVR does not allow to have the fullscreen exclusive mode on two separate displays at once, I have played the same video in windowed mode and run the input lag tool on top ( duplicated desktop ).
So basically. I should set at least 100ms as time shift in MPC-HC and lipsync issue shouldnt occur. I have tested 10 different films where lot of dialogues were present. After further perception, I decided to increase the audio time shift ( audio delay ) for playback on the SONY TV to around 200ms to get ( near ) perfect lipsync. I dont use any video image processing at the SONY´s picture settings beside some colour/B/W calibration. Im not in a game mode, but using the general SONY scene mode. No frame interpolation.
LAVaudio 0.26.
madVR 0.61 with working autochange to 23/24 - 1080p and absolute smooth playback.

1. Can the SONY TV add such terrible input lag of around 200ms?
2. Why multiple tests showed difference between U2311H and the SONY LCD TV only 90ms, but from my perspective I had to set almost around 200ms?
3. Does madvr add something specific what increases for example the rendering times when in fullscreen exclusive mode?
4. Does the input lag goes up with bigger and bigger LCD TVs ( like 46, 52, 60 etc)?
5. Would it be possible to have different audio time shift settings for each display device in MPC-HC :-) ?

Thanks for any input.

Pluto

[nevcairiel]

Somehow i dont think this belongs in this thread. :D

I do have the same issue with my TV however, it needs around ~200ms if delay, except it only needs it at 24p, on 50Hz or 60Hz, its fine. It must do something special in that mode that causes, but i couldn't figure it out yet. I also have any image processing features disabled.
Briefly i thought maybe the delay is in number of frames, as ~5 frames delay at 24p is roughly 205ms, but 5 frames at 50Hz is only 100ms, and maybe i just don't notice the 100ms anymore? But i still think i would notice 100ms, thats still alot of delay..

[Plutotype]

Thanks nevcairiel for your input, sorry for asking here this stuff. Im glad that I´m not the only one that has sensible eyes/ears and needs to add 200ms of audio delay..(-:

Anyway, if you will manage later to add downmixing capabilities for LAVaudio, pls dont forget to add audio delay feature there. As some users have 2 or 3 different display devices it would be nice to have the possibility to set separate audio delay for each display device like madshi has done for the refresh rate autochanger in madVR. But this is just an idea.

The downmixing feature is much needed as with integration of Arcsoft HD audio decoder, user can not change the default number of channels outputted by Arcsoft ( the win.ini hack ). If 7.1 is inputed,7.1 goes out. This would help us to have 7.1 channels downmixed into 5.1 setups.

[nevcairiel]

The Audio decoder will definitely not go around trying to figure out which display devices you have. :P

Not sure about actual downmixing yet. Upmixing is so much easier, you know!
I actually found some article about proper mixing of channels the other day, maybe i'll make 0.27 the big audio processing version.
Already have to work on alot of this stuff for perfect DTS-HD decoding.

Spent the better part of the afternoon (and evening) writing a parser for all kinds of DTS headers .. Core, XCh, XXCh, HD, HD-XCh, HD-XXCh .. the guy who designed the HD header deserves to be shot or something, so freaking complicated. Glad i don't care about the other HD headers, i only want the info about channels and speaker layout.

[Plutotype]

One small add-on ( HD -audio ) and so many details to be fixed afterwards..but i like your approach..

I have another request for correct channel number identification by the LAVsplitter. For example 7.1 audio stream is being reported correctly as 8 channel in Reclock, but as 5.1 ( 6channel ) at the LAVsplitter. For average users like me, it can be confusing..(-:
Last time I have asked about 96k audio streams files reported by LAVsplitter as 48k, you wrote that this is not possible for now. Maybe you will manage to fix this and the correct number of channels reported by LAVsplitter in the future too. LAVaudio is reporting this correctly...superb.

[nevcairiel]

Unlikely.
ffmpeg does all that, and hacking the parsing in ffmpeg just for a cosmetic issue, nah.

[Markolc81]

This LAVsplitter is amazing. Combined with the Arcsoft HD Audio Decoder, I'm getting fully decoded LPCM through Reclock to my AVR and it's awesome when it's lipsynced! However I'm having a problem with my MKVs that contain TrueHD. I don't get any sound when there's a TrueHD track. It won't bitstream properly, and also if I let LAV Audio decode it, it's just a series of pops and hisses. AC3, DTS, DTS-HDMA and FLAC all bitstream or decode perfect with LAV Audio, but no go when it comes to TrueHD. Specifically I tried the rips of Batman Begins BD and Star Trek (2009) BD. Both discs were ripped with Mike's Another EAC3to-gui. Also both these movies bitstreamed fine when I had my ffdshow configured for bitstreaming. Any thoughts?

[nevcairiel]

Sounds like you're not using LAV Splitter, rather some other splitter like Haali or the MPC-HC splitter. TrueHD will not work with those.

[Markolc81]

I do have Lav Splitter listed in my external filters list, and I have it set to preferred. I see that LAV Splitter is being used when playing the MKV by right clicking and looking in the filters being used. I'm going to get Graphstudio installed for a better picture of what's going on. Thanks

[Sebastiii]

The Audio decoder will definitely not go around trying to figure out which display devices you have. :P

Not sure about actual downmixing yet. Upmixing is so much easier, you know!
I actually found some article about proper mixing of channels the other day, maybe i'll make 0.27 the big audio processing version.
Already have to work on alot of this stuff for perfect DTS-HD decoding.

Spent the better part of the afternoon (and evening) writing a parser for all kinds of DTS headers .. Core, XCh, XXCh, HD, HD-XCh, HD-XXCh .. the guy who designed the HD header deserves to be shot or something, so freaking complicated. Glad i don't care about the other HD headers, i only want the info about channels and speaker layout.

:) good luck :) it's amazing stuff again :P

[boyumeow]

Thanks for .26!

BTW, any chance to teach LAVF audio decoder RealMedia COOK format?

If my setup is not wrong, RealMedia COOK format is supported, but not RealMedia video. Thanks.

[Markolc81]

Sounds like you're not using LAV Splitter, rather some other splitter like Haali or the MPC-HC splitter. TrueHD will not work with those.

You were absolutely right. fter setting up Graphstudio I learned that I forgot to uncheck Matroska from the internal filters list of MPC-HC. Thanks Nevcairiel!

Keep up the good work!

BTW, do you take paypal donations for your efforts?

[kennae]

Hi. Thanks for your great filters!

Only feature I really need for my whole PC/HTPC setup is profiles. I need to have two separate profiles for headphones and for AVR.
I have great USB DAC and good headphones, and use spdif/coax out to my AVR receiver when i'm not using the headphones.

With AVR I can use all fancy HD-codecs and stuff, but with the USB DAC it only supports 48KHz 24bit stereo.

So what I need is an easy way to change every audio related setting from one to another several times a day, and it's pain in the ass right now.

I've used ffdshow for audio and there you have profiles but it isnt so fluent and would like to skip the part.
I also need AC3Filter so I can set the channels right but I would love to be able to ditch it and just use your filter and reclock.

The setup I have now is all latest: MPC-HC - FFDSHOW (just the video right now) - LAV Filters - MadVR - Relock.
If I want to change from headphones to AVR I need to change settings in reclock and in LAV audio, and it's not that fast when you have to open a file first.

I can live with my setup since this is what I have been using for years but it would be lovely to have it change profiles/settings by wich device is in use.

[Mark_A_W]

Sounds kinda why I use Zoom Player for movies on the projector (full blown custom filter setup, madVR, LAV, Reclock, Room Correction), but MPC-HC (internal filters and VMR9) for general video/tv on the preview 24" monitor.

And WMP for background music on the "PC speakers" (focal bookshelves), but JMRC for music on the HT system (it has the room correction set up via a VST filter).


Sometimes you want it to "just work" and sometimes you want perfection....

[Andy o]

Why do you have to change settings in LAV? Also, what USB interface do you have? Have you tried using WASAPI exclusive? I'm using a Xonar DX for Dolby Headphone, and exclusive mode takes care of channel mapping. I get correct 5.1 mapping and I don't have to change anything but devices.

[mindbomb]

lets say you have a dts-hd file, and you only have dts checked in the bitstreaming options, but you have the arcsoft dll in the lav audio folder.

what happens? Do you bitstream dts core, or do you decode dts-hd?

I can imagine people using spdif prefer the former, while people using hdmi without PAP prefer the latter.

[Andy o]

You bitstream the core. If you're using S/PDIF though, you might want to use lossless decoding and then have ReClock convert to AC3 so you can take advantage of ReClock.

[nevcairiel]

lets say you have a dts-hd file, and you only have dts checked in the bitstreaming options, but you have the arcsoft dll in the lav audio folder.

what happens? Do you bitstream dts core, or do you decode dts-hd?

I can imagine people using spdif prefer the former, while people using hdmi without PAP prefer the latter.

If DTS bitstreaming is active, it'll always bitstream all DTS formats, no other checks are done.

[Budtz]

Hey guys. Im using ffdshow audio right now to decode audio. Any reason to switch to LAV audio? Is there a quality difference between them? And what about settings like "jitter control" and "resampeling to 48khz" that ffdshow has. Will LAV audio have those or maybe its just not needed with LAV.

[nevcairiel]

The "jitter control" is stupid and not needed. An option that kinda replaces it is the "Auto A/V sync correction" or how i named it, but its really alot different.

Resampling is not supported yet.

[Dogway]

Is there a way to delay audio? Im not sure if it has to do with MPC or LAV audio, but just asking.
Also how can I be sure audio is being bitstreamed? Does it depend on Graphic Card capabilities?

[nevcairiel]

Currently, there is no option in LAV Audio to delay the audio. I'll implement that option in the future, however.

If you enable bitstreaming in LAV Audio, it'll always bitstream. If you get audio, and LAV Audio is still being used, then its bitstreaming.
Yes, it depends on your HDMI connection, usually supplied by the graphics card. Of course also depends on your receiver, if it understands that specific audio format.

[Gleb Egorych]

Im using ffdshow audio right now to decode audio. Any reason to switch to LAV audio? Is there a quality difference between them?
Depends on your usage profile. Some formats are better supported in ffdshow, some are better supoorted in LAV Audio. ffdshow also has a lot of postprocessing features.

[ney2x]

@nevcairiel - can you make a tray icon for LAV CUVID and LAV Filters? I mainly used WMP12 for movies and music, I just want to make sure if they're being used when playing audio/videos. Thanks.

[Sven75]

@nev - did you find some time to look at the lavsplitter + PowerPoint issues I posted on page 130 (#2597)?

I tested version 0.26 and the behaviour is still the same in both cases: flv not working in PowerPoint 2003/2007 and the mpeg file not working at all. In graphstudio everything is working fine.

When using MPC-FLV-Splitter for the flv and Elecard Demuxer for the mpeg both files are fine in PowerPoint, too.

FYI, we are about to finish our first big congress (in Hamburg, btw) with LAV on our computers and it's doing a great job so far. So thanks a lot again!

[nevcairiel]

@nev - did you find some time to look at the lavsplitter + PowerPoint issues I posted on page 130 (#2597)?

Not yet, i've been busy working on new shiny features, its always so much more tempting.... :)
I'll keep it in mind and look at it when i get a chance.

[Sven75]

No prob :)

It is just the icing on the cake (and me being so curious to know as to what is really going wrong with PowerPoint).

[joeydrunk]

There is only a brief flash (desktop and/or wallpaper if applicable) between the end of MPC-HC playback and resumption of XBMC browser. If blank or the same backround image then it is even less noticeable.

I run XBMC with the switch "-p" for portable so the user data is in a subfolder of the application and therefore the playcorefactory.xml for an external player. See below and edit monitor number and file types as necessary.

C:\Program Files (x86)\XBMC\portable_data\userdata\playercorefactory.xml

<playercorefactory>
<players>
<player name="MPC-HC" type="ExternalPlayer" audio="false" video="true">
<filename>C:\Program Files (x86)\MPC-HC\mpc-hc.exe</filename>
<args>"{1}" /monitor 2 /fullscreen /close</args>
<hidexbmc>false</hidexbmc>
<hideconsole>false</hideconsole>
<warpcursor>none</warpcursor>
</player>
</players>
<rules action="prepend">
<rule filetypes="avi|flv|mkv|mov|mp4|m2ts|ts|vob|wmv" player="MPC-HC"/>
</rules>
</playercorefactory>


MPC-HC + madVR + LAV + ArcSoft or CyberLink + XBMC = :)

:thanks: to gabest, madshi, nevcairiel, and other developers.

Off topic but what is the purpose if the portable data file?

[Budtz]

Depends on your usage profile. Some formats are better supported in ffdshow, some are better supoorted in LAV Audio. ffdshow also has a lot of postprocessing features.

I play mostly hd mkv-files with dts sound or dolby digital. The resampeling to 48Khz in ffdshow seems like a nice postproccessing feture that should give audio a boost.

Also what excactly does a splitter do? All i know is i cant seem to switch between different subs/audio without one.

[nevcairiel]

HD audio always has at least a 48kHz sample rate already. I fail to see the point. Besides i let ReClock do resampling to 96000Hz before output to the AVR. :)

[madshi]

"shiny"
I think you've watched too much Firefly... :p

[CruNcher]

Hi Andy,

I have great news. I know we've had our differences in the past; however, I was hoping we can call it truce and try to repair our relationship by extending an olive branch... offering a pre-release MpegDemux.ax which sounds like you may be very interested in. This build is only hours old. I was given permission to share this demux for people who needed it. I've been working directly with the developer at sageTV who appears to be responsible for all code changes made to MpegDemux.ax. This demux should hopefully be included in the next sageTV beta.

This pre-release version of sageTV demux is now fullly compatible with Arcsoft Audio Decode HD (full support for DTS-MA/TrueHD decoding without a hitch). In addition to that, it now also supports LAV Audio Decoder's TrueHD. If you have the arcsoft audio decoder package installed, all you have to do it tell sageTV to enable HD audio support and specify a single audio decoder: Arcsoft Audio HD. If you dont have arcsoft, you can at least have TrueHD audio decoding via LAV Audio decoder. My sageTV client has it's internal demuxing off; everything but bluray audio formats (and I think TV).

You might have noticed an earlier post from me today playing back a sample 7.1/96khz m2ts file directly in sageTV.
[/URL]

I got the same result no matter which container I used . I also had the same results with DTS-MA. Of course, it also works just as great with blu-ray discs/iso's as well on on my extender.

Here's the file:
http://hotfile.com/dl/117954203/045f397/MpegDeMux.zip

I've only tested this with TMT3 standalone decoder package (doesn't require to install the entire player just to be able to have access to the audio filter... only a small change in the Windows win.ini file:

1) Open up C:\Windows\win.ini in notepad.
2) Add the following text to the end:
Code:

[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=6
AudioDynamicRange=1

3) Change the numbers to match the settings you prefer.

Code:

AudioDynamicRange:
0: Light
1: Normal
2: Threaded

AudioOutputMode:
0,1,2,5: Stereo
6,13: 5.1 Channels
17: 7.1 Channels
16: S/PDIF

Sincerely,
MKANET

Hui i guess that one doesn't suffer from the Frame accurate seeking issues [url]http://forum.doom9.org/showpost.php?p=1499012&postcount=2642 (http://i67.photobucket.com/albums/h283/mkanet/Screenshot-1509.jpg) with Mainconcepts Decoder @ least i hope so :)

[nevcairiel]

I think you've watched too much Firefly... :p
Too much Firefly? Thats possible? :p

[mkanet]

There seems to be a problem with win.ini hack in Arcsoft standalone. It works perfect if you choose AudioOutputMode: 17 (for 7.1 full bitdepth and bitrate 24/96kz or above). However, if I try to downconvert in win.ini to 5.1 by using AudioOutputMode: 13 (or 6), DTS-MA is limited to 48khz.

I was hoping to let LAV audio to have a universal bitrate/bitdepth/speaker configuration of arcsoft audio HD DTS-MA or other audio formats.. so if I I can set speaker config to 5.1, bit depth to 24bit, and set bitrate to "unlimited". Maybe this is already in the works?

PS: What audio formats will LAV audio decoder allow Arcsoft audio decoder to use? Is it just limited to DTS-MA?

Thanks again,
MKANET

[SamuriHL]

Yes, just use the dtsdecoderdll.dll with LAV Audio and let it decode the Dolby formats itself. Best of all worlds. No more hacking anything.

[mkanet]

Thanks.. I'll try it out for sure.

Yes, just use the dtsdecoderdll.dll with LAV Audio and let it decode the Dolby formats itself. Best of all worlds. No more hacking anything.

[SamuriHL]

Basically just drop the dll in the LAV Audio folder and set LAV Audio to decode everything. Done.

[CruNcher]

Yes, just use the dtsdecoderdll.dll with LAV Audio and let it decode the Dolby formats itself. Best of all worlds. No more hacking anything.

Yep the other stuff is more performant (or as good) via ffmpeg and libav anyways only DTS-HD MA was lacking and some error resilience for latm ;)

[andyvt]

@nevcairiel - can you make a tray icon for LAV CUVID and LAV Filters? I mainly used WMP12 for movies and music, I just want to make sure if they're being used when playing audio/videos. Thanks.

If you do implement this feature please make it optional.

[Carpo]

Lav Audio + Arcsoft is working a treat here, the dtsma is streaming as normal dts, as my receiver cant handle HD Audio, can anyone suggest a replacement for a Z5500 setup? So i might be able to enjoy full HD Audio :)

[SamuriHL]

I took my 5500 speakers, replaced the sub, and added a Pio 820-k for my bedroom. :) Works awesome. IOW, just get a receiver and a new sub and use the 5500's speakers and you're good.

[dbone1026]

Lav Audio + Arcsoft is working a treat here, the dtsma is streaming as normal dts, as my receiver cant handle HD Audio, can anyone suggest a replacement for a Z5500 setup? So i might be able to enjoy full HD Audio :)

Onkyo 608, great price, HDMI 1.4

[Andy o]

You might wanna run some tone sweeps (foobar2000 or Audacity can generate them), cause those small satellites match with the sub pretty high up, probably higher than 200Hz. The DTS sample I posted (http://forum.doom9.org/showthread.php?p=1501399#post1501399) before has pink noise (for checking tone issues between channels) and a sweep that starts around 200Hz, but the sweep is only for the LFE channel. Some standalone subs don't go that high up.

[Boltron]

Quick question WRT to bitstreaming.

Dolby and DTS stuff all bitstream great with LAV Splitter/Audio. MKV's with PCM however don't. Is this normal?

I use latest MPC-HC, LAV Splitter/Audio, madVR, ATI 5670 HDMI. I use MKVs exclusively.

Thanks.

[Andy o]

What do you mean with "bitstream" PCM? Do you need your channels to switch automatically (e.g. 7.1 to 5.1, etc)? If so, use ReClock and WASAPI mode.

[SamuriHL]

You might wanna run some tone sweeps (foobar2000 or Audacity can generate them), cause those small satellites match with the sub pretty high up, probably higher than 200Hz. The DTS sample I posted (http://forum.doom9.org/showthread.php?p=1501399#post1501399) before has pink noise (for checking tone issues between channels) and a sweep that starts around 200Hz, but the sweep is only for the LFE channel. Some standalone subs don't go that high up.

If this is directed at me, I'm not terribly concerned. I mean, we're talking about my bedroom setup. The fact that I have a receiver and 5.1 surround in there at all is pretty sick. :D But it's nice for watching movies at night. I don't expect it to match my living room setup. :)

[SamuriHL]

Quick question WRT to bitstreaming.

Dolby and DTS stuff all bitstream great with LAV Splitter/Audio. MKV's with PCM however don't. Is this normal?

I use latest MPC-HC, LAV Splitter/Audio, madVR, ATI 5670 HDMI. I use MKVs exclusively.

Thanks.

If by bitstream PCM you mean exclusive mode, then you're correct. PCM is sent through Windows Audio. However, that depends on your audio renderer not the decoder, so, you need an audio renderer that can use exclusive mode to bypass the Windows Audio settings. ReClock is one such audio renderer.

[bjd]

PS: What audio formats will LAV audio decoder allow Arcsoft audio decoder to use? Is it just limited to DTS-MA?

LAV is happy pumping out DTS 96/24 as well using the Arcsoft Decoder. If the status tab shows 24bit int for the decode/output, you are utilising the Arcsoft Decoder. 32bit float means ffmpeg

[nevcairiel]

All DTS formats will be decoded by the ArcSoft decoder if its installed.

[CruNcher]

Hi Andy,

I have great news. I know we've had our differences in the past; however, I was hoping we can call it truce and try to repair our relationship by extending an olive branch... offering a pre-release MpegDemux.ax which sounds like you may be very interested in. This build is only hours old. I was given permission to share this demux for people who needed it. I've been working directly with the developer at sageTV who appears to be responsible for all code changes made to MpegDemux.ax. This demux should hopefully be included in the next sageTV beta.

This pre-release version of sageTV demux is now fullly compatible with Arcsoft Audio Decode HD (full support for DTS-MA/TrueHD decoding without a hitch). In addition to that, it now also supports LAV Audio Decoder's TrueHD. If you have the arcsoft audio decoder package installed, all you have to do it tell sageTV to enable HD audio support and specify a single audio decoder: Arcsoft Audio HD. If you dont have arcsoft, you can at least have TrueHD audio decoding via LAV Audio decoder. My sageTV client has it's internal demuxing off; everything but bluray audio formats (and I think TV).

You might have noticed an earlier post from me today playing back a sample 7.1/96khz m2ts file directly in sageTV.


I got the same result no matter which container I used . I also had the same results with DTS-MA. Of course, it also works just as great with blu-ray discs/iso's as well on on my extender.

Here's the file:
http://hotfile.com/dl/117954203/045f397/MpegDeMux.zip

I've only tested this with TMT3 standalone decoder package (doesn't require to install the entire player just to be able to have access to the audio filter... only a small change in the Windows win.ini file:

1) Open up C:\Windows\win.ini in notepad.
2) Add the following text to the end:
Code:

[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=6
AudioDynamicRange=1

3) Change the numbers to match the settings you prefer.

Code:

AudioDynamicRange:
0: Light
1: Normal
2: Threaded

AudioOutputMode:
0,1,2,5: Stereo
6,13: 5.1 Channels
17: 7.1 Channels
16: S/PDIF

Sincerely,
MKANET

As expected the splitter works fine @ seeking but it fails on some of the more (exotic) .ts so in those regards it's no competition (Mainconcept Demuxer code based) ;)

[clsid]

When you implement channel downmixing, can you add an option to boost the volume of the center channel?

A common problem with downmixing 5.1 AC3 (and probably other formats as well) to stereo is that the volume level of the sounds that originate from the center channel (typically voices) are way too low compared to the volume levels of the sounds coming from other channels.

[nevcairiel]

If possible, i would ask the decoder to downmix AC3, as it usually carrys special matrix information how to downmix into stereo.

Anyhow, i won't be doing downmixing yet, i rather work on improving sound quality, rather then destroying it.

[Boltron]

If by bitstream PCM you mean exclusive mode, then you're correct. PCM is sent through Windows Audio. However, that depends on your audio renderer not the decoder, so, you need an audio renderer that can use exclusive mode to bypass the Windows Audio settings. ReClock is one such audio renderer.

Ah, I will try it with ReClock then. Thank you.

[mkanet]

Hi Nav, I think I must be missing something very fundamental here; or maybe not doing something right. I just tried testing LAV splitter/ LAV audio (version .26) with dtsdecoderdll.dll for the first time. For some reason I only see the new DTS, DTS-HD under SPDIF/HDMI. I dont see them under "Formats" section. I have no problems making a filtergraph for TrueHD, AC3, and everything else in the "Formats" section of the LAV audio filter prop screens. But I can't use LAV audio filter for DTS-MA.

I've been using Arcsoft Audio Decoder HD (ASAudioHD.ax) using the other method with win.ini to decode all possible audio formats (including dolby digital plus) at 24 bit/48kz (and above) to my 5.1 analog speakers.

Thanks,
-MKANET

[Andy o]

If "dca" is enabled in the LAV Audio options and the Arcsoft dll is in the LAV folder, then it should be working. You can double check in the LAV Audio status tab. It should say "dts" or some variant depending of the audio of course. If it says "dca" then it's not using Arcsoft.

[Gleb Egorych]

I play mostly hd mkv-files with dts sound or dolby digital. The resampeling to 48Khz in ffdshow seems like a nice postproccessing feture that should give audio a boost.

I think for the purposes there is no point to switch from ffdshow to LAV Audio.

Resampling itself does not improve audio quality, in some cases it helps to workaround bugs in audio drivers or hardware.

[mkanet]

Thanks, its now able to decode. In Win.ini, I used to set speaker output to 5.1. IIRC, Nav mentioned that Win.ini isn't necessary anymore. How do I set speaker configuration to 5.1 then? Also, is there an option for Dolby digital plus decoding? I have all the checkboxes now selected under formats. No big deal if its not supported, just curious, since that format is also supported under Arcsoft Audio Decoder HD.

If "dca" is enabled in the LAV Audio options and the Arcsoft dll is in the LAV folder, then it should be working. You can double check in the LAV Audio status tab. It should say "dts" or some variant depending of the audio of course. If it says "dca" then it's not using Arcsoft.

[SamuriHL]

Thanks, its now able to decode. In Win.ini, I used to set speaker output to 5.1. IIRC, Nav mentioned that Win.ini isn't necessary anymore. How do I set speaker configuration to 5.1 then? Also, is there an option for Dolby digital plus decoding? I have all the checkboxes now selected under formats. No big deal if its not supported, just curious, since that format is also supported under Arcsoft Audio Decoder HD.

You can add ffdshow audio processor to downmix to 5.1. Don't confuse decoding with playing, as there is a marked difference between the two. You WANT the full number of channels decoded properly. Then you want to downmix that to whatever number of channels you want. I recommend NOT allowing windows audio to do that. You could force it through ReClock and have it handle it for you. Use ReClock as the audio renderer and then you can set it for 5.1.

[mkanet]

Hmm... I thought that arcsoft audio decoder did the downmixing (set in win.ini)... originally a registry setting for the TMT player) in addition to decoding audio.
[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=13
AudioDynamicRange=1

I already use reclock WASAPI. It seems like one of the other dlls in the arcsoft codec folder might be handling the downmixing part.

You can add ffdshow audio processor to downmix to 5.1. Don't confuse decoding with playing, as there is a marked difference between the two. You WANT the full number of channels decoded properly. Then you want to downmix that to whatever number of channels you want. I recommend NOT allowing windows audio to do that. You could force it through ReClock and have it handle it for you. Use ReClock as the audio renderer and then you can set it for 5.1.

[SamuriHL]

Hmm... I thought that arcsoft audio decoder did the downmixing (set in win.ini)... originally a registry setting for the TMT player) in addition to decoding audio.
[HKEY_LOCAL_MACHINE\SOFTWARE\ArcSoft\MPEG Codec\Decoder Setting\]
AudioOutputMode=13
AudioDynamicRange=1

I already use reclock WASAPI. It seems like one of the other dlls in the arcsoft codec folder might be handling the downmixing part.

You can allow the ArcSoft decoder to do the downmixing using the settings you're referring to. I don't agree that it's the right thing to do, however. I think it really belongs to post processing or the renderer to do the downmixing. Decoding, in my mind, should always JUST decode. And provide post processing and/or the renderer with unaltered, decoded audio. Period. That's just my opinion and is what LAV Audio currently does.

[nevcairiel]

The ArcSoft DirectShow component controlled the downmixing. Those settings have no effect on LAV Audio.

[Andy o]

Thanks, its now able to decode. In Win.ini, I used to set speaker output to 5.1. IIRC, Nav mentioned that Win.ini isn't necessary anymore. How do I set speaker configuration to 5.1 then? Also, is there an option for Dolby digital plus decoding? I have all the checkboxes now selected under formats. No big deal if its not supported, just curious, since that format is also supported under Arcsoft Audio Decoder HD.

If the content is 5.1, you'll get 5.1 out of the Arcsoft decoder, and for stereo you'll get stereo. The "7.1" setting is for max channels, but it doesn't always output 7.1. It works that way with TMT3 too, actually, and I presume TMT5 as well.

For actual 7.1 content, you can downmix either with Windows (when you set the mixer to 5.1) or with your receiver. The Windows mixer allows to even set 2 "virtual" surround backs, so you can do that, which is better than just setting it to 5.1.

[Mercury_22]

When you implement channel downmixing, can you add an option to boost the volume of the center channel?

A common problem with downmixing 5.1 AC3 (and probably other formats as well) to stereo is that the volume level of the sounds that originate from the center channel (typically voices) are way too low compared to the volume levels of the sounds coming from other channels.
+1 but for all channels (volume controls for each channel like in FFDshow)

[Andy o]

You can allow the ArcSoft decoder to do the downmixing using the settings you're referring to. I don't agree that it's the right thing to do, however. I think it really belongs to post processing or the renderer to do the downmixing. Decoding, in my mind, should always JUST decode. And provide post processing and/or the renderer with unaltered, decoded audio. Period. That's just my opinion and is what LAV Audio currently does.

There are metadata for proper downmixing in the DTS and Dolby codecs though, and once you've already decoded, it would be hard for the post-processor or renderer to downmix as the encoder intended. I don't see real-world problems with just letting Windows downmix though, but then you have to ditch WASAPI exclusive.

[mindbomb]

question, perhaps slightly off topic since it's likely an arcsoft issue, but do you guys notice that with the arcsoft dll, for anything outside of 6 channels, it can intelligently choose the sample format, but for 6 channels, it always uses 24 bit pcm?

is there a a technical explanation for that?

[VipZ]

Nev, with some more general use of 0.26 I find that in general the new AV sync option for AC3 bitstreaming causes sync issues. I think DTS is ok, but haven't done much checking on that one.

[SamuriHL]

There are metadata for proper downmixing in the DTS and Dolby codecs though, and once you've already decoded, it would be hard for the post-processor or renderer to downmix as the encoder intended. I don't see real-world problems with just letting Windows downmix though, but then you have to ditch WASAPI exclusive.

Yea, I guess I see your point. Windows Audio, however? Not a chance in hell in my setup. I don't trust it. Plus it'll upsample (I would hope! ;)) the audio because you'll at least want it set to 96/24. So a 48/16 native track like Kill Bill would get upsampled to 96/24. Not such a bad thing as it's really just padding, but, unnecessary IMO. Personally in my setup I just decode the audio with the correct number of channels and then let my Pio deal with it. :)

[nevcairiel]

Nev, with some more general use of 0.26 I find that in general the new AV sync option for AC3 bitstreaming causes sync issues. I think DTS is ok, but haven't done much checking on that one.

So when turning it off, you think its better?

If so, then its all in your mind, AC3 is perfectly packed into all containers, the sync option will not affect it at all, unless there is an actual sync problem.
I have not seen a single AC3 file that had any jitter whatsoever, so there is no chance the sync option gets confused (unless you have a special file again). The only possibility would be that my duration calculations for AC3 bitstreaming are wrong, but that would show no matter if the sync option is on or not (same problem DTS suffered from on your special file, but maybe not so severe)

AC3 is really one of the easiest formats to get right, especially because everyone knows how to mux it perfectly, so i don't see any margin for error there.

I should really figure out a way for you to produce debug logs. Every action the AV sync option takes is output there, if it does anything at all.

[Andy o]

Yea, I guess I see your point. Windows Audio, however? Not a chance in hell in my setup. I don't trust it. Plus it'll upsample (I would hope! ;)) the audio because you'll at least want it set to 96/24. So a 48/16 native track like Kill Bill would get upsampled to 96/24. Not such a bad thing as it's really just padding, but, unnecessary IMO. Personally in my setup I just decode the audio with the correct number of channels and then let my Pio deal with it. :)

The Pioneers have a similar "virtual" back channel option than what I mentioned about the Windows mixer, and I would use it if I only ran 5.1 channels. I see Windows's mixer getting a bad rap almost everywhere, but really, I haven't seen much evidence of it messing up anything substantial. Resampling (changing sample rate) is not just padding though, but again, haven't seen evidence that Windows's resampling is bad at all. I don't think it's that hard to do good resampling these days anyway, and even then, you can resample with ReClock to whatever Windows is set at, and it won't do again even if not using WASAPI exclusive.

My main reason not to use it is the channel switching, but that's more of a problem when you're running 7.1 channels. If I was using just 5.1 I'd probably be happy with DirectSound.

[VipZ]

So when turning it off, you think its better?

If so, then its all in your mind, AC3 is perfectly packed into all containers, the sync option will not affect it at all, unless there is an actual sync problem.
I have not seen a single AC3 file that had any jitter whatsoever, so there is no chance the sync option gets confused (unless you have a special file again). The only possibility would be that my duration calculations for AC3 bitstreaming are wrong, but that would show no matter if the sync option is on or not (same problem DTS suffered from on your special file, but maybe not so severe)

Yep, for me bitstream with it off is the same as decoded with it on, I find it quite clear that the sync is bit off with it on when bitstreaming. If I had to guess I would put it at around 400ms.

Edit: I have VS installed, so just let me know on what debugging or whatever you need me to do :)

[Andy o]

question, perhaps slightly off topic since it's likely an arcsoft issue, but do you guys notice that with the arcsoft dll, for anything outside of 6 channels, it can intelligently choose the sample format, but for 6 channels, it always uses 24 bit pcm?

is there a a technical explanation for that?

I don't know why, might be a harmless bug, but I can confirm it, at least with 5.1 vs 7.1 16-bit content.

[SamuriHL]

The Pioneers have a similar "virtual" back channel option than what I mentioned about the Windows mixer, and I would use it if I only ran 5.1 channels. I see Windows's mixer getting a bad rap almost everywhere, but really, I haven't seen much evidence of it messing up anything substantial. Resampling (changing sample rate) is not just padding though, but again, haven't seen evidence that Windows's resampling is bad at all. I don't think it's that hard to do good resampling these days anyway, and even then, you can resample with ReClock to whatever Windows is set at, and it won't do again even if not using WASAPI exclusive.

My main reason not to use it is the channel switching, but that's more of a problem when you're running 7.1 channels. If I was using just 5.1 I'd probably be happy with DirectSound.

I have a bad taste in my mouth when it comes to Windows Audio. Probably because I suffered through TMT3's evolution of decoding. And have seen numerous problems with PDVD's channel mapping over the years, as well. Is that a Windows Audio problem? Not entirely, but, it certainly didn't help matters. For me I want it simple...decode audio, send decoded audio *AS IS* to the receiver. I trust my Pio. I don't trust Windows. That's why I've preferred bitstreaming all these past few years. However, I'm ok with decoding if I can get the decoded PCM to the receiver untouched. For that I need exclusive mode. ReClock is working well for that right now and MC16 is something I need to test more. In any case, I'm good. :)

[nevcairiel]

Yep, for me bitstream with it off is the same as decoded with it on, I find it quite clear that the sync is bit off with it on when bitstreaming. If I had to guess I would put it at around 400ms.

Like i said, i don't think thats possible, unless you have some kind of broken file that i have not encountered before.

The A/V sync option is not constantly adjusting things, its just monitoring, and keeping track of things. If it sees a drift in timestamps, it'll adjust. But it really has to go bad for that to happen.

Maybe i should've named it differently, so peoples minds dont directly associate it with any sync issues they think to see. Its a funny thing, really. To a certain degree, if you think there are issues, you'll see issues.

Does it happen for all AC3 files?

You're compiling now, right?
You can launch MPC-HC from within Visual Studio, and get me a debug log.

- Set workspace profile to debug
- Set LAV Audio as startup project (right click on it in the solution explorer)
- Open LAV Audio project properties, in the Debugging tab (under Configuration Properties), set the MPC-HC executable as "Command"

Remember to register the output file with regsvr32 first, don't move it, it has to stay in the output folder (bin_Win32d)

Launch, open file, watch the Output box. All the thread infos and dll loading messages you can ignore, but during playback LAV Audio will say something like "Jitter Stats: min: 0 - max: 0 - avg: 0", 0's everwhere are perfect.
If the A/V sync option does somethign, it'll say "::Deliver(): corrected A/V sync by <number>"

Look for those.

[VipZ]

Ok I will do that and let you know.
With the sync at 1st its always ok, but at some stage it does seem to change, its not a drifting sync. I will watch something with it on and keep my eye on the debugging to see if the event you mentioned shows.

[nevcairiel]

I'll try to get soem debug file logging going.

[VipZ]

Here is the debug log http://www.mediafire.com/?y96sec6k4p9pbgw

There weren't any AV sync messaged you asked to look out for, maybe that log will prove it my mind playing tricks on me or there is a possible issue :)

Surprised I got it going without much effort :p

[nevcairiel]

Yeah that looks perfect, jitter is constantly 0, no A/V sync messages. No other warnings or errors.

The A/V sync option has no influence whatsoever in that log, and from what i can see, sync should just be perfect.

[VipZ]

I guessed as much, maybe the file is just bit bad or my mind is playing ticks on me.

[nevcairiel]

If its just the file, it doesn't make sense that toggling the option changes anything ;)

[VipZ]

lol fine, then it must be me :p

[assassinhtpc]

Any issues with live tv files?

[nevcairiel]

For the future, i just added debug logging into a file to LAV Splitter and LAV Audio. Both will dump their log into LAVAudio.txt and LAVSplitter.txt respectively, on your Desktop. Of course only when running debug builds, but no longer required to run it from a debugger or anything like that.

I'll be supplying debug versions of those two dlls with the next release as well then.

PS: The log files will be truncated every launch, so only one run will be shown in there. But at least the files don't grow excessively. ;)

[VipZ]

Cool, at least I got to play with debugging in VS before you made it nice and easy.

Is there any negative aspect to run the debug builds most of the time?

[glorp]

I apologize if this is a known problem but the thread is long and the changes are fast :)

I have a couple of mkv files with a 768k DTS 1.0 (core only no -HD MA) stream in them (old ~1968-1970 films that originally had mono PCM audio). These DTS audio streams don't decode with LAV Audio on PC with a 2.0 speaker system. I can bitstream it with LAV Audio on a different PC that goes to a receiver with 5.1 configuration. I can decode the DTS 1.0 stream to the 2.0 speakers when I use MPC's internal DTS codec or AC3Filter just fine, but it's a total no-go with LAV. MPC just hangs and never starts playing the file at all. Interestingly the Arcsoft decoder from TMT2 (1.1.0.8) can't handle it either. At least when I try to decode the stream to PCM or wav with eac3to, it throws an error and quits.

So is this just me or is it a bug?

Oh, and stellar job on the filters in general. Incredible effort.

[e-t172]

I have a bad taste in my mouth when it comes to Windows Audio. Probably because I suffered through TMT3's evolution of decoding. And have seen numerous problems with PDVD's channel mapping over the years, as well. Is that a Windows Audio problem? Not entirely, but, it certainly didn't help matters. For me I want it simple...decode audio, send decoded audio *AS IS* to the receiver. I trust my Pio. I don't trust Windows.

It's true that the audio engine in XP had serious problems. However they rewrote the whole thing since Vista. It does all processing in 32-bit float, with a sampling rate of your choosing. It won't resample if it doesn't need to, and when it does, I've never heard anyone complain about its quality. It does proper dithering when converting to integer (according to the Microsoft blogs from the people who actually developed the thing). And I've yet to see proof that the new audio engine was/is responsible for any channel mapping or downmixing issues. In fact, I'm not aware of any issue, past or present, with the Vista audio engine, except maybe from gamers who don't like the fact Microsoft ditched all hardware processing capabilities (which we don't care about).

I'm not sure why TMT's audio processing, or your receiver's, would be more "trustable" than the Windows Audio engine.

[Budtz]

How do I get the ArcSoft DTS decoder. Can i get that separatly or do i get it by installing their player: TotalMedia Theatre 5?
And will LAV audio support DTS-HD on its own someday?

[jmone]

If you want to use the Arcsoft decoder you should buy TMT5. There is a TMT5 trail that you could download that would include the file for testing etc. LAV Audio will not be able to nativly decode DTS-HD till it is supported by the ffmpeg project and given the spec has not ever been relased publically it may be some time.

[SamuriHL]

It's true that the audio engine in XP had serious problems. However they rewrote the whole thing since Vista. It does all processing in 32-bit float, with a sampling rate of your choosing. It won't resample if it doesn't need to, and when it does, I've never heard anyone complain about its quality. It does proper dithering when converting to integer (according to the Microsoft blogs from the people who actually developed the thing). And I've yet to see proof that the new audio engine was/is responsible for any channel mapping or downmixing issues. In fact, I'm not aware of any issue, past or present, with the Vista audio engine, except maybe from gamers who don't like the fact Microsoft ditched all hardware processing capabilities (which we don't care about).

I'm not sure why TMT's audio processing, or your receiver's, would be more "trustable" than the Windows Audio engine.

Again, my distrust of these things is from past experience. Where the problem lies, I don't care. What I know is that if I simply decode and send that directly to the receiver without manipulation, I get perfect sound. Equivalent to bitstreaming...all channels mapped correctly, no resampling, etc. LAV Audio gives me this experience when I use a renderer that can use exclusive mode to bypass Windows Audio. It's probably true that Windows Audio itself is fine, but, that the drivers from AMD and nVidia have caused a lot of the problems I've dealt with in the past. Also buggy decoders which we beat on ArcSoft to fix. :D So the situation today is FAR different from when I first got into this game. I just prefer the audio to be as untouched as possible. Even if it does perfect resampling and channel mapping, why? If I don't need that to be done, why do it? For those that DO need downmixing, maybe doing it in Windows Audio is a good solution. I do it through ffdshow audio processor on my laptop. Nonetheless, LAV Audio gives me EXACTLY what I want from a decoder. Unmolested decoded audio. That's it. :)

[Andy o]

And I've yet to see proof that the new audio engine was/is responsible for any channel mapping or downmixing issues.

With all the 7.1 audio devices I've tried (Ati 4000, 5000, Nvidia Gtx460 and 9200mGS, Intel G35, 3 C-Media chipsets, various onboard), when your player sends 5.1 audio and the Windows mixer is set to 7.1, Windows sends the surrounds to the back surrounds, and keeps the sides silent, when it should do the opposite, according to Dolby literature, and pretty much every HDMI receiver out there.

[Mark_A_W]

With all the 7.1 audio devices I've tried (Ati 4000, 5000, Nvidia Gtx460 and 9200mGS, Intel G35, 3 C-Media chipsets, various onboard), when your player sends 5.1 audio and the Windows mixer is set to 7.1, Windows sends the surrounds to the back surrounds, and keeps the sides silent, when it should do the opposite, according to Dolby literature, and pretty much every HDMI receiver out there.


Does this with my Xonar too. IMO Windows gets this wrong. Consistently wrong, but wrong all the same.

Luckily, some helpful bugger put me onto the hidden DPLIIx in the soundcard drivers :)

[andyvt]

With all the 7.1 audio devices I've tried (Ati 4000, 5000, Nvidia Gtx460 and 9200mGS, Intel G35, 3 C-Media chipsets, various onboard), when your player sends 5.1 audio and the Windows mixer is set to 7.1, Windows sends the surrounds to the back surrounds, and keeps the sides silent, when it should do the opposite, according to Dolby literature, and pretty much every HDMI receiver out there.

Check your NVIDIA card again w/ 270.61.

[Andy o]

It's not the 7.1 surround channel swap that plagued Nvidia HDMI devices at least twice (they actually shipped broke, fixed it, broke it again, and fixed it again, last time I checked). It's a different issue that happens with 5.1 output by the player into Windows set at 7.1. When playing 7.1 content, all channels are OK.

Actually, because of the channel swap bug, with those broken Nvidia drivers you'd not get the effect of this bug on 5.1, but with 7.1 obviously the surround sies/backs would be swapped.

[betaking]

Hi is this a lav filters bug? lav filtsers mpeg and mpegts can not connect coreaac ffdshow(lpcm) and ffdshow aac (libfaad2) ffdshow ac3(liba52)! use ffdshow ac3(libavcodec) ffdshow aac(libavcodec) and use ac3filter (LPCM)no problem! and use other ts filter no problem too!

[DrNein]

Off topic but what is the purpose if the portable data file?

Portable creates the configuration files under the XBMC folder rather than Users. So it is just a preference and affects where to place the playercorefactory.xml edited for MPC-HC:

wiki.xbmc.org
To customize playback behaviour, users need to create an extra playercorefactory.xml file inside their "\UserData\" folder.

[nevcairiel]

Is there any negative aspect to run the debug builds most of the time?

Its a bit slower, otherwise should pretty much be the same.
The logfile shouldn't grow excessively, as like i said its cleared every launch. After watching a few hours of a movie, it might still be quite large, however.

[Carpo]

MPC-HC + madVR + LAV + ArcSoft or CyberLink + XBMC = :)
:thanks: to gabest, madshi, nevcairiel, and other developers.

Indeed, now i have it all working in xbmc (windows media center sucks!!!), i am reconsidering my move to a linux/mythtv setup :D

Just have to see how the current windows setup handles eac3

[andyvt]

It's not the 7.1 surround channel swap that plagued Nvidia HDMI devices at least twice (they actually shipped broke, fixed it, broke it again, and fixed it again, last time I checked). It's a different issue that happens with 5.1 output by the player into Windows set at 7.1. When playing 7.1 content, all channels are OK.

Actually, because of the channel swap bug, with those broken Nvidia drivers you'd not get the effect of this bug on 5.1, but with 7.1 obviously the surround sies/backs would be swapped.

the current driver doesn't push l/r surrounds to the l/r rear surrounds when playing 5.1 on a 7.1 system

[Carpo]

After more tests i can play dts-hd files as it defaults back to core dts, but no such luck with truehd, unless I've not set it right, suppose it will save me having to re-encode all those blurays with eac3/truehd :p (until i can afford a new receiver)

Still at least using lav i have been able to remove ffdshow :D

[SamuriHL]

Decode TrueHD to PCM.

[Carpo]

How would i do that with Lav, looking at it now, but i must be missing something

[nevcairiel]

Just don't enable Bitstreaming for TrueHD, and it'll be decoded.

[Carpo]

tried that and it still squeaks at me, wonder if its because im trying to play the .truehd file directly, will mux it and see

edit: yes muxing it works, just have to change receiver to DD PLII Movie otherwise it just shows stero, suppose its bearable till i can afford a HD receiver :)

Now i just have to decide if i want to redo them all and at what crf :p

[SamuriHL]

If it shows stereo that's what your Windows Audio is set to output. Try setting Windows Audio to 5.1.

[Carpo]

Using dfault windows drivers on windows 7 as asus hasnt updated them for my board, and i dont have any spare slots to put the audigy card i had back in here :( Looks like windows is only going to let me have 2 channels or spdif, will see if i can hunt down some drivers for another board with are supposed to be compatible and see what happens with them :(

[SamuriHL]

Well that sucks. :(

[nevcairiel]

If you're actually using S/PDIF (coaxial or optical), try using ReClock and setting it to encode to AC3, that way you can get 5.1 to your Receiver. S/PDIF itself only supports 2 channel PCM. For more you need HDMI.

[SamuriHL]

Oh right, I was thinking there was an analog connection. DOH. Yes, Nev is 100% correct.

[Andy o]

the current driver doesn't push l/r surrounds to the l/r rear surrounds when playing 5.1 on a 7.1 system

I opened a thread (http://www.avsforum.com/avs-vb/showthread.php?p=20453722#post20453722) at AVS to continue this, pls head on there.

[iSeries]

Hi,

Regarding dts-ma decoding with the arcsoft dll. With the 1.1.0.0 version, LAV audio is telling me my soundtrack is 7.1, but its actually 5.1. Where are these extra channels coming from? With the 1.1.0.8 version, if the movie gets loud I hear a lot of static (quieter parts ok). If anyone has any ideas about these 2 issues please let me know :thanks:

[nevcairiel]

1.1.0.0 is not fully supported yet. The next version will add support for this.

For the other issue, it may just be a bug in the decoder, as i do not process the output at all.
Once supported fully, 1.1.0.0 will most likely be the best version to use anyway.

[Carpo]

If you're actually using S/PDIF (coaxial or optical), try using ReClock and setting it to encode to AC3, that way you can get 5.1 to your Receiver. S/PDIF itself only supports 2 channel PCM. For more you need HDMI.

Any particular things i should look at, apart from AC3 encoding? seeing as i don't require the video part of it i can just leave it alone?

[Andy o]

Once supported fully, 1.1.0.0 will most likely be the best version to use anyway.

Why is this? BTW, from googling there seems to be a dtsdecoderdll.dll version 1.1.0.10 that came with "TMT 3.5", by which I think the poster meant TotalMedia 3.5. Apparently it doesn't work with eac3to, so I don't know if it could work with LAV.

[nevcairiel]

1.1.0.0 is the version eac3to was developed against, and it reproduces all audio perfectly. 1.1.0.7/8 drop the back-center in 6.0 layouts, and molest mono audio for no good reason.
I'm not sure when they changed the decoder, so i cannot say which versions between 1.1.0.0 and 1.1.0.7 have what bugs/features.

[Carpo]

so guessing TMT5 version is right out then

[nevcairiel]

Its fine, you just won't have a back center in 6.0 DTS files. Luckily, thats a rather rare layout. 6.1 works fine again.

[Carpo]

glad i saved it before i updated :)

[SamuriHL]

I'm using 1.1.0.8 and not having any issues. So, unless you care about that back center channel, meh.

[Gleb Egorych]

Version 1.1.0.0 came with TMT2. But TMT2 installs 2 dtsdecoderdll.dll-s, one to Common files and the other to the program folder, the first from 21.04.2008 and the second from 25.04.2008. They are 1.1.0.0 both, but binary different. nevcairiel, which particular version are you testing with? In that thread (http://forum.doom9.org/showthread.php?p=1484933#post1484933) mindbomb wrote about mono and 16 bit bugs in 1.1.0.7 which were fixed in 1.1.0.8. As I understood those bugs did not exist in 1.1.0.0, right?

[Markolc81]

I'm encountering a new problem, but seemingly not with the decoder. What happens is if I use LAVsplitter to decode DTS/TrueHD/whatever instead of bitstream, whenever I'm done playing my movie and go back into windows media Center there's no sound anymore (or sounds are very distorted). When this happens I can no longer play Live TV or Netflix or Boxee because they error out. However I can go right back into MPC-HC and play another movie just fine. If I restart the computer, then all works fine again until I play another MKV with lossless audio. I do use re-clock if that makes any difference.

Now If I have LAV Audio set to Bitstream, then all works perfect and Windows sounds (including WMC, Netflix, and Boxee) all work fine before and after I play an MKV. Anyone know what could be the problem?

I'd rather LAV Audio decode the lossless audio so I can maintain the benefits of ReClock.

[SamuriHL]

It's not a LAV Audio issue. It's ReClock and pretty much every other exclusive mode renderer that's causing it. It doesn't reset the audio connection when the exclusive mode lock is dropped. I personally feel that's a bug in the audio driver. (I'm guessing you're on AMD hardware?)

[Markolc81]

It's not a LAV Audio issue. It's ReClock and pretty much every other exclusive mode renderer that's causing it. It doesn't reset the audio connection when the exclusive mode lock is dropped. I personally feel that's a bug in the audio driver. (I'm guessing you're on AMD hardware?)

Yup. AMD Radeon 6850. AMD Sempron 140 cpu also.

[Markolc81]

It's not a LAV Audio issue. It's ReClock and pretty much every other exclusive mode renderer that's causing it. It doesn't reset the audio connection when the exclusive mode lock is dropped. I personally feel that's a bug in the audio driver. (I'm guessing you're on AMD hardware?)

However, when I play a movie that had lossless decoded to FLAC (an old BD rip of mine) then Reclock won't reproduce the same symptoms. Odd.

[SamuriHL]

However, when I play a movie that had lossless decoded to FLAC (an old BD rip of mine) then Reclock won't reproduce the same symptoms. Odd.

That is odd. One would think if ReClock is making an exclusive mode connection it wouldn't matter what the source is. It's when the exclusive mode lock is released that the trouble begins. TMT5 does the same thing when I bitstream, for example. The audio connection gets all screwed up. I generally go to the Audio CP and play the test sound or open Pandora to reset it. :D

[Andy o]

Are you using Media Center as a GUI to call up MPC-HC? If you're not using ReClock's WASAPI exclusive mode, and aren't using a 7.1 system, you can try the Realtek HDMI driver.

Also, try disabling MC menu sounds, or setting your Windows mixer to stereo, before you try Realtek.

[nevcairiel]

I think its a AMD problem with WASAPI or something like that, never happend on NVIDIA for me.

[SamuriHL]

I agree. I believe it's an AMD problem with the AMD HDMI driver. That's why I pegged what hardware he has.

[Markolc81]

Are you using Media Center as a GUI to call up MPC-HC? If you're not using ReClock's WASAPI exclusive mode, and aren't using a 7.1 system, you can try the Realtek HDMI driver.

Also, try disabling MC menu sounds, or setting your Windows mixer to stereo, before you try Realtek.

I'm using Media Browser to call on MPC-HC. And I have ReClock setup to output to WASAPI exclusive mode for PCM (WaveOut for Bitstream, but I don't think it makes a difference).

[Markolc81]

That is odd. One would think if ReClock is making an exclusive mode connection it wouldn't matter what the source is. It's when the exclusive mode lock is released that the trouble begins. TMT5 does the same thing when I bitstream, for example. The audio connection gets all screwed up. I generally go to the Audio CP and play the test sound or open Pandora to reset it. :D

I spoke too soon, I played some more MKVs with FLAC and it disabled the Windows sounds as well.

When I go into the Audio CP to play the test sound it just gives me an error lol. The only way to cure the problem is to reboot windows, or bitstream something (Live TV that happens to be streaming AC3, or set LAV audio to bitstream again.)

[Markolc81]

I agree. I believe it's an AMD problem with the AMD HDMI driver. That's why I pegged what hardware he has.

I'm gonna try the Realtek driver, if that doesn't work I'm going to go bother the folks on the ReClock forums about it, or back on AVS.

[SamuriHL]

It's not a ReClock problem. You won't get it resolved by SlySoft. Trust me. ;)

[nevcairiel]

There, i just commited a re-worked interface to the DTS decoder. It will now always reduce the decoded output to its native channel count and layout, matching the channel layout of the ffmpeg dca decoder.

Next step is a function to actually increase that channel count again to one of the standard formats properly recognized by your typical AVR. ;) (Mono/Stereo/5.1/6.1/7.1)

BTW, anyone sitting on 6.1 samples that are NOT dts? TrueHD? Other multi-channel formats?
I need to confirm the channel order..

[SamuriHL]

I don't think I've ever seen a non-5.1 or 7.1 TrueHD track. All the 6.1 stuff I've seen is DTS.

[Bryanhoop]

I agree. I believe it's an AMD problem with the AMD HDMI driver. That's why I pegged what hardware he has.

I've had the same issue happening to me on my nVidia HDMI output. I've posted about it a few times on INTERACT but no one said much. I also do the sound test to "free" the playback device.

[SamuriHL]

I've had the same issue happening to me on my nVidia HDMI output. I've posted about it a few times on INTERACT but no one said much. I also do the sound test to "free" the playback device.

Fascinating. I hadn't seen that issue on nVidia hardware before.

[Plutotype]

I experienced this only if the audio output device in Reclock was the same as the default in Windows and was used in WASAPI mode. After exiting MPC-HC, the Wasapi mode hanged in Windows, so I had to restart all other apps like browser, other players, audio apps to refresh the actual status. Windows somehow can not "tell" to all running apps, that the Wasapi mode ended its session and default audio device should be restored in realtime to shared mode.

Therefore Reclock nicely offers 2 ways od playback. Directsound ( can be the default audio output device - mostly 2.0 or headphones ) and Wasapi. Wasapi can be the HDMI audio ( or any other 24/96 capable surround analog output device - Realtek ALC889 for me ). I switch to Wasapi only if Im going to watch a full lenght movie.

[Andy o]

The Evangelion blu-rays are THD 6.1. I'm heading out to finally get my receiver upgraded, and then I'll have to do some direct connections to my display, I'll have some clips later if you still need them.

[Andy o]

I'm gonna try the Realtek driver, if that doesn't work I'm going to go bother the folks on the ReClock forums about it, or back on AVS.

The Realtek driver has other problems when using WASAPI exclusive in ReClock, so that probably is not a solution even if it fixes your problem, that's why I said "if you're not using ReClock's WASAPI...".

Did you try disabling the MC menu sounds, or setting your Windows mixer to stereo?

[nevcairiel]

The Evangelion blu-rays are THD 6.1. I'm heading out to finally get my receiver upgraded, and then I'll have to do some direct connections to my display, I'll have some clips later if you still need them.

Would be great to get a sample to confirm i'm not breaking 6.1 layout.

[Markolc81]

The Realtek driver has other problems when using WASAPI exclusive in ReClock, so that probably is not a solution even if it fixes your problem, that's why I said "if you're not using ReClock's WASAPI...".

Did you try disabling the MC menu sounds, or setting your Windows mixer to stereo?

My Windows mixer is always set to stereo because if it's 5.1 or 7.1 then stuff Like Netflix and Pandora only come out of the L and R speakers and my AVR would not allow me to add processing (like Pro-Logic II for a Netflix Movie).

But I have set it to 5.1 and 7.1 temporarily and it does not make a difference. :mad:

Also now even if I do have LAV Audio set to bitstream, sometimes Windows audio will be incredibly distorted after I exit playing a movie from MPC-HC...This is getting frustrating :mad:

I'm gonna go tinker some more and let you know what's up.

[jakmal]

BTW, anyone sitting on 6.1 samples that are NOT dts? TrueHD? Other multi-channel formats?
I need to confirm the channel order..

Not dts or "not dts or dts-hd" ?

The sample I uploaded in the bug report here: https://github.com/Nevcairiel/LAVFSplitter/issues/42 : has a 6.1 channel DTS-HD soundtrack.

[nevcairiel]

Not DTS at all, not HD or otherwise, but thanks. .)

I actually have samples for all conceivable DTS channel layouts now, and the only one that does not seem to work properly at all is 7.0. But who builds 7 channel without a LFE track anyway!

[Markolc81]

The Realtek driver has other problems when using WASAPI exclusive in ReClock, so that probably is not a solution even if it fixes your problem, that's why I said "if you're not using ReClock's WASAPI...".

Did you try disabling the MC menu sounds, or setting your Windows mixer to stereo?

Well you pegged it. I finally had a lil more time to play around and disabling MC sounds worked. I kinda miss the sounds in WMC but I'd trade it for a working system that's for sure. Now I can bitstream or have LAV Audio decode and windows sounds work fine.

[SamuriHL]

Oh, right, I forgot about that. I disabled 7MC sounds a long time ago on both my AMD and nVidia machines.

[Andy o]

Would be great to get a sample to confirm i'm not breaking 6.1 layout.

You don't need video, do you? I'll post mka clips.

[CruNcher]

@nevcariel

Multithread Encoding Results (Video only Mainconcept Mpeg-2 Decoder) :(

encoded 4531 frames, 29.04 fps, 25364.98 kb/s (Cyberlink Demuxer 2.0) only max 50% core utilization
encoded 4531 frames, 31.13 fps, 25364.98 kb/s (lav splitter 0.26) heavy core utilization fluctuation
encoded 4531 frames, 42.65 fps, 25364.98 kb/s (MPC-HC Splitter) steady core utilization
encoded 4531 frames, 42.79 fps, 25364.98 kb/s (Mainconcept Demuxer) steady core utilization
encoded 4531 frames, 43.10 fps, 25364.98 kb/s (Arcsoft Mpeg Demux) steady core utilization


I guess this is a result of using still big parts of the single threaded ffmpeg decoder ?

[Andy o]

Here's the TrueHD 6.1 sample (http://www.megaupload.com/?d=WKCB97ZA). BTW, if can anyone tell me if it sounds distorted? I think it's in the original audio (explosions and such).

[madshi]

BTW, anyone sitting on 6.1 samples that are NOT dts? TrueHD? Other multi-channel formats?
I need to confirm the channel order..
I've not seen 6.1 E-AC3. But 6.1 TrueHD should exist. You can create 6.1 WAV and FLAC files by decoding DTS with eac3to, of course.

[nevcairiel]

I guess this is a result of using still big parts of the single threaded ffmpeg decoder ?

I don't understand wtf you're on about. LAV Splitter is a splitter, not a decoder. It can without trouble output more then 40fps, seems like some limitation in your encoder.

I've not seen 6.1 E-AC3. But 6.1 TrueHD should exist. You can create 6.1 WAV and FLAC files by decoding DTS with eac3to, of course.

Thanks, i'll check that. Why didn't this occur to me..
BTW, isn't FLAC only strictly defined to up to 5.1?

[e-t172]

BTW, isn't FLAC only strictly defined to up to 5.1?

Some time ago I got FLAC files encoded from Blu-ray audio in 7.1/96kHz/24bit. Kinda extreme. When I tried to play them, I've found that the only player capable of handling them was AIMP. foobar2000, VLC, mplayer, CoreFLAC and vuplayer all failed to open the file.

[madshi]

isn't FLAC only strictly defined to up to 5.1?
No. FLAC supports 1-8 channels, and custom channel masks via metadata information. I somehow doubt that the ffmpeg decoder supports custom channel masks, though. eac3to and madFlac do, of course.

[nevcairiel]

I know it supports 8 channels, i just read somewhere that standard channel layouts are only defined for up to 5.1, and i didn't know about the layout metadata. Thanks for clearing that up.

[CruNcher]

I don't understand wtf you're on about. LAV Splitter is a splitter, not a decoder. It can without trouble output more then 40fps, seems like some limitation in your encoder.

hmm im not sure yet why but it behaves like that on my SB it loses 10 fps efficiency due to this (cores utilization fluctuates) with roozhous direct264 same happens with cyberlinks demuxer (cores get only 50% utilized never reach 100%). It is reproducible everytime not a 1 time thing always if either lav splitter or cyberlink demuxer 2.0 get used in those encoding chains the cores aren't fully utilized anymore.

http://forum.doom9.org/showthread.php?t=141441

I prepare a bigger test with different decoder and content currently, fully automated to gather results faster, could be some interoperability issue i guess as well between Mainconcepts Mpeg-2 Decoder and Cyberlinks and your Demuxer (maybe content and settings based).

[ontherocks]

Anyhow, i have a Onkyo TX-SR607, and DD+ works just perfectly for me - i would assume its somewhat similar decoding units in those two. ;)

Works fine on my 5450 to an Onkyo 876 also.

Yea I missed that page. The page I saw only listed DD, TrueHD but it didn't have DD+. Awesome. :) No idea why it's not working, either. Works fine with my Pio's.

DD+ good here on LAV->Reclock->HD5670->HDMI->TX-NR905

And my Denon 988. Just got a new AVR though a Yamaha V867 and will see how that pans out.

I have come to think that it may be a EDID issue in my case. If any of you guys could post the real time EDID of your receiver (especially the audio block), that will be great.

...also I've a ton of DD+ stuff from all my HD-DVD discs and these also play fine (though all are 5.1, no 7.1) with LAVAudio doing the decoding.
What you are talking about is decoding via LAV. I am talking about bitstreaming.
Moreover the issue is not that DD+ files don't play. In fact they play flawlessly in my system. The problem is the DD+ light doesn't show up in my receiver. What show up is just DD light (without the + sign, just like regular DD media), which makes me believe that its not bitstreaming DD+ correctly.

[SamuriHL]

I have come to think that it may be a EDID issue in my case. If any of you guys could post the real time EDID of your receiver (especially the audio block), that will be great.


Could be. I suggest visiting the EDID thread on AVSForum and get tulli or Vladd to help you out.


What you are talking about is decoding via LAV. I am talking about bitstreaming.
Moreover the issue is not that DD+ files don't play. In fact they play flawlessly in my system. The problem is the DD+ light doesn't show up in my receiver. What show up is just DD light (without the + sign, just like regular DD media), which makes me believe that its not bitstreaming DD+ correctly.

The easiest way to verify that is to try Andy_O's 7.1 DD+ sample linked to earlier in the thread. If you get 7.1, you're bitstreaming DD+.

[ontherocks]

Could be. I suggest visiting the EDID thread on AVSForum and get tulli or Vladd to help you out.
I know what to do with EDIDs, that's not the problem.
I am also sure that my EDID is correct. I just want to see others EDIDs if anything is unusual.

The easiest way to verify that is to try Andy_O's 7.1 DD+ sample linked to earlier in the thread. If you get 7.1, you're bitstreaming DD+.
My receiver is 5.1, so cannot check for 7.1. The only way to confirm if working correctly I guess is the DD+ light showing up.

[SamuriHL]

Well that's a good time. I don't have things setup on my machine to capture the EDID info right now so I'm no help there. I hope you're able to get this resolved.

[nevcairiel]

E-AC3 is not backwards compatible with AC3. If you have a true E-AC3 stream, and you get audio just fine, then it is decoding E-AC3, no matter what any light on the receiver says. E-AC3 does not have a AC3 "core" or something like that.

The exception are Blu-rays, they have a 5.1 AC3 track, and ship an E-AC3 substream, which contains up to 4 channels, and if supported, replaces the AC3 rear channels with its own, as well as adding up to two additional ones for 7.1. But E-AC3 on Blu-rays is really not used at all, its probably only on the Dolby demo disc, and no-where else.

True E-AC3 content is usually found on HD-DVDs, i dunno where else its really in use.

[ontherocks]

E-AC3 is not backwards compatible with AC3. If you have a true E-AC3 stream, and you get audio just fine, then it is decoding E-AC3, no matter what any light on the receiver says. E-AC3 does not have a AC3 "core" or something like that.
I too think so. May be a bug in the firmware. Since you have an Onkyo TX-SR607 which has the same display as mine, could you tell me what lights show up when you play DD+ files from here http://www.demo-world.eu/trailers/high-definition-trailers.php

its probably only on the Dolby demo disc, and no-where else.Any link which demo disc you are talking about? I would like to have a look.

True E-AC3 content is usually found on HD-DVDs, i dunno where else its really in use.
Do you have any such samples? I would like to play around with them.

[Andy o]

Any link which demo disc you are talking about? I would like to have a look.

The DD+ and TrueHD samples on that page you linked apparently are from that blu-ray, unless a demo HD-DVD came out as well that had those trailers. I don't know if the ones uploaded there have been messed with (remuxed, re-encoded video), but I uploaded the m2ts here (http://www.megaupload.com/?d=A18UUKPN) before.

[noee]

@nev,
Last code looks good for first blush, for DTS 5.1 and 7.1 WASAPI and DS. BTW, does DTS-ES count as a legit 6.1 source?

fwiw, there was a syntax error in dtsdecoder.cpp when I did a pull. Not sure if it was my problem or not, but I haven't messed with the decoder code....

- } else if (header.ChannelLayout == 9 && !header.LFE && header.XChChannelLayout) {
+ if (header.ChannelLayout == 9 && !header.LFE && header.XChChannelLayout) {

[nevcairiel]

fwiw, there was a syntax error in dtsdecoder.cpp when I did a pull. Not sure if it was my problem or not, but I haven't messed with the decoder code....

- } else if (header.ChannelLayout == 9 && !header.LFE && header.XChChannelLayout) {
+ if (header.ChannelLayout == 9 && !header.LFE && header.XChChannelLayout) {

Maybe you hit some weird interim version, i dont have any local changes anymore right now, and everything builds fine.

And yes, of course DTS-ES 6.1 is "legit", i don't see why not.
Although, 6.1 in general is so messed up, every decoder seems to handle it differently. Some put the side channels before the BC, some after, its really a mess.

HDMI seems to want the side channels before the BC channel, so i have to shuffle things around, and cannot trust the decoders. :/

I bet the next version will produce funny problems with some of the more "weird" channel layouts, its just impossible to cover all cases.
Although the default layouts like mono, stereo, 5.1, 6.1 and 7.1 should hopefully still work like before.

Sadly, i cannot test native 6.1 content, as my HTPC and AVR dont seem to be on speaking terms when that format is active.
I'll put a test version up tomorrow when i have the options finished, and hopefully someone else can test the 6.1 content on their setup to see if its going out the right channels. ;)

[jakmal]

Do you have any such samples? I would like to play around with them.

Check the file uploaded here:

https://github.com/Nevcairiel/LAVFSplitter/issues/48

[nevcairiel]

That looks like one of those rare DD+ on Blu-ray files. You wouldn't happen to have gotten that from the Dolby Demo Blu-ray? :p
Looks alot like a demo file, tbh. :)

I haven't actually seen a real BD with E-AC3. Like i said, E-AC3 on Blu-rays is special, and may or may not be supported when bitstreaming. I don't really plan to invest much time into this, as its a practically non-existent format.

[madshi]

FWIW, E-AC3 is used on some Blu-Rays for PiP commentary tracks. That's a whole different thing, though. Those tracks have the same format as HD DVD E-AC3 tracks (no AC3 core).

[nevcairiel]

I also haven't seen those on a live BD, all secondary audio tracks i found are DTS Express.

[jakmal]

That looks like one of those rare DD+ on Blu-ray files. You wouldn't happen to have gotten that from the Dolby Demo Blu-ray? :p
Looks alot like a demo file, tbh. :)

I haven't actually seen a real BD with E-AC3. Like i said, E-AC3 on Blu-rays is special, and may or may not be supported when bitstreaming. I don't really plan to invest much time into this, as its a practically non-existent format.

Yes, it is from the BD demo disc.

The OTT services like Netflix and Vudu are using DD+ for their streams in some cases, so maybe it will start gaining traction for physical media ?

I thought you mentioned that one of the advantages of LAV Audio over ffdshow is the fact that LAV Audio actually bitstreams EAC3 :)

Btw, IIRC, that demo file bitstreamed perfectly with 0.25 and started giving issues only in 0.26 .

[madshi]

I have seen quite a few Blu-Rays with E-AC3 PiP tracks. E.g. Transformers 1, IIRC.

[madshi]

Yes, it is from the BD demo disc.
I wish that stupid BD demo disc would never have seen the light of day. Do you really want nevcairiel to spend hours of his time supporting a format which has only ever been used on exactly 1 demo BD disc world wide? There's not a single movie studio which has *ever* used this format.

IMHO every user who posts a sample from that BD demo disc should be banned for a week. :devil: Just kidding, of course...

The OTT services like Netflix and Vudu are using DD+ for their streams in some cases
I doubt they're going to use the stupid Blu-Ray 7.1 E-AC3 logic, though. They're most probably using the HD DVD 5.1 + 7.1 E-AC3 logic which is supported by LAV Splitter / Audio Decoder just fine.

[CruNcher]

Ahh about the encoding issue the same happens @ fastforward in MPC-HC @ 2x i get only 30 fps max not 50 fps as you would expect from that 25fps .ts (Mainconcept Decoder) so something is locking cyberlink demuxer and lav splitter up to 30 fps seems to only happen with them.

[SamuriHL]

There is NO WAY that broadcasts and other streaming services are going to use BD's implementation. Seriously, I agree here....not useful to spend time on this. If anything, I'd prefer to see the effort spent on getting the Dolby decoder dll's from ArcSoft working if you're going to even bother.

[Andy o]

Is there even a 7.1 E-AC3 in existence using HD-DVD's method?

[SamuriHL]

Not that I've ever seen. And broadcasts are never going to use it, either. They'll stick to 5.1. At least for the forseeable future.

[jakmal]

I wish that stupid BD demo disc would never have seen the light of day. Do you really want nevcairiel to spend hours of his time supporting a format which has only ever been used on exactly 1 demo BD disc world wide?

There is NO WAY that broadcasts and other streaming services are going to use BD's implementation. Seriously, I agree here....not useful to spend time on this. If anything, I'd prefer to see the effort spent on getting the Dolby decoder dll's from ArcSoft working if you're going to even bother.

Sorry guys :) Being a verification engineer IRL, I always try to find corner cases where the design doesn't work.. My main concern with respect to filing that issue was that it used to work previously..

[CruNcher]

No go it always locks to 30 fps no difference what you set, it seems like a hard coded parser force, it happens also with other decoder so it seems to be a lav splitter issue it's interesting that Cyberlinks Demuxer has the same problem, many others don't have this issue so it seems avoidable, it also makes fast forward 2x in MPC-HC impossible :(

[SamuriHL]

Sorry guys :) Being a verification engineer IRL, I always try to find corner cases where the design doesn't work.. My main concern with respect to filing that issue was that it used to work previously..

Understood, but, the real world issue is that Nev's time is limited and if he's screwing around with something that we'll never encounter outside of a demo disc, it takes away from real functionality that we can really use. So yes, it's a nice edge case and as a software engineer myself I appreciate those things, it's just not something that's worth giving time to. I'm sure Nev will feel the same way.

[whurlston]

I've not seen 6.1 E-AC3. But 6.1 TrueHD should exist. You can create 6.1 WAV and FLAC files by decoding DTS with eac3to, of course.

If I remember correctly, the DVE HD DVD has 6.1 DD+ and TrueHD samples. I would need to find my copy though.

[Mark_A_W]

There is no code whatsoever that forces any decoder. Blame ZoomPlayer, its "smart play" is usually not so smart.



nevcairiel

Pankov has done some testing for me, and he figured out that ZP is using the Filter Profile for E-AC3 instead of the Filter Profile for FLAC (for FLAC audio). I had LAV Audio chosen in the E-AC3 Profile - which is why it appeared to be forced to me.

It sounds like a Zoom Player issue. But it only happens with the LAV Splitter - it's a bit murky.

Here's what Pankov said to me:

"I've just tried the file and I do think that there is something wrong in ZoomPlayer.
Even though the media subtype is one that's specified in the FLAC profile it insists on loading the filter configured in the E-AC3 profile (added in v8.00 RC2 on my request) which has a totally different media subtype ({AFBC2343-3DCB-4047-9655-E1E62A61B1C5}). If you press "I" while playing the file you'll see the Media Information window and there at the bottom ZP states that it's E-AC3 !?!?"


So either is a ZP issue, and I apologise, or something about LAV Splitter is sending the wrong media type to ZP or something like that...(I've also posted this in the testing section on the inmatrix forum).

Anyway I'm glad Pankov was able to duplicate my issue, therefore I'm not insane (well, maybe a little...) :cool:


Sorry to give you and madshi the run-around.

Mark

[nevcairiel]

it also makes fast forward 2x in MPC-HC impossible :(

LAV Splitter does not support adjusting the playback rate. 1x, or get out! :p

My main concern with respect to filing that issue was that it used to work previously..

It most likely broke because i added the ffmpeg parsers to re-assemble frames, it may not know how to do that with the dependent E-AC3 substream. Otherwise there are no changes that would affect EAC3, it does not use the new timing logic.

Note that E-AC3 will not work with the MPC-HC splitter, as it claims that its AC-3.

I can run some quick tests on that.

[tetsuo55]

Understood, but, the real world issue is that Nev's time is limited and if he's screwing around with something that we'll never encounter outside of a demo disc, it takes away from real functionality that we can really use. So yes, it's a nice edge case and as a software engineer myself I appreciate those things, it's just not something that's worth giving time to. I'm sure Nev will feel the same way.FYI this was exactly ATI's opinion when they decided not to support any streams beyond the strictest L4.1 specs for DXVA h264, look where that brought us.

Other than that, i agree.

[nevcairiel]

Except, there really are no Blu-rays with their main audio in E-AC3 - with the one exception of the Dolby Demo Blu-ray, where they had to show-off their features - and i don't think there are free tools available to create such streams, unlike H264, which has x264 with which you can encode to any level. :p

Its also not like this is a strict limitation in some hardware chip or something, if there ever surface those discs, for whatever reason, i can still try to fix it.

look where that brought us.

To NVIDIA. :)

[madshi]

FYI this was exactly ATI's opinion when they decided not to support any streams beyond the strictest L4.1 specs for DXVA h264, look where that brought us.
The difference is that high spec h264 files were already widely used when ATI made that decision. Furthermore using higher spec h264 encoding can actually be useful sometimes. Both facts don't apply to Blu-Ray style E-AC3 tracks. They're junk, not useful, have never been used in real life yet and most likely never will be.

[tetsuo55]

I think thats a good benchmark.

Give us a commercial or homebrew sample and work on it can be started.
Imho EAC3 is a bad idea anyway.

[nevcairiel]

EAC3 itself is fine, it basically just extends AC3 from 5.1 to 7.1 and allows higher bitrates, no magic involved. Its similar enough to AC3 for decoders to adapt quickly, better then inventing a new format. Its basically the Dolby variant to the DTS-ES extensions or certain DTS-HD HRA profiles.

What they did to Blu-ray style E-AC3 tracks is another matter. To keep the tracks backwards compatible to AC3 decoders, they invented this stupid way of only using this 4 channel E-AC3 to replace the AC3 side channels, and add back channels.

[pankov]

nevcairiel

Pankov has done some testing for me, and he figured out that ZP is using the Filter Profile for E-AC3 instead of the Filter Profile for FLAC (for FLAC audio). I had LAV Audio chosen in the E-AC3 Profile - which is why it appeared to be forced to me.

It sounds like a Zoom Player issue. But it only happens with the LAV Splitter - it's a bit murky.

Here's what Pankov said to me:

"I've just tried the file and I do think that there is something wrong in ZoomPlayer.
Even though the media subtype is one that's specified in the FLAC profile it insists on loading the filter configured in the E-AC3 profile (added in v8.00 RC2 on my request) which has a totally different media subtype ({AFBC2343-3DCB-4047-9655-E1E62A61B1C5}). If you press "I" while playing the file you'll see the Media Information window and there at the bottom ZP states that it's E-AC3 !?!?"


So either is a ZP issue, and I apologise, or something about LAV Splitter is sending the wrong media type to ZP or something like that...(I've also posted this in the testing section on the inmatrix forum).

Anyway I'm glad Pankov was able to duplicate my issue, therefore I'm not insane (well, maybe a little...) :cool:


Sorry to give you and madshi the run-around.

Mark

Mark, Nev,
the problem turned out to be not in ZoomPlayer itself but in the E-AC3 profile that I've created and spread around.
I've used the incorrect media subtype (the one stated above) instead of the real MEDIASUBTYPE_DOLBY_DDPLUS.
After fixing it everything is working OK.

Nonetheless, it's strange why it happens only with LAV Splitter. At the Inmatrix forum Gleb Egorych noticed that LAV Splitter initially sends MEDIASUBTYPE_FFMPEG_AUDIO ({AFBC2343-3DCB-4047-9655-E1E62A61B1C5}) and then switches to the correct one. Is this intentional?
If there are other affected ZoomPlayer users here is the discussion and the correct E-AC3 profile
http://forum.inmatrix.com/index.php?showtopic=13088

[nevcairiel]

LAV Splitter does not change media types on the fly, instead it will always export at least two media types on every Audio Pin, first all format specific types, and at the end of the list the MEDIASUBTYPE_FFMPEG_AUDIO. Connection Logic should always try the media types in order, starting with the first, and if that one fails advancing to the next.

MEDIASUBTYPE_FFMPEG_AUDIO is my own special media type which allows LAV Audio to playback all audio codecs that LAV Splitter can split without having to hook every codec up to its special separate media type first. Using this, i can basically ensure that every file that ffmpeg supports can be played - well, at least the Audio part of it.

[Mark_A_W]

Yep, I corrected the sub-type for E-AC3, and it's all working now :)

[pankov]

Nev,
I see the logic and it's perfect. And ZoomPlayer works according to it. It reached to this MEDIASUBTYPE_FFMPEG_AUDIO because it didn't find any profile using the correct MEDIASUBTYPE_DOLBY_DDPLUS. It did exactly what it was "told" to do by user. That is so typical of software - doesn't do what the user wants but only what it's told to do.
;)

[Gleb Egorych]

http://forum.inmatrix.com/index.php?showtopic=13088
Note it's a thread from a closed forum, not everyone has an access there.


BTW, I tested some M2TS files with E-AC3 tracks, there is something about LAV Source. I've found that LAV Source + ArcSoft Audio Decoder HD (ASAudioHD.ax) work badly, channel mapping is wrong and a part of channels are muted.

LAV Source + LAV Audio and LAV Source + ffdshow are OK. MPH-HC mpeg splitter is OK with LAV, ffdshow and ArcSoft.

[CruNcher]

LAV Splitter does not support adjusting the playback rate. 1x, or get out! :p

Huh and what for a reason is their for such a limitation, i mean Cyberlink does it also but many others don't, I mean MPC-HC seems also to force it but only when used in a Player environment it doesn't happen when direct264 for example requests it with this bitstream and 25 -> 30 fps (and lock) isn't 1x either ?

[nevcairiel]

The "rate" only matters for playback, as that defines if timestamps need to be adjusted. You can still pull frames as fast as you want. The only difference a different rate would make would be different timestamps, so in a playback scenario, stuff plays back faster.

[CruNcher]

Ok but why it locks then to 30 fps if you could pull the frames as fast as you want (in my example 42 fps the max i can get from the other tested splitter with those encoding settings http://forum.doom9.org/showthread.php?p=1502105#post1502105 ) it accelerates once to 35 fps and then goes down and stays @ 30 fps some strange behavior for lav splitter and cyberlinks demuxer compared to how the others handle it.

[nevcairiel]

A limitation/bug in that tool you're using.

[CruNcher]

A limitation/bug in that tool you're using.

so graphstudio is also limited buged ?

MPC-HC Splitter
http://imageshack.us/m/689/9136/mpchcsplitterframereque.png

Lav Splitter
http://img29.imageshack.us/img29/9689/lavsplitterframerequest.png

Something itchy is going on if you ask me ;)

[nevcairiel]

I don't have the MainConcept decoder, but on some H264 file, CoreAVC pumps out reliable 370-380fps for me.

[CruNcher]

As i said this is reproducible with any Decoder not only Mainconcepts (tried so far FFdshow,Dscaler,MPC-HC) and i can't say anything about H.264 bitstreams in .ts didn't checked if the same happens also in that scenario yet only Mpeg-2 .ts so far and only this specific Studio stream tested.
Im also going to try some older lav splitter builds maybe 1 of them works though if it is ffmpeg specific so a problem between how ffmpeg handles that and dshow i guess it wont be easy to fix. And as you can see the read process push into memory is unevenly distributed it looks hackish compared to MPC-HC continuous evenly distributed read. Also that its 30 (NTSC) fps or near that makes me wonder if it's hard coded behavior somewhere (maybe it was even more efficient todo it on *nix this way) ?

[Carpo]

using LAV Filters + LAV CUVID, playing Bluray discs on my laptop sounds better and plays better than PowerDVD :-)

[CruNcher]

hehe not surprising depending on your dshow player you use :) PowerDVD is bloated with features that can also be very destructive (how many screenshots i saw of users having their motion interpolation enabled and so strange results from it cant count that anymore or their dynamic color enhancement on) :P
Especially if you rate some source by that lower quality then it actually is in Real :(

[Carpo]

it came with the laptop, so i used it, now i dont have to, coupled with XBMC i have a nice HTPC now, also saves me having to move to Linux :D

[VipZ]

Ok but why it locks then to 30 fps if you could pull the frames as fast as you want (in my example 42 fps the max i can get from the other tested splitter with those encoding settings http://forum.doom9.org/showthread.php?p=1502105#post1502105 ) it accelerates once to 35 fps and then goes down and stays @ 30 fps some strange behavior for lav splitter and cyberlinks demuxer compared to how the others handle it.

I just did a quick test in MPC with LAV Audio+Splitter+ffdshow on MKV/AC3/h264. I had no issues getting 50fps at 2x playback while getting no dropped frames at under 10% CPU usage.




Nev, I am not sure what has changed but new builds are sweet, I can finally change from decoding to bitstreaming and back without MPC crashing :)

[ranpha]

With the latest git commit (https://github.com/Nevcairiel/LAVFSplitter/commit/f8698abe69f96ed13aaaf245bd358d482c296300), audio track switching from a FLAC track (decoded by LAV Audio) to a DTS-HD MA track (also decoded with LAV Audio with Arcsoft dll) will cause loud static sound to be produced. This happens only if I have AC3Filter between LAV Audio and the audio renderer.

This problem doesn't happen with the stable 0.26, where audio format switching happens just fine.

[kniff]

First time poster, long time reader!

Got nothing to contribute, i just registered to thank you Nev for the development and release of LAV filters. I recently setup my new fusionbased htpc and after changing a few settings my Onkyo greeted me with this

http://i.imgur.com/cHZO6.png

I will keep an eye on this thread for updates, thanks again (you should think about getting a paypal account up for donations like someone else asked about)

[nevcairiel]

With the latest git commit (https://github.com/Nevcairiel/LAVFSplitter/commit/f8698abe69f96ed13aaaf245bd358d482c296300), audio track switching from a FLAC track (decoded by LAV Audio) to a DTS-HD MA track (also decoded with LAV Audio with Arcsoft dll) will cause loud static sound to be produced. This happens only if I have AC3Filter between LAV Audio and the audio renderer.

This problem doesn't happen with the stable 0.26, where audio format switching happens just fine.

It should be fixed again. The problem is that AC3Filter does fail if you send it a special channel mask, which is totally weird.

In general, i strongly recommend against using AC3Filter for anything. Its outdated, unmaintained and severly broken.

[nevcairiel]

So, here is a test build with the new code that will map channels around.
The main use of that function is to make sure that when using WASAPI output over HDMI, the receiver will get a channel layout that it knows how to play back, because there is no metadata to indicate which channel goes to what speaker.
To solve this, all layouts will get expanded to 5.1/6.1/7.1 (except Mono and Stereo).

In addition, there are options to expand Mono to Stereo, and 6.1 to 7.1

http://files.1f0.de/lavf/LAVFilters-0.26-31-g6b06639.zip

I have one specific testing request.
Anyone that can natively play 6.1 content (i cannot, my receiver always only outputs stereo in that case), please run this test:

- Use ReClock for WASAPI playback, so the windows mixer is out of the game
- Play http://files.1f0.de/samples/331.dts
- Check if the Back-Center really comes out the Back channel(s) (even if you dont understand the guy, the order he says it is FL/FR/FC/SL/SR/BC)

Thanks!
PS:
There is also debug versions included which will create debug logs on your desktop. Either just register the debug versions, or rename them appropriately to activate.

[VipZ]

Nev I have a favor to ask if its not to much of a hassle :)

When the splitter get's reg'ed it deletes the forced extensions, what would your thoughts of a reg key which if set to 0, wont do this?

This function could make installers bit more flexible.

If the above is too much of a hassle, could you let me know where to edit the code for this?

Thanks

[Andy o]

In addition, there are options to expand Mono to Stereo, and 6.1 to 7.1

http://files.1f0.de/lavf/LAVFilters-0.26-31-g6b06639.zip

I have one specific testing request.
Anyone that can natively play 6.1 content (i cannot, my receiver always only outputs stereo in that case), please run this test:

- Use ReClock for WASAPI playback, so the windows mixer is out of the game
- Play http://files.1f0.de/samples/331.dts
- Check if the Back-Center really comes out the Back channel(s) (even if you dont understand the guy, the order he says it is FL/FR/FC/SL/SR/BC)

6.1 working OK now with the dca decoder. The 6.1->7.1 option is also working as intended (similarly to what the Arcsoft decoder does).

[ranpha]

It should be fixed again. The problem is that AC3Filter does fail if you send it a special channel mask, which is totally weird.

In general, i strongly recommend against using AC3Filter for anything. Its outdated, unmaintained and severly broken.

Just recompiled the build after the commit here (https://github.com/Nevcairiel/LAVFSplitter/commit/a17c8aa60eb42ee355ef0d529db509ee1affb373), and audio format switching now works with AC3Filter, at least in my tentative tests.

But I forgot to mention in my previous post that the same problem also happens if I have ffdshow Audio Decoder between LAV Audio and the audio renderer too. And even with the latest commit, the screeching noise will still happen if I were to switch from the FLAC track to the DTS-HD MA track.

[nevcairiel]

The difference is only channel masks. If one specific channel mask makes a filter go crazy, its a bug in that filter, not in my decoder. The logic right now is good the way it is, i cannot (and will not) change it because of buggy software like ffdshow or AC3Filter.
It has nothing to do with the outgoing data at all, its just one flag in the media type. I don't know how any filter can be so stupid to mess up because of that.

Also, i cannot work with the zero information you provide. There are dozens of different DTS and FLAC channel layouts around, how the hell am i to know which causes this problem. The layout is shown in the channel field on the status page of LAV Audio, it'll say "6 / 0x3f" for example, channel count and layout.
Also, i included those debug versions for a reason. :)

[ranpha]

The difference is only channel masks. If one specific channel mask makes a filter go crazy, its a bug in that filter, not in my decoder. The logic right now is good the way it is, i cannot (and will not) change it because of buggy software like ffdshow or AC3Filter.
It has nothing to do with the outgoing data at all, its just one flag in the media type. I don't know how any filter can be so stupid to mess up because of that.

Also, i cannot work with the zero information you provide. There are dozens of different DTS and FLAC channel layouts around, how the hell am i to know which causes this problem.
Also, i included those debug versions for a reason. :)

I believe it has nothing to do with channel configurations, but more on stream format.

Switching from a stereo FLAC track to a 5.1 DTS-HD MA track will cause static sound in ffdshow.
Switching from a 5.1 FLAC to a 5.1 DTS-HD MA track also cause the same thing.

Sample: http://www.filesonic.com/file/1000967784/
Log: http://dl.dropbox.com/u/132004/lavsplitterlog.7z

AC3Filter/ffdshow audio decoder are still important because of their mixer support, I can upmix stereo or downmix 7.1 tracks to my 5.1 speakers with them

[nevcairiel]

I cannot reproduce any problems with ffdshow. I was using LAV Audio for decoding, with ffdshow accepting uncompressed audio. Make sure you actually turn off decoding in ffdshow, at least DTS and FLAC in this case, or on track switch ffdshow might take over, at least in MPC-HC.
(or use ffdshow audio processor instead of decoder)

With AC3Filter, i encounterd a playback problem if DTS decoding was not turned off in its config, because AC3Filter cannot deal with DTS-HD, but it was just broken, no constant noise or anything.

Any particular settings you're using? Which output formats?
Which player?

[ranpha]

Player is MPC-HC build 3123 x86.
ffdshow is x86 build 3853.
Audio renderer is Reclock in WASAPI mode.
The Arcsoft dll is from TMT5.
LAVSplitter is still the one committed at this stage (https://github.com/Nevcairiel/LAVFSplitter/commit/a17c8aa60eb42ee355ef0d529db509ee1affb373), haven't pulled newer revisions yet.

Audio decoding has been turned off in ffdshow Audio Decoder (all decoders are disabled). Tried it with ffdshow Audio Processor too and the problem still persists. The same happened even if all post-processing filters are disabled.

BTW, you won't get the static sound immediately after you switch the audio track in the sample. When I first load the clip, MPC-HC will play the clip normally, with the default FLAC audio track being played. Then I switched the audio track to the DTS-HD MA track and the video continued playing, but without any sound. I then tried to seek the clip backwards (or forward) and the high-pitch static sound will start to appear.

Switching back to the FLAC track and the audio will return to normal.

More tests with AC3Filter seems to confirm that the problem has been solved there. Have to use it then as a mixer.

[nevcairiel]

BTW, you won't get the static sound immediately after you switch the audio track in the sample. When I first load the clip, MPC-HC will play the clip normally, with the default FLAC audio track being played. Then I switched the audio track to the DTS-HD MA track and the video continued playing, but without any sound. I then tried to seek the clip backwards (or forward) and the high-pitch static sound will start to appear.


I can reproduce it now, and this didn't happen with 0.26?
I'll run some tests.

First tests look like ffdshow is still thinking its getting 24-bit audio, but i'm actually sending 16-bit audio.

Edit:
Ah, there we go, i see what the problem is.
ffdshow is not capable to switch between 16int and 24int input on the fly. This sucks.
You can fix this by only selecting one output format in ffdshow.

If you want, you can bring this up with the ffdshow developers, maybe they can fix this, as it should be perfectly possible to switch on the fly.
Other decoders don't suffer from this as they typically do not switch to the best format on the fly, and rather are stuck with one format.

I recommend using either always 24-bit int or always 32-bit float output in the ffdshow processor, it internally uses float for matrixing anyway, i think.
Just disable all other output formats, keep 32-bit float active, and it should be fine.

Added to the FAQ too.

For the record, if this is just about up/down mixing, LAV will be able to do this at a later stage.

[clsid]

I know you are not a fan of workarounds. Adding an option to force 32float output in LAV when being connected to ffdshow is not going to happen, right?
Hopefully this can be fixed in ffdshow, since I prefer things to work directly out-of-the-box.

[nevcairiel]

Right now, i don't have the capabilities to force a specific output format.
Once i add format conversion, there will be an option for the user to force it, i may also consider forcing it when connecting to ffdshow.

The problem is, its not easy to detect that. I would need to know if its ffdshow connecting before i send it any media type .. we'll see.

[Virtual_ManPL]

Is bug with crack and noise when playing some explosions know maybe ?
It happens IIRC in some FLAC/DTS/AC3 (5.1/6.1/7.1). I don't remember details, but I can give it it will be needed ;p

[nevcairiel]

If there is any audio distortion, its rooted in ffmpeg itself, as i don't modify the audio.

Samples which clearly show the problem would be appreciated, though.

[Mercury_22]

Small request :) can you please, please ... separate the settings for x86 and x64 versions

[Gleb Egorych]

nevcairiel, am I right that for the latest test version the best dtsdecoderdll.dll version is 1.1.0.0 from 25.04.2008?

[nevcairiel]

Yes. However, 1.1.0.8 works nearly as good, just 6.0 content will not work properly, you'll not get the back channel. 6.1 works as expected, and there is not much 6.0 content .. so in practice, it wont matter much.

[yesgrey]

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.
I get the "dts" word even when the dtsdecoder.dll file is not in the LAV Audio folder. So, how could I know if the Arcsoft decoder is being used?

[nevcairiel]

I get the "dts" word even when the dtsdecoder.dll file is not in the LAV Audio folder. So, how could I know if the Arcsoft decoder is being used?

I just instruct it to load the dll, if its in any common dll path (windows system dir, etc), it'll load it from there. I didn't think it would be super useful to only load it from the LAV Audio dir, but if people think thats better, i can switch it to that.

Anyhow, if it says "dts", then its using it - from somewhere. :)

[SamuriHL]

I didn't think it would be super useful to only load it from the LAV Audio dir, but if people think thats better, i can switch it to that.


No....

[yesgrey]

if people think thats better, i can switch it to that.
No, keep it like it is now, it's the best way. Update your instructions instead, saying what you have just said. This way we only need to have it in one place. ;)

By the way, thanks for this new feature. You seem to be trying to dethrone madshi on giving us things that we didn't know that we would need them. :D

[bjd]

I get the "dts" word even when the dtsdecoder.dll file is not in the LAV Audio folder. So, how could I know if the Arcsoft decoder is being used?

Check the status page of the LAV Audio.

ffmpeg outputs 32bit float and Arcsoft 24 bit int - what do you see?

[nevcairiel]

giving us things that we didn't know that we would need them. :D

Those are the best things. :)

[yesgrey]

ffmpeg outputs 32bit float and Arcsoft 24 bit int - what do you see?
I see 24 bit int, but nevcariel already explained that the dtsdecoder.dll doesn't need to be in LAV Filter directory to be used... but thanks for the tip.

[nevcairiel]

It can also be 16 bit, when a lossless track is marked as such. ;)

[Dogway]

I couldn't bitstream, ffdshow audio gets used instead. I dont know what might be but sorry for my ignorance, to bitstream do I need a receiver? or can I just plug it to the TV and let the TV do the downmix? It could be the TV or just the HDMI cable, a very cheap one... Not that Im stressed about this, I was just testing out of curiosity.

[Sebastiii]

Hi,

To follow discussion on IRC, i just see that with .m2ts or .mpls and when ffdshow video decoder is in graph, when changing subtitle stream (memory grow up) if i do multiple sub change, my app crash (MP or Graphstudio).
This is log :
http://dl.dropbox.com/u/10536084/lav/LAVSplitter-log1.rar
http://dl.dropbox.com/u/10536084/lav/LAVSplitter-log2.rar

I have tested with official and own build (LAVF and FFDShow).
It didn't happen with other splitter (i have try with old xbmcsplitter ver7, MPC MPEG Gabest).

On mkv, it seems ok or memory didn't jump higher with m2ts/mpls.

I hope that someone can try and see the same thing, i have tested on 2 PC(s).
It did happen when ffdshow is decoder or post-process filter.
Cheers,
Seb.

[Mark_A_W]

I just instruct it to load the dll, if its in any common dll path (windows system dir, etc), it'll load it from there. I didn't think it would be super useful to only load it from the LAV Audio dir, but if people think thats better, i can switch it to that.

Anyhow, if it says "dts", then its using it - from somewhere. :)

Umm...ahhh...I'll have to beg to differ from the others.

I'd really prefer it to ONLY load from the LAVAudio directory. Then you are in control.

Or, at the very least report the version.

Otherwise you are going to get people complaining that this version does this, or that version does something stupid....when all along they are running some other version stuck somewhere on their system.

I for one have a few versions floating around, and bonus paths set in environment from jumping through hoops trying to get the arcsoft decoders to work for directshow for eac3to in the early days. (And then madshi lets the cat out of the bag that he didn't really need the directshow registration...).

Keep up the good work!

Mark

[VipZ]

Hi Nev, I have found that LAV Splitter cant deal with MP3's with embedded cover art, here is a quick sample for this http://www.mediafire.com/?xq9nw1kiiaz04iz

I have also got some MPEG-PS files that don't load LAV Splitter, will PM sample.

Thanks

[nevcairiel]

Fixed, both samples work fine with my very latest change.

[VipZ]

Fixed, both samples work fine with my very latest change.

Thanks :)

[ranpha]

Umm...ahhh...I'll have to beg to differ from the others.

I'd really prefer it to ONLY load from the LAVAudio directory. Then you are in control.

Or, at the very least report the version.

Otherwise you are going to get people complaining that this version does this, or that version does something stupid....when all along they are running some other version stuck somewhere on their system.

I for one have a few versions floating around, and bonus paths set in environment from jumping through hoops trying to get the arcsoft decoders to work for directshow for eac3to in the early days. (And then madshi lets the cat out of the bag that he didn't really need the directshow registration...).

Keep up the good work!

Mark

I also second this proposal. LAV Splitter should only load it from the the same directory as the .ax files. I have some of those dll files splattered in Program Files.

At least make it an option?

[jmone]

nevcairiel, I've been testing subtitle handling for MC16 latest beta changes with V0.26 of LAVSplitter (you opinions in that thread would be welcome) using Avatar to test Forced Subs. I notice that at the 47:00 minute mark is the forced sub "Why me? That's not fair...", and this one hangs around for a very very long time (eg till the next forced sub at around 53:45). I know you touched on the logic for this previously...

[nevcairiel]

Thats a known problem, and there is still alot of room for improvement in the PGS filtering.

[jmone]

How about just timing them out after say 5 seconds or so?

[nevcairiel]

I rather work on the proper solution once time permits.

[jmone]

Sounds good

[Mercury_22]

@Nev
I thought that LAVSplitter it's able to play ts /m2ts files with 2 video streams (just the first video stream ) am I wrong ? cause it can't play e.g. this one (http://www.multiupload.com/MV7RNMR1LW) On the full m2ts file it can't even start to play (MPC-HC it's "freezing") but on the sample it's playing just a black screen while MPC-HC's internal splitter it's working OK

P.S. Before you ask yes it's straight from Blu-ray :)

[nevcairiel]

It probably breaks because the second stream seems .. broken. Is that a 3D Blu-ray? Either that, or its some funny PIP stream.
I'll take a look tomorrow why it doesn't play the working stream.

[jmone]

Quick Q - Where/How are external sub files (eg idx/sub) displayed (eg ignored by LAVSplitter as they are not part of the "file" but then decoded by FFDSHOW?)

[nevcairiel]

LAV Splitter does not care about external files at all, the player needs to process those (or if ffdshow does it itself, then that).
In MPC-HC, its actually the player itself that loads external files, like audio and sub streams.

[Mercury_22]

It probably breaks because the second stream seems .. broken. Is that a 3D Blu-ray? Either that, or its some funny PIP stream.
I'll take a look tomorrow why it doesn't play the working stream.
No it's not a 3D Blu-ray and again MPC-HC's internal splitter it's playing it fine:confused:

http://bluray.highdefdigest.com/3563/crimsonwing.html

[assassinhtpc]

Any issues with live tv files?

Can anyone help me with this question?

[Kaotech]

Hello, with a 5.1 DTS MA Track, Lavaudio set the track at 7.1 with 0.26.

[nevcairiel]

MPC-HC's internal splitter it's playing it fine:confused:

I wish people would stop with these apple and orange comparisons. How is the MPC-HC splitter related to LAV Splitter at all?
Let me answer that for you: It is not, not in *any* way. It does not share *any* code. LAV Splitter is unique, as its the *only* DirectShow splitter that uses ffmpeg for demuxing.

PS:
Your file is fixed.

[assassinhtpc]

So has anyone used LAV with live tv?

[nevcairiel]

Suggestion: Instead of asking if something works, just go out and try it.

[assassinhtpc]

Suggestion: Instead of asking if something works, just go out and try it.

Well I thought since I did a search in this massive thread and couldn't find any mention of it that it might not be possible.

And I haven't tried it but a friend has and it didn't work.

Therefore I just asked.

From a friend:
I am not getting it to work with live TV. I am just starting to delve into that mystery and the doom9 thread is huge. I'll get back to you on that ome..

[Andy o]

At least take the time to write all relevant information if you want people to take the time to help you out. What "live TV" files do you mean? What format? What player? What OS? etc etc.

[assassinhtpc]

At least take the time to write all relevant information if you want people to take the time to help you out. What "live TV" files do you mean? What format? What player? What OS? etc etc.

No problem. Thanks for offering to help.

It would be a live digital tv stream from an OTA antenna, not a stored file on a HDD, that is being rendered so it is the Microsoft decoders built into Win7x86 using WMC for Live TV playback that are being messed up by LAV.

Also, I wanted to do some searching but what the issue is that in live tv it will get into a 2-second or so audio loop that lasts about maybe a minute before it posts up the file is missing. This is the exact same issue I incountered with Shark007 until I switched to Win7x86 so I could use the alternate Video decoder which fixed the Audio problem.

[assassinhtpc]

Suggestion: Instead of asking if something works, just go out and try it.

Didn't mean to offend you. I have spent the last few days trying to search for a discussion on this to help him out and have come up empty.

Just hoping you could help.

[noee]

Some of the guys on the jRiver forums, IIRC, are doing jRiver TV with LAVSplitter, or they're giving it a go, perhaps they'll chime in.

[assassinhtpc]

Some of the guys on the jRiver forums, IIRC, are doing jRiver TV with LAVSplitter, or they're giving it a go, perhaps they'll chime in.

Is there anything free that is a simple fix?

[Andy o]

No problem. Thanks for offering to help.

It would be a live digital tv stream from an OTA antenna, not a stored file on a HDD, that is being rendered so it is the Microsoft decoders built into Win7x86 using WMC for Live TV playback that are being messed up by LAV.

Also, I wanted to do some searching but what the issue is that in live tv it will get into a 2-second or so audio loop that lasts about maybe a minute before it posts up the file is missing. This is the exact same issue I incountered with Shark007 until I switched to Win7x86 so I could use the alternate Video decoder which fixed the Audio problem.

Are you by any chance using HDMI with an ATI card? If so, turn off the media center menu sounds, and see if it helps.

[andyvt]

It would be a live digital tv stream from an OTA antenna, not a stored file on a HDD, that is being rendered so it is the Microsoft decoders built into Win7x86 using WMC for Live TV playback that are being messed up by LAV.


WMC uses the SBE to record content. Live TV works a little different from Recorded TV in that it doesn't use a circular buffer, but both hit the disk.

None of the LAV filters should get used for TV playback in WMC, if they are it's probably because the filter has a very high merit value. Changing that to a more reasonable # might help.

[assassinhtpc]

WMC uses the SBE to record content. Live TV works a little different from Recorded TV in that it doesn't use a circular buffer, but both hit the disk.

None of the LAV filters should get used for TV playback in WMC, if they are it's probably because the filter has a very high merit value. Changing that to a more reasonable # might help.

Thanks so much. How do I change their merit value?

[andyvt]

Thanks so much. How do I change their merit value?

I usually go with GraphStudio. If UAC is enabled (or you're running as a user), you will need to right click and run as administrator.

I did a brief write up (http://babgvant.com/blogs/andyvt/archive/2008/11/30/understanding-merit-to-optimize-playback.aspx) on how to use GraphStudio a while back; don't think anything has changed so much that it's no longer valid.

[assassinhtpc]

I usually go with GraphStudio. If UAC is enabled (or you're running as a user), you will need to right click and run as administrator.

I did a brief write up (http://babgvant.com/blogs/andyvt/archive/2008/11/30/understanding-merit-to-optimize-playback.aspx) on how to use GraphStudio a while back; don't think anything has changed so much that it's no longer valid.

Absolutely perfect. I can't thank you enough.

[assassinhtpc]

Are you by any chance using HDMI with an ATI card? If so, turn off the media center menu sounds, and see if it helps.

No it is the integrated intel GPU on an i3 550 (Clarkdale) with no discrete graphics card.

[assassinhtpc]

I usually go with GraphStudio. If UAC is enabled (or you're running as a user), you will need to right click and run as administrator.

I did a brief write up (http://babgvant.com/blogs/andyvt/archive/2008/11/30/understanding-merit-to-optimize-playback.aspx) on how to use GraphStudio a while back; don't think anything has changed so much that it's no longer valid.

Holy hexidecimal!

Do you have any suggested initial settings that I should try?

[jmone]

I have no issues with "Live TV", "Timeshifting Live TV", or "Recorded Live TV" using LAVAudio and MC16. In fact thanks to feedback from nevcairiel a bug was fixed in the MC16 TV Splitter (JTV) so you can now switch between DVB-T channels with different format Audio without issue. One reason it plays "nice" is that MC16 uses its own splitter to present just the single Video and Audio streams instead of just dumping the whole collection of DVB-T Sub Channels in one file.

[assassinhtpc]

I have no issues with "Live TV", "Timeshifting Live TV", or "Recorded Live TV" using LAVAudio and MC16. In fact thanks to feedback from nevcairiel a bug was fixed in the MC16 TV Splitter (JTV) so you can now switch between DVB-T channels with different format Audio without issue. One reason it plays "nice" is that MC16 uses its own splitter to present just the single Video and Audio streams instead of just dumping the whole collection of DVB-T Sub Channels in one file.

Understood.

I think it would be nice if there was a free option to make this work too.

[Andy o]

No it is the integrated intel GPU on an i3 550 (Clarkdale) with no discrete graphics card.

I don't know much about Intel, but you might still wanna try turning off MC sounds see if that makes a difference.

[wo0zy]

Understood.

I think it would be nice if there was a free option to make this work too.

Works fine with Mediaportal.

Wo0zy

[nevcairiel]

TV applications typically have their own source filter and mpegts demuxer, therefor LAV Splitter doesnt really fit into that picture to begin with.

LAV Audio might work with it, depending on the source filter. If the source unpacks the PES data, it will work, if the source directly sends the PES stream without unpacking, it will not.
FWIW, DVBViewer works fine with LAV Audio and LAV CUVID.

[jmone]

Wo0zy

Now there is a NIC that broought back a flood of early HTPC memories trying to get those Intel G33 working at 24hz (shudder). I expect to see archibael show up next!

[Gleb Egorych]

FWIW, DVBViewer works fine with LAV Audio and LAV CUVID.
My results: LAV Audio - yes, LAV CUVID - no. On channel change - no picture.

[Aleksoid1978]

LAV very bad with this sample http://www.multiupload.com/MV7RNMR1LW:
1 - playback don't start, black video in MPC-HC
2 - second video stream detect with error.
3 - detect many "fake" subtitle.

Play fine with MPC-HC internal splitter.

[pankov]

My results: LAV Audio - yes, LAV CUVID - no. On channel change - no picture.
Gleb,
if you wait a few seconds (10-15) the video will come.
Or you can disable "Fast Channel Switching" in Options -> TV + Radio and it will work suddenly but you'll get a black screen in between.
I thought it was only me that had problems with this fast channel switching but I guess it was actually LAV CUVID that is taking too long to adapt/switch ... or potentially clear it's buffers.
Nev,
do you use DVBViewer? Is there anything that can be done to improve it?

[Mercury_22]

LAV very bad with this sample http://www.multiupload.com/MV7RNMR1LW:
1 - playback don't start, black video in MPC-HC
2 - second video stream detect with error.
3 - detect many "fake" subtitle.

Play fine with MPC-HC internal splitter.

:) Already reported
@Nev
I thought that LAVSplitter it's able to play ts /m2ts files with 2 video streams (just the first video stream ) am I wrong ? cause it can't play e.g. this one (http://www.multiupload.com/MV7RNMR1LW) On the full m2ts file it can't even start to play (MPC-HC it's "freezing") but on the sample it's playing just a black screen while MPC-HC's internal splitter it's working OK

P.S. Before you ask yes it's straight from Blu-ray :)
and "fixed"
I wish people would stop with these apple and orange comparisons. How is the MPC-HC splitter related to LAV Splitter at all?
Let me answer that for you: It is not, not in *any* way. It does not share *any* code. LAV Splitter is unique, as its the *only* DirectShow splitter that uses ffmpeg for demuxing.

PS:
Your file is fixed.
@Nev :thanks: and I was just trying to say that the file works (with other splitters)
@Aleksoid Can you fix MPC-HC's MPEG splitter to detect the DTS Express audio too ?

[JaylumX]

Nevcaireiel, is it possible as a feature to implement in the Lavfilters splitters a method of switching video streams for Bluray - for example my King Kong, Mirrors and Gladiator bluray have more one version - Theatrical, Directors Cut and/or Extended cuts on one BD. When using MPC-HC to play the disk it always defaults to playing the longer version. Can you implement that feature of stream switching or is that something i should be asking the MPC-HC coder(s)

[Gleb Egorych]

Gleb,
if you wait a few seconds (10-15) the video will come.
Or you can disable "Fast Channel Switching" in Options -> TV + Radio and it will work suddenly but you'll get a black screen in between.
You're right, disabling the option does the job, thanks. However, deinterlacing problems with DVB streams still persist. So usage of LAV CUVID in DVBViewer is limited at the moment.

[jj666]

JaylumX,

I believe the best way to achieve what you want is to find the correct playlist file for the different versions using BDINFO. Then run the appropriate playlist file.

Cheers,

-jj-

[SamuriHL]

Nevcaireiel, is it possible as a feature to implement in the Lavfilters splitters a method of switching video streams for Bluray - for example my King Kong, Mirrors and Gladiator bluray have more one version - Theatrical, Directors Cut and/or Extended cuts on one BD. When using MPC-HC to play the disk it always defaults to playing the longer version. Can you implement that feature of stream switching or is that something i should be asking the MPC-HC coder(s)

That feature is on Nev's list to implement. No ETA.

[assassinhtpc]

TV applications typically have their own source filter and mpegts demuxer, therefor LAV Splitter doesnt really fit into that picture to begin with.

LAV Audio might work with it, depending on the source filter. If the source unpacks the PES data, it will work, if the source directly sends the PES stream without unpacking, it will not.
FWIW, DVBViewer works fine with LAV Audio and LAV CUVID.

The problem is that if someone uses LAV they might not be able to use WMC Live TV.

Do you think that changing the merits will correct this problem?

[Andy o]

Nev, I you have planned output bit depth selection for the future, right? If so, could you please consider an option akin to ffdshow's "16-bit LPCM" (not "16-bit integer")? I've just found out that PowerDVD 8's audio decoder can connect to ffdshow's 16-bit LPCM output in order to apply Dolby Headphone, which as far as I can tell is the only way to get DH in a DirectShow player (foobar2000 does it (http://www.hydrogenaudio.org/forums/index.php?showtopic=39404) by way of the Cyberlink DH dll).

If it's not too much to ask :)

[nevcairiel]

Nev, I you have planned output bit depth selection for the future, right? If so, could you please consider an option akin to ffdshow's "16-bit LPCM" (not "16-bit integer")? I've just found out that PowerDVD 8's audio decoder can connect to ffdshow's 16-bit LPCM output in order to apply Dolby Headphone, which as far as I can tell is the only way to get DH in a DirectShow player (foobar2000 does it (http://www.hydrogenaudio.org/forums/index.php?showtopic=39404) by way of the Cyberlink DH dll).

If it's not too much to ask :)

I don't think i'll be doing that.

[nevcairiel]

LAV very bad with this sample http://www.multiupload.com/MV7RNMR1LW:
1 - playback don't start, black video in MPC-HC
2 - second video stream detect with error.
3 - detect many "fake" subtitle.

Play fine with MPC-HC internal splitter.

I don't think the subtitles are fake, it detects them properly, and the subtitle PIDs are set properly. Looks just fine.
BTW, the MPC-HC splitter doesn't detect the second video stream at all on the beginning, it just finds it after a few seconds of playback. I think its either broken, or the stream starts really late in the file, i am filtering video streams without a resolution now.
Anyhow, i also increased the size of the probe buffer, which makes it detect the stream.

[nevcairiel]

The problem is that if someone uses LAV they might not be able to use WMC Live TV.


I do not care about any interaction with a seriously locked down application like WMC. Its a horrible player as it is.

[madshi]

3 - detect many "fake" subtitle.
FWIW, eac3to detects a bunch of PGS subtitles, too, so I'd say that LAV Splitter probably is correct about the subtitles.

[Andy o]

I don't think i'll be doing that.

OK, thanks.

[assassinhtpc]

I do not care about any interaction with a seriously locked down application like WMC. Its a horrible player as it is.

Wow. Okay then.

[andyvt]

I do not care about any interaction with a seriously locked down application like WMC. Its a horrible player as it is.

What about making it easier for users to control the filter's merit?

The default merit is so high that it will most likely be considered in every graph that isn't explicitly built with direct connections.

[BloodySword]

The splitter does not accept to connect to ffdshow Audio Decoder when splitting MPEG2 with LPCM or PCM in WAV-Files. Is this by design?

[nevcairiel]

What about making it easier for users to control the filter's merit?

The filters merit can only be controlled with UAC elevation, so its not possible to change it from within the properties page, unless the player is launched with elevation, which isn't really something people will want to do.

I'm sorry, but i don't write software for clueless users, thats what those codec packs are for, which do the merit adjustment for them.
I'll add that low-merit compile mode at some point, if that helps codec packs makers or something.

[wo0zy]

TV applications typically have their own source filter and mpegts demuxer, therefor LAV Splitter doesnt really fit into that picture to begin with.

LAV Audio might work with it, depending on the source filter. If the source unpacks the PES data, it will work, if the source directly sends the PES stream without unpacking, it will not.
FWIW, DVBViewer works fine with LAV Audio and LAV CUVID.

I'd taken that as a given and was preferring to LAV Audio. How the h**l Assassin got LAV anything into MC7's live TV chain without seriously messing it up is the mystery to me (but I've honestly never tried).:p

LAV audio in MP is working well for me with one exception but I won't bore you with that until you've finished more important features ;)

All the best,

Wo0zy

[wo0zy]

Now there is a NIC that broought back a flood of early HTPC memories trying to get those Intel G33 working at 24hz (shudder). I expect to see archibael show up next!

Hehe. I'm very much still around mate and still working with Intel stuff during the day (by night I'm an ATI guy ATM but don't tell anyone :p). I think Archibeal is watching but not posting so much these days.

If only JRiver would extend DVR support to DVB-S (and improve DVR support in general) then you'd have come across me a lot more. I love it other than that and appreciate the work you've done over there.

Good to hear from you mate.

All the best,

Wo0zy

PS Sorry for OT post Nev. Won't happen again :)

[assassinhtpc]

I'd taken that as a given and was preferring to LAV Audio. How the h**l Assassin got LAV anything into MC7's live TV chain without seriously messing it up is the mystery to me (but I've honestly never tried).:p

LAV audio in MP is working well for me with one exception but I won't bore you with that until you've finished more important features ;)

All the best,

Wo0zy

Like I said I am trying to help a friend.

Since I am a "clueless user" I will wait on my TV tuner card to arrive and try this out first hand.

I think its pretty limited thinking to restrict yourself to "non-clueless users" btw.

I am trying to be as nice as possible here. Don't appreciate being insulted when I am trying out something new (for me anyway) with your splitter. And I am a LAV supporter.

[BloodySword]

Sorry for doubleposting, but:

The splitter does not accept to connect to ffdshow Audio Decoder when splitting MPEG2 with LPCM or PCM in WAV-Files. Is this by design?

[nevcairiel]

I think its pretty limited thinking to restrict yourself to "non-clueless users" btw.

Its not like i get paid or anything, so shrug.
I do this for fun, and dealing with complaints why WMC broke is not something i find entertaining. If it works for you, great, if it doesn't - well, "i told you so" comes to mind.

All i ask of people is that they know how to install a DirectShow filter manually, and know what that means for them. If they don't, then they better stick with an automated codec installer.

PS:
If "clueless user" is an insult to you, which i didn't even address at you specifically, then i'm sorry. (i would however wonder how you survived on the interwebs until now)

[assassinhtpc]

Its not like i get paid or anything, so shrug.
I do this for fun, and dealing with complaints why WMC broke is not something i find entertaining. If it works for you, great, if it doesn't - well, "i told you so" comes to mind.

All i ask of people is that they know how to install a DirectShow filter manually, and know what that means for them. If they don't, then they better stick with an automated codec installer.

PS:
If "clueless user" is an insult to you, which i didn't even address at you specifically, then i'm sorry. (i would however wonder how you survived on the interwebs until now)

I never complained. I asked how to get it working. I appreciate your work on all of this and even started the LAV thread over at AVS.

I thought my initial question was very cordial and appropriate. Sorry if I somehow angered you or brought up a sore subject.

I use WMC and it works for what I use it for. Evidently it doesn't work for how many in this thread use it.

[nevcairiel]

Your question was way too general, don't expect to get those answered here.
I will nearly always ignore questions where i had to ask three follow up questions just to get in the overall area of what you wanted.

Lets take this as an example:
Any issues with live tv files?

First off, live tv is files now? :)
Secondly, what live tv? Yes, i can use it fine with DVBViewer. It also works fine with MediaPortals TV thing.

You basically supplied no information whatsoever, and you bumped your own question just a few posts later.
Thats something i'll just ignore - thats not proper forum conduct, and i'll not reward it with a response (especially because i cannot answer your question without asking 3 other questions first).

Maybe this can help you formulate better questions in the future.

PS:
AFAIK, Live TV in WMC only works with the Microsoft components, because WMC is completely locked down.

[assassinhtpc]

Your question was way too general, don't expect to get those answered here.
I will nearly always ignore questions where i had to ask three follow up questions just to get in the overall area of what you wanted.

Lets take this as an example:


First off, live tv is files now? :)
Secondly, what live tv? Yes, i can use it fine with DVBViewer. It also works fine with MediaPortals TV thing.

You basically supplied no information whatsoever, and you bumped your own question just a few posts later.
Thats something i'll just ignore - thats not proper forum conduct, and i'll not reward it with a response (especially because i cannot answer your question without asking 3 other questions first).

Maybe this can help you formulate better questions in the future.

Points well taken. I will get specific information and report back. Thanks for the help and guidance.

[Mercury_22]

I don't think the subtitles are fake, it detects them properly, and the subtitle PIDs are set properly. Looks just fine.
BTW, the MPC-HC splitter doesn't detect the second video stream at all on the beginning, it just finds it after a few seconds of playback. I think its either broken, or the stream starts really late in the file, i am filtering video streams without a resolution now.
Anyhow, i also increased the size of the probe buffer, which makes it detect the stream.

:thanks: for the second video stream (incidentally yes it's a PIP stream)

Now only if you could make the audio for this video stream work ... it's the DTS Express [secondary] stream
When playing the full m2ts file or the index.bdmv and I'm selecting this secondary audio (the DTS Express stream) MPC-HC stops (freeze) playing and it's continuing to play only when I select another audio (I know that on the sample it's continuing to play without sound)


P.S. I don't now if you saw my post before so (if it's of any use to you) again the disc is "The Crimson Wing: Mystery of the Flamingos" (http://bluray.highdefdigest.com/3563/crimsonwing.html)(US)

Edit if you need a bigger sample let me know

[nevcairiel]

Now only if you could make the audio for this video stream work ... it's the DTS Express [secondary] stream

DTS Express is a pure HD stream, only the ArcSoft decoder would be able to play it anyway, it does not have a "core" to play with other decoders.

Its planned to support it at some point in LAV Audio (through the ArcSoft decoder), but for now its not supported. Those are only used for PIP audio, anyhow.

[Mercury_22]

DTS Express is a pure HD stream, only the ArcSoft decoder would be able to play it anyway, it does not have a "core" to play with other decoders.

Its planned to support it at some point in LAV Audio (through the ArcSoft decoder), but for now its not supported. Those are only used for PIP audio, anyhow.
:eek: OK :thanks: I thought it's already supported (through the ArcSoft decoder)

I'll wait :p

[bjd]

Just wanted to say thanks Nev, for the work you have done on this project. V0.26 with arcsoft dts integration is amazing and has simplified my HTPC set up. I have been testing for a week now and not come up against any issues. So impressed with what you have managed to accomplish.

[whurlston]

Points well taken. I will get specific information and report back. Thanks for the help and guidance.

I use the audio decoder for live/timeshifted TV. I sometimes get out of sync audio on Dolby 2.0 channels but I haven't looked into whether it's LAV or not that's causing it.

I don't see why the splitter couldn't be used but the application would have to be written to support it. My prefered software (NextPVR) always uses the MS demux.

[andyvt]

The filters merit can only be controlled with UAC elevation, so its not possible to change it from within the properties page, unless the player is launched with elevation, which isn't really something people will want to do.


+1 for specialization :)

http://babgvant.com/downloads/lavfiltersettings/filtersettings.png

x86 binary (http://babgvant.com/downloads/lavfiltersettings/lavfiltersettings.zip) (a reboot may be required for merit changes to take effect).

[andyvt]

I'll add that low-merit compile mode at some point, if that helps codec packs makers or something.

Low merits (like the one in the patch) aren't useful for codec packs or IC graph building because they will almost never join the graph automatically. Instead of escalating the merit "arms race" I prefer to tightly control the merit of filters on my system so I can influence graph building in a much more tactical way.

[assassinhtpc]

Another question. I'll be as specific as possible here.

I am running Win7 64 bit with an ATI 5450 card bitstreaming to a Denon AVR.

I tried to play an AVI file tonight for the wife and at first there wasn't any sound. Then it wouldn't play at all.

So I completely uninstalled LAV and re-installed it an intentionally left AVI UNchecked. Then I went into WMC (yes, I use WMC) and it played.

So I thought I had it fixed.

Then I used graph studio and selected all my bitstreaming optoins (DD and DTS only as my receiver doesn't decode HD Audio).

I then went back to WMC and the file wouldn't play at all.

Any idea how to fix this?

[skingery]

Another question. I'll be as specific as possible here.

I am running Win7 64 bit with an ATI 5450 card bitstreaming to a Denon AVR.

I tried to play an AVI file tonight for the wife and at first there wasn't any sound. Then it wouldn't play at all.

So I completely uninstalled LAV and re-installed it an intentionally left AVI UNchecked. Then I went into WMC (yes, I use WMC) and it played.

So I thought I had it fixed.

Then I used graph studio and selected all my bitstreaming optoins (DD and DTS only as my receiver doesn't decode HD Audio).

I then went back to WMC and the file wouldn't play at all.

Any idea how to fix this?
There is a lot going on here.

First, install Mediainfo (Google it). This will let you right-click on a file and see what it contains as far as video and audio. Just saying AVI doesn't give a good clue to what the file contains and then what would play it back.

2nd, my opinion so take it as such, don't mess around with live and recorded TV in Windows Media Center. Just let the native codecs do what they do. I like tweaking stuff as much as anyone here but I've learned to be minimalist when it comes to codecs. I let MS handle TV.

For movies that have been ripped, I use LAV splitter, LAV audio and LAV CUVID for video (since my GPU is Nvidia). I play my movies with MPC-HC so I can tell it to use LAV with out having to muck about with Merit.

A couple other resources for you:
missingremote.com
hack7mc.com

[jazzysmooth]

Assassin, does that file have DD or DTS audio in it? If you uncheck bitstreaming for those items does it play again?

If you run WMC in a window, play the file, then connect to remote graph with GraphStudio, does it even show LAV in the chain?

[VipZ]

+1 for specialization :)

x86 binary (http://babgvant.com/downloads/lavfiltersettings/lavfiltersettings.zip) (a reboot may be required for merit changes to take effect).

That's a cool little app you got there, I was thinking about trying to do something that like, but i am clueless about programming :)

Any chance this could be updated to support x64 as well?

[assassinhtpc]

Assassin, does that file have DD or DTS audio in it? If you uncheck bitstreaming for those items does it play again?

If you run WMC in a window, play the file, then connect to remote graph with GraphStudio, does it even show LAV in the chain?

Thank you so much for trying to help.

It has Dolby 2.0. Its actually some crappy exercise video of my wife's.

I tried to uncheck bitstreaming and that didn't work.

It does show LAV in the chain and will even play prefect inside the Graph window. Just when I try to play it in WMC do I have issues.

Again, I know that many of you don't use WMC and I recognize that it isn't the greatest player to use but its pretty popular in the US HTPC market. So I am trying to learn how to make everything compatible because I think LAV is absolutely phenomenal.

I ended up using handbrake and ripping it to MKV for her which of course played flawlessly. There is just something simple that probably needs to be changed to get LAV/WMC/AVI to work together.

[andyvt]

Any chance this could be updated to support x64 as well?

Yes. That would be very easy.

[VipZ]

Yes. That would be very easy.

Cool, that would be great :)

[nevcairiel]

LAV Filters 0.27

General
- Added debug versions which generate log files on your desktop

LAV Splitter
- Fixed a memory leak when switching streams
- Increased probe buffer size (which results in files to be properly detected in more cases)
- Fixed playback of MPEG-TS files with a odd program configuration

LAV Audio
- Improvements to the DTS decoder interface
- Support for 1.1.0.0
- Try to reverse any channel upmixing done by the decoder, outputting the audio as close to the original source as possible.
- New Channel Mapping functionality
- Output only "standard" channel layouts (Mono, Stereo, 5.1, 6.1, 7.1), filling non-existant channels with silence. Required by some HDMI receivers with "odd" source layouts. (default)
- Convert Mono to Stereo (simply double the audio channel)
- Convert 6.1 to 7.1 (Back Center gets doubled into both back channels)


Download: Installer (both x86/x64) (http://files.1f0.de/lavf/LAVFilters-0.27.exe) -- Zips: 32-bit (http://files.1f0.de/lavf/LAVFilters-0.27.zip) & 64-bit (http://files.1f0.de/lavf/LAVFilters-0.27-x64.zip)

So, quick notes about the features:

My AV Receiver isn't particularly happy if it receives multi-channel PCM which is not in a "default" configuration. For example, i was playing 5.0 content some time back, and my center vanished. I *think* it played it as 4.1, sending the center to the LFE.. So anyhow, since you cannot send the channel layout metadata over HDMI, and the receiver only has the channel count to work with, i decided to implement a feature to actually map the output onto those default layouts.

I'm not 100% if its perfect with all codecs and all files, please report if you found any issues.
To help with this, the ZIP files now also contain debug versions of the two filters, which will write a debug log to your Desktop - or so i hope. The log is cleared on every launch, so it won't grow into the extreme.

As always, have fun and take care.

[SamuriHL]

Thanks, Nev! Looks like fun. :)

[Sebastiii]

Thanks :) and thanks again :)

[madshi]

Wonderful, I appreciate the native audio output - thanks!

(Now you got that version out, maybe you can look at the uncompressed YCbCr videos? :))

[nevcairiel]

Now you got that version out, maybe you can look at the uncompressed YCbCr videos? :)

I'll work on that soon'ish, work is keeping me busy ATM, at the latest on the weekend.

[wo0zy]

Thanks Nev.
The mono to stereo update should fix one of my little issues :)

All the best,

Wo0zy

[madshi]

I'll work on that soon'ish, work is keeping me busy ATM, at the latest on the weekend.
Sounds good to me, thanks.

[Gleb Egorych]

Thanks, nev! :)

[assassinhtpc]

Thanks Nev!

[VipZ]

Thanks, new shiny version :)

[Boltron]

Thanks, can't wait to test it out.

[BloodySword]

Hey, Nev, grat work!

I have a small suggestion. The stream switcher is not reachable for me because I use my own player.
Could you please make an optional tray icon for the stream switcher? This would be very nice!

[nevcairiel]

Maybe you should teach your player. :)

Some day, there might be a tray icon, but not soon.
If you use ffdshow, or madVR - they produce tray icons to switch as well.

[BloodySword]

That's the problem. Your splitter reports only one audio pin. How can I configure it that all streams are shown to ffdshow audio decoder?
And I have the problem that I must pipe LPCM through your audio decoder filter because ffdshow does not accept the LPCM stream from your spitter (but from every other splitter).
And how should I teach my player? It is written in VB6 but very stable. I did not do any work since many years.

[kennae]

Great work on the few latest versions. Everything is working perfectly on my end.
One small question, do I still need the "File Source (Async.)" in my MPC-HC's external filters list?

[nevcairiel]

One small question, do I still need the "File Source (Async.)" in my MPC-HC's external filters list?

Nothing changed in that regard (and probably won't change on my end, unless MPC-HC changes or you use another way to force a specific filter).
You can always just try not having it, in many cases it'll "just work"

And I have the problem that I must pipe LPCM through your audio decoder filter because ffdshow does not accept the LPCM stream from your spitter (but from every other splitter).

I just tried a VOB LPCM file, and it works just fine.
Blu-ray LPCM also work perfectly. Other formats do not have "LPCM", MKV for example just has PCM, which does not need a decoder.

[andyvt]

+1 for specialization :)

http://babgvant.com/downloads/lavfiltersettings/filtersettings.png

x86 binary (http://babgvant.com/downloads/lavfiltersettings/lavfiltersettings.zip) (a reboot may be required for merit changes to take effect).

x86 and x64 binaries are now available on SourceForge (https://sourceforge.net/projects/babgvant/files/LAVFilterSettings/), source (http://babgvant.svn.sourceforge.net/viewvc/babgvant/LAVFilterSetttings/) is too (not that it's very interesting :)).

[BelowSky]

nevcairiel, did you accidentally fix that problem regarding the usage of unregistered LAV filters in the debug versions (LAVAudio-debug.ax and LAVSplitter-debug.ax)
MPC-HC no longer crashes when playing a playlist.
However, in a very weird way, the release versions of LAV are still crashing MPC-HC (LAVSplitter.ax and LAVAudio.ax).

[nevcairiel]

There are no real differences between debug and release, must be some weird side-effect.
Its still a MPC-HC bug, and should be fixed there.

[LordMerlin]

Hello.
Arose a question, I hope that the experts gathered here will clarify the situation.
Could anyone realties of you compare this one splitter (http://avmedia.su/) with what is discussed here. Pros, cons. And if the other is too general to be super.
Thanks in advance.

[nevcairiel]

They violate the ffmpeg license agreement. They use ffmpeg, but there is no word of a license anywhere, nor do they provide or link to the source of ffmpeg. From a quick look at the file, it also looks like they use parts of the MPC-HC splitters, which are released under the GPL and cannot be re-used without releasing the source. More license violations.

For me, thats enogh to distrust them and never consider taking seriously.
Also, its not written by me. :p

[dbone1026]

For me, thats enogh to distrust them and never consider taking seriously.

Those people smiling at the top of the page are enough for me not to trust them, it looks like an ad for a dentist. Plus, we all know serious coders don't smile!

[nevcairiel]

Plus, we all know serious coders don't smile!

Am i bad for smiling now when i read that? :rolleyes:
Those people are weird too. Its the same shit with MPC-HCs sourceforge page, get rid of stupid happy people. :p

I wonder whatever happend to the redesigned MPC-HC page..

[Aleksoid1978]

LAV Splitter - bad open Blind_Fury.m2ts (http://www.multiupload.com/ZGDZFMT4Q0) - very slowly opens and then black screen, with Cyberlink Decoder - crash at open.

[BelowSky]

There are no real differences between debug and release, must be some weird side-effect.
Its still a MPC-HC bug, and should be fixed there.

I'll look silly nevcairiel if I reported this bug to MPC-HC team.
I've more than 10 splitters and decoders in my system, including LAV debug versions, and they all don't have this issue. It's just LAV, and the release version in particular, have this problem.
However, I will fully understand if you don't fix it. In the end you are doing it for free.
So please don't consider this post some kind of whining or complaining

[boyumeow]

Am i bad for smiling now when i read that? :rolleyes:
Those people are weird too. Its the same shit with MPC-HCs sourceforge page, get rid of stupid happy people. :p

I wonder whatever happend to the redesigned MPC-HC page..

Hmmm, isn't a Happy nev, is a great programmer too :p.

Especially I saw nplayer.git which could have evolute into lavideo.ax :D.

[LordMerlin]

For me, thats enogh to distrust them and never consider taking seriously.

With this understood, but what about functional?
With this splitter you can make a solid connection container-splitter, decoder, and for different types of content, you can choose a different decoder.
Put and take a look. Is it possible that something like this do you have?

[nevcairiel]

Thats something the player should do, not the splitter. The splitter should not be made responsible for graph building.

[BloodySword]

I just tried a VOB LPCM file, and it works just fine.
Blu-ray LPCM also work perfectly. Other formats do not have "LPCM", MKV for example just has PCM, which does not need a decoder.

Strange, do you have an idea what could be the cause, or what I'm doing wrong? I even tried to build the graph by myself. When I rightclick on the LAVSplitter Audio output pin an say "render" it won't do anything. There is no filter wich accepts this output in my whole 32 bit DirectShow! Strange is, that the format for this pin says WAVEFORMATEXTENSIBLE, which is PCM, and has nothing to do with LPCM, right? But if the wFormatTag is not PCM, then it would be another issue.

[nevcairiel]

Strange is, that the format for this pin says WAVEFORMATEXTENSIBLE, which is PCM, and has nothing to do with LPCM, right? But if the wFormatTag is not PCM, then it would be another issue.

Please post the full media type, and if you can a sample file.

[VipZ]

x86 and x64 binaries are now available on SourceForge (https://sourceforge.net/projects/babgvant/files/LAVFilterSettings/), source (http://babgvant.svn.sourceforge.net/viewvc/babgvant/LAVFilterSetttings/) is too (not that it's very interesting :)).

Awesome, thanks :)

[BloodySword]

Please post the full media type, and if you can a sample file.

http://img34.imageshack.us/img34/4190/20110525181102.png

Should I cut a small piece of a VOB? Because of file size and copyright?

[nevcairiel]

Should I cut a small piece of a VOB? Because of file size and copyright?

A small piece would be great.

LAV Splitter - bad open Blind_Fury.m2ts (http://www.multiupload.com/ZGDZFMT4Q0) - very slowly opens and then black screen, with Cyberlink Decoder - crash at open.

Fixed.
Didn't test cyberlink, but the file plays fine otherwise now.

[BloodySword]

A small piece would be great.

Here you are. I'm from Germany FYI, but I write in english for the others. :>

http://www23.zippyshare.com/v/28534864/file.html

I hope MPEG2CUT did not modify any headers.

[nevcairiel]

Thanks, should be fixed for the next version.

[BloodySword]

Thank you very much. Did you find a problem?

[Sven75]

The following is no longer playing correctly since 0.27. It's still playing fine with versions up to 0.26-31:

http://www.mediafire.com/?1nr37wujo5g8c65 (cvid in mov container)
http://www.mediafire.com/?veeby6o9u9fb82m (debug log)

Decoder is current ffdShow...

Thank you!

[nevcairiel]

The file does not play with ffdshow at all here, not with 0.26 or 0.27.
The MS AVI decoder works fine, however!

[Sven75]

Just tried it again in graphstudio to double check and it is working just fine with ffdShow (lav versions 0.25 - 0.26-31) on my setup. I am currently using ffdShow builds 3842 - 3857.

http://www.mediafire.com/?ba3zq9l6zxj8c1f (screenshot)

[Inspector.Gadget]

Hi nevcairiel,

Thanks for your great work on this project. I've used it in conjunction with your CUVID decoder, MPC-HC as a frontend, and ffdshow as an audio downmixer for my stereo speakers and have found it to be much less prone to pathological behavior across a range of content than other decoders and certainly splitters (Haali, for example).

In upgrading from .26 to .27, I noticed that text subtitles in MKV were no longer displayed. Subsequent investigation revealed that "Only with matching language" had been checked in the splitter's config page with no languages selected; however, the existence of the subtitles was still exposed to MPC-HC's "navigate" menu so I assumed I was having some sort of video renderer issue. Does it make sense to change the behavior so users are somehow made aware of the fact that valid subtitles exist but won't be passed through using the splitter's current settings? I can see that the present behavior might confuse some users in the future, but perhaps the alternatives are worse. Thank you for your consideration of this issue.

[nevcairiel]

There have been no changes in default values or subtitling behaviour between 0.26 and 0.27.
The default configuration, since no languages are specified, should always show full subtitles (having no languages configured renders the "only with matching language" checkbox rather moot). Once you configure a set of languages, the checkbox makes sense, and will only show subtitles you have in your language list, and no others.

For me, this setting made the most sense.
- By default (without *ANY* configuration), just show the first subtitle stream it can find.
- After configuration of languages (and no other settings changed), just show the sub streams the user wants to see (based on the language preference).

[tetsuo55]

Does simply running the unregister.bat also delete all the settings?

[Sebastiii]

Hi,

This is my test with cppcheck, i don't know if it help or it's totally wrong lol :)
http://dl.dropbox.com/u/10536084/lav/cppcheck/Lavsplitter_cppcheck_Clean.txt
Thanks,
Sebastiii

[nevcairiel]

Does simply running the unregister.bat also delete all the settings?

No, only the (un)installer can remove settings.

[CruNcher]

Thats something the player should do, not the splitter. The splitter should not be made responsible for graph building.

I guess they override Vista/7 restrictions doing it this way without the need for manipulating the registry ?
Though it's strange no contact information no forum no nothing just a change log some bad translated help and that was it,that this comes as a executable most probably somewhere out of east europe (russia) also should make some nervous :D

[madshi]

No, only the (un)installer can remove settings.
FWIW, if you wanted, you could delete settings via batch file, too ("start /min reg delete HKEY_CURRENT_USER\Software\LAV Whatever /f").

[BelowSky]

When I seek in an Ogg audio the sound stops working (MPC + LAV filters).

http://samples.mplayerhq.hu/ogg/Vorbis/coyote.ogg

[BloodySword]

Vorbis in ogg is a weird format. It seems that it uses variable frame lengts and sometimes <1 ms... Seeking is a bit slow for me, it takes about 500ms till the playback begins on the position i clicked. But this was with EVERY splitter I EVER used. I have the problem since I have the laptop with crappy Realtek HD Audio Chipset! With other splitters such as Haalis etc. sometimes playback broke after seeking. The problem is the crappy drivers of crappy crap Realtek crap chip... ._.

[BelowSky]

I can play and seek in Ogg just fine with MPC and Haali, combined or individually.
I can play and seek them even with ffplay.
BTW I don't have a Realtek chipset.

[nevcairiel]

Seeking in ogg files should be improved for the next version.

[BloodySword]

I think the main problem with vorbis is, that the decoded frames are piped as is. So the buffer is the decoded frame. Since there are very small frames, some audio chipsets can get problems with this. So the main problem is on decoder side: The decoder has to group very small frames together in one buffer. Then the problem should be solved.

[nevcairiel]

LAV Audio already does this, TrueHD for example also has very small frames, 0.833ms to be exact. There was another problem with seeking in ogg.

[BloodySword]

Nice to know. Can it affect the behavior of ffdshow audio decoding when you fix this? So, that the framing is more solid?

[nevcairiel]

The problem i found was in my audio decoder, not in the splitter. with ffdshow audio decoding, i never got the silence.

[BloodySword]

Okay I think I will switch to your vorbis decoder and TrueHD decoder then. For effects, ffdshow connects to the decoded raw samples. Can't wait for new version! :)

[Sebastiii]

Hi,

Nevcairiel, i don't know if you have follow IRC yesterday :)

1)
I have (it seems) found what crash Mediaportal.
I have also add GC cleanup in "Cleanup" method of MP to free memory when using FFDSHOW Video Decoder.
I have tested on 30 sample play/start one by one but very quickly and no crash and with and without reclock :) (Tested on MKV and M2TS).
My building test was easy to make, i have create directory MKV and M2TS and put all my sample in each directory, in MP we can select the folder and select to play all file in it.
So start to play and i go on next file (like sample start to play a few second and goning to the next and so on).
All crash seems related to bitstream function (i don't know why and even more on TrueHD).
If i uncheck all bitstream decoder, all PCM goes nicely with and without reclock and no crash :)
I don't know how i can help you for helping us to catch error and solve it (maybe it was related to my AVR but with fddshow audio, it didn't crash).

2)
Now, i have also somes questions :) and i need your expert confirmation :)
Related to bitstream and PCM.
I have sample 7.1 DTS-HR (didn't work on bitstream with 7.1 channel, it's detected as 5.1(DTS-MA) but works perfectly on 7.1 with PCM and Reclock Wasapi or not Reclock.
I have E-AC3 sample (Working in BITSTREAM as 7.1 channel) but not in PCM (detected as 5.1 Channel).
On my mind, i was thinking that Bitstream is the best mode to get perfect audio (because no alteration) but i'm not sure lol, PCM with Wasapi is it really good ? (from my hear -> yes it seems).

Yes with all my test, i have detect this 7.1 bitstream or not detection.
I'm using Licenced Arcsoft TMT so i use the right DLL (1.1.08).

3)
And the last :)
I have this sample (surely broken, i can't seek, playback stop) but working in seek with other splitter, i was thinking that is work with older LAVF version but not sure lol : http://dl.dropbox.com/u/10536084/lav/Sample/Therapy_SwitchAudio.m2ts

A big Thanks :)
Sebastiii.

PS : sorry for the long post and hope that my explanation was enough clear :)

[LordMerlin]

I guess they override Vista/7 restrictions doing it this way without the need for manipulating the registry ?
Though it's strange no contact information no forum no nothing just a change log some bad translated help and that was it,that this comes as a executable most probably somewhere out of east europe (russia) also should make some nervous :D

As far as I myself understand there are 3 ways to override the connection:
1. Media type in the tab Output. (Audio not working)
2. Download the selected filter tab Graph Builder.
3. Compulsory connection to the filter tab Connections.

And yes, a developer from Russia. Therefore, the discussion forums is available only in Rousseau, unfortunately.
We will be glad to see here http://www.forum.team-mediaportal.ru/index.php/topic,1939.0.html and here http://forum.ixbt.com/topic.cgi?id=10:58214-111#3692

About the use of someone else's code the developer wrote the following:
Partly pieces of code from ffmpeg, other libraries, music players, splitters, codecs, razumetso present, but naturally in a highly processed form.
It is normal practice, and everybody does it. From scratch software was written years since 20 years.

[nevcairiel]

It is normal practice, and everybody does it. From scratch software was written years since 20 years.

Sure, but you have to give credit to the people you use code from, and if you use GPL code, you have to be GPL compatible yourself. Otherwise, its license violations, and plainly illegal.
You can't just go around and "use" all the code you find. You have to obey to the licenses that code was released under, and if you don't, its stealing of intellectual property.

Even if the code is completely free (public domain, etc), its just common courtesy to give credit to the people that did all the work for you. Just using their work and claiming it your own without comment is disrespectful and just arrogant.

[Sven75]

Did some further testing regarding the CVID file from #3461 not playing with ffdShow anymore on my setup. And the breaking change is probably the minor ffmpeg update right before the release of 0.27.

As I suspected ffmpeg to be the cause, I replaced the according dlls from 0.27 one by one with those from 0.26-31 (I know it's brute force) and found out that avformat-53.dll is responsible.

Now, I know that it may very well be ffdShow's fault and - of course - I will bring it up there it, if this is the case, but maybe you find the time to have another look - just in case that it is ffmpeg in LAV that is broken with this specific format...

[nevcairiel]

LAV Filters 0.28

LAV Splitter
- Improved PGS filtering, forced subtitles should be cleared off the screen properly more often.
- Improved playback of H264 in MPEG-TS with certain files (mostly from camcorders, but some Blu-rays as well)
- Fixed 16-bit LPCM on DVDs
- Fixed the Media Types for uncompressed video

LAV Audio
- Added the ability to decode DTS Express when using the ArcSoft DTS Decoder
- Fixed missing audio after seeks with Ogg Vorbis


Download: Installer (both x86/x64) (http://files.1f0.de/lavf/LAVFilters-0.28.exe) -- Zips: 32-bit (http://files.1f0.de/lavf/LAVFilters-0.28.zip) & 64-bit (http://files.1f0.de/lavf/LAVFilters-0.28-x64.zip)

DTS Express
DTS Express is a special low-bandwidth format used on Blu-rays for PiP commentary tracks. It consists of only HD frames, and therefor cannot be decoded by ffmpeg. You can only decode it with the ArcSoft DTS Decoder right now. Bitstreaming it is also not supported at this time.
Still, this is the first time i see it actually decoded outside of TMT itself, the standalone ArcSoft decoder crashed when i fed it to it. :(

All other changes are more or less bugfixes. The PGS filtering should hopefully no longer keep some forced subs on the screen for way too long, but i didn't watch a full movie to confirm, just did some quick tests on known problem spots.

Have fun!

[BloodySword]

Alright, LPCM works now with ffDShow, so I save one filter in graph. Thank you!

But what about stream switching? How can I get this functionality on players wich don't support it? ffDShow does not show anything. :o(

[Gleb Egorych]

nevcairiel, thanks for the new version!

1) 0.28 plays fine my Panasonic AVCHD files while 0.27 doesn't.

2) Re-sync issue on AVI files (http://forum.doom9.org/showthread.php?p=1501306#post1501306) is fixed with (I guess) that ogg patch.

3) Zoom Player (http://forum.doom9.org/showthread.php?p=1501013#post1501013) hang problem was fixed along with AVI crash issue after pre-0.26 patch.

4) LAV Splitter +ArcSoft Audio Decoder (http://forum.doom9.org/showthread.php?p=1502459#post1502459) problem on M2TS with E-AC3 still persists. It appears after seek, and if you seek to a middle position in a M2TS file. Although this combination of container, audio format and decoder is not usual I think it is good to fix it because the bug may exibit is some other place (like fixed bugs 2) and 3)). Sample file M2TS + E-AC3 can be found here (http://forum.doom9.org/showpost.php?p=1343585&postcount=5103).

Thanks again, your work is greatly appreciated!

[Thunderbolt8]

dts express support is nice, never could get it to play before (in remuxes)

[nevcairiel]

4) LAV Splitter +ArcSoft Audio Decoder (http://forum.doom9.org/showthread.php?p=1502459#post1502459) problem on M2TS with E-AC3 still persists. It appears after seek, and if you seek to a middle position in a M2TS file. Although this combination of container, audio format and decoder is not usual I think it is good to fix it because the bug may exibit is some other place (like fixed bugs 2) and 3)). Sample file M2TS + E-AC3 can be found here (http://forum.doom9.org/showpost.php?p=1343585&postcount=5103).

I tested with the only E-AC3 in m2ts i have, and it decodes just fine with the ArcSoft Audio Decoder HD.
The sample links you posted are dead.

[Gleb Egorych]

Sent a new link.

[nevcairiel]

Thats the same file i already had, and it decodes just fine with the ArcSoft DD+ decoder. I get audio out of all channels, with or without seeking.

[Snowknight26]

Another sample with E-AC3: http://stfcc.org/misc/transformers.sample.m2ts

[Andy o]

Is that a 7.1 sample? For the record, I've always had this problem after seeking with MPC-HC too. jmone also confirmed it. I haven't been at home this whole week so I can't test right now.

[Gleb Egorych]

Thats the same file i already had, and it decodes just fine with the ArcSoft DD+ decoder. I get audio out of all channels, with or without seeking.
I use the following ArcSoft files:
ASAudioHD.ax v1.3.3.219 26.07.2010
DdpDec71.dll v1.0.1.12 16.04.2010
Tested in Zoom Player. After seeking left channel is mapped to left, center to right, right to center, the rest is muted.

[nevcairiel]

I'm using 1.3.3.273 (from TMT3.190) and 1.0.1.12, and i experience no such issues.

[Andy o]

btw, the transformers sample above appears to be TrueHD 5.1

[nevcairiel]

I did some more testing with this, and it looks like the older versions of the TMT decoder are indeed having this problem. But i don't think there is anything i can fix.
It works with the MPC-HC splitter, because its stupid and tells the decoder that its AC3 instead of EAC3, so my guess is that its actually only decoding the normal AC3 part of the stream when using the MPC-HC splitter, but not the E-AC3 part that adds the 2 extra channels.

Anyway, its fixed in the decoder from TMT3.190 - use that.

[ney2x]

0.28 fixed "thousand glitches" problem with madvr 0.61 :) i don't know how but I'm happy...

[Gleb Egorych]

It works with the MPC-HC splitter, because its stupid and tells the decoder that its AC3 instead of EAC3, so my guess is that its actually only decoding the normal AC3 part of the stream when using the MPC-HC splitter, but not the E-AC3 part that adds the 2 extra channels.

Anyway, its fixed in the decoder from TMT3.190 - use that.
You're right. With MPC-HC splitter ArcSoft outputs 6 channels while with LAV Splitter it outputs 8 channels.

The problem with ASAudioHD.ax v1.3.3.273 is that it's limited to 16/48 output.

Anyway it looks like bug in ArcSoft decoder and not in LAV Splitter. Thanks for the tests and the explanation.

BTW am I right that LAV Audio does not decode E-AC-3 extension to an AC-3 core in M2TS files? I read somewhere ffmpeg does not support E-AC-3 over 5.1 config.

[nevcairiel]

BTW am I right that LAV Audio does not decode E-AC-3 extension to an AC-3 core in M2TS files? I read somewhere ffmpeg does not support E-AC-3 over 5.1 config.

Thats right. The format is very rare, AFAIK it only exists on the Dolby Demo Blu-ray, so there is no need to invest much more time in it.

[Andy o]

Not that this is your problem, but just wanted to say that with the asaudiohd.ax from .190 it's even worse for me. I get the channel wrong/silent issue even when not skipping. With .185 at least I got it correct when playing from the beginning.

[BelowSky]

Thank you for the Ogg fix
And regarding amr, are there any news on replacing the ffmpeg native amr with OpenCore? Playing an amr file containing dtx mutes entire system sound. That's until I stop the amr playback.
http://www.mediafire.com/?pi6ev8wagqaaq2u

[BloodySword]

In my system, your AMR is played back correctly with ffdshow libavcodec decoder.

[ney2x]

In my short stay here at doom9, here's the summary:

LAV Filters - perfect
LAVCUVID - near to perfection
madvr - perfect
Reclock - perfect
MPC-HC - perfect if I could replace the windows 95 gui/skin
and last but not the least.... LAV Player - the question is when???

[BloodySword]

I found a small bug, not realy a problem for normal user but for me: Sometimes I discover some videos played back slower. When I change the IMediaPosition->Rate property, A/V is out of sync because video is much slower than audio. Jitter correction is active in ffDshow - does not work.

[CruNcher]

Maybe you stumbled over this current limitation http://forum.doom9.org/showthread.php?p=1502471#post1502471 ?

[BloodySword]

The strange thing is, that if I use a rate <1 (slower) video plays ~2x slower than audio. If I set it to >1 (faster) then video plays ~2x faster than audio.

[nevcairiel]

Like mentioned up in that post, rate adjustments are currently not supported. It may be added in the future. Any behavior when you change rate is purely random.

[pirlouy]

Sorry if it has been asked before but... Is it possible to bitstream DTS-HD only, not simple DTS ?

Maybe in a future version, or it's not recommended to do this ?

[Boltron]

Sorry if it has been asked before but... Is it possible to bitstream DTS-HD only, not simple DTS ?

Maybe in a future version, or it's not recommended to do this ?

For me, DTS bitstream works fine with MPC-HC, LAV Splitter/Audio and madVR. All DD and DTS work fine. The only one I have had a problem to bitstream is PCM.

[Andy o]

For me, DTS bitstream works fine with MPC-HC, LAV Splitter/Audio and madVR. All DD and DTS work fine. The only one I have had a problem to bitstream is PCM.

The only way to stream bit-perfect LPCM with Win 7 and Vista is to use WASAPI exclusive. With most 32-bit DirectShow players, you can do that with ReClock.

[nevcairiel]

You cannot/do not "bitstream" PCM. Bitstreaming refers to sending the compressed audio to the receiver, so it can decode it. PCM is not compressed, its already uncompressed audio. So you do not need to "bitstream" it. But like Andy said, the closest you can get is to use a exclusive mode WASAPI renderer, so it by-passes the windows mixer and output the audio "as is".

[neoufo51]

I have a question:

Many people ask me why LAV Filters don't include support for video decoding and I turn them to CUVID if they have Nvidia cards but for the people who don't have compatible hardware, do you plan to support video decoding (LAV Video) in a future release of LAV Filters to go with LAV Audio and thus give people a basic alternative to ffdshow?

[nevcairiel]

Its planned, but there is no ETA.

[BloodySword]

Like mentioned up in that post, rate adjustments are currently not supported. It may be added in the future. Any behavior when you change rate is purely random.

I hope it does not have a low priority because for some this feature is essential. I can use VLC so far, but it is a little bit buggy in this point (crashes sometimes, adjustments not really accurate).

[nevcairiel]

It has a pretty low priority. I focus on perfect playback/watching experience, not artificial circumstances where you would need slower or faster playback.

[BloodySword]

You could lock the rate like the Windows Media DMO filters till the development of this feature is done. Do you understand what I mean?

[Kaotech]

Are you sure Wasapi is bit perfect ? If your track is a 48khz sample and your soundcard is 44khz, wasapi change your soundcard at 48khz ? I know with ASIO it do that, but with wasapi i'm not sure.

[SamuriHL]

WASAPI exclusive bypasses windows audio and sends the PCM audio to your receiver as is. It doesn't mess with it. It's as close to "bitstreaming" PCM as you're going to get.

[nevcairiel]

As long as your sound card supports 48Khz output, WASAPI exclusive mode will switch it to 48khz and output the PCM untouched. If it doesn't support 48khz, it'll just fail, and the application has to take care of converting the audio before output.

[BelowSky]

In my system, your AMR is played back correctly with ffdshow libavcodec decoder.

ffdshow doesn't use libavcodec for amr as far as I can tell. And this isn't really about LAV not able to playback amr which is okay. It is about LAV filter muting the system sound.
To give you an example, let say you are on Skype, and you opened one of these dreaded amr files, the sound in skype will just stop working until you stop the playback of that amr.

[nevcairiel]

LAV Audio does not do anything to the system sound. It just sends decoded audio to the renderer .. or it does not, if decoding fails. It does not communicate with the system at all.

[BelowSky]

Since you have dvbviewer can you open it and watch any channel and then open my uploaded file with another player and wait for dtx code to kick in (around the 16th second).
If that didn't affect the sound in your system, then it will mean that the problem MUST be related to my "USB SPEAKERS".

[Boltron]

It has a pretty low priority. I focus on perfect playback/watching experience, not artificial circumstances where you would need slower or faster playback.

"I focus on perfect playback/watching experience" Ahhhh... :)

<3 Nevcairiel.

[Gleb Egorych]

BelowSky, I also loose sound after 16 sec on my Audigy 2 ZS with that amr file.

[BloodySword]

ffdshow doesn't use libavcodec for amr as far as I can tell. And this isn't really about LAV not able to playback amr which is okay. It is about LAV filter muting the system sound.
To give you an example, let say you are on Skype, and you opened one of these dreaded amr files, the sound in skype will just stop working until you stop the playback of that amr.

http://img832.imageshack.us/img832/6160/20110530195456.png

ffdshow rev3851 icl10

Hmm you better should not use skype, it's AdWare. It installs software wich you didn't want to install even if you press "no!".

Wich kind of sound hardware do you have? Can it be that it runs in half duplex mode?

It could be a "feature" of Skype. Check the settings.

You could try if Skype is even muted when playing back 8000 Hz PCM. I have a bad suspicion.

[assassinhtpc]

Update: I was able to get LAV working with Live TV via WMC today. No issues at all with the new release.

Quick question: If I have DTS-HD and DTS both available in my MKV file how do I switch between the two using WMC?

[BelowSky]

The problem isn't Skype. The same thing can happen also if you open two instances of MPC-HC.

http://img832.imageshack.us/img832/6160/20110530195456.png
ffdshow rev3851 icl10


You can check the source code of ffdshow, look for the file libamr.c

[BloodySword]

The problem isn't Skype. The same thing can happen also if you open two instances of MPC-HC.



You can check the source code of ffdshow, look for the file libamr.c

Why should I? It's not my fault that they named it wrong. And for me it works.

Did you test it to play WAVE-File with 8000 Hz?

[hoborg]

This sample (https://rapidshare.com/#!download|169l35|1479703935|crash.zip|26215|R~0) will crash graphstudio if LAVF splitter + FFDshow DXVA decoder is in use, working fine with MPC-HC mpeg splitter.
More info (http://forum.doom9.org/showthread.php?p=1504485#post1504485).

[nevcairiel]

Crashes in a decoder are the decoders fault, unless something is going terribly wrong. If it crashes simply because it receives the H264 data a tiny bit differently then from other splitters, there is nothing i can do to fix it, especially when its buggy ffdshow.
A decoder needs to have some kind of error resilience against broken streams, no matter how they arrive. The MPC-HC DXVA decoder also manages..

[Andy o]

Re: rate change ("slower/faster playback").

Can't ReClock do this already? Or is the question about something else?

[hoborg]

Crashes in a decoder are the decoders fault, unless something is going terribly wrong. If it crashes simply because it receives the H264 data a tiny bit differently then from other splitters, there is nothing i can do to fix it, especially when its buggy ffdshow.
A decoder needs to have some kind of error resilience against broken streams, no matter how they arrive. The MPC-HC DXVA decoder also manages..

I wrote it here, becouse i didnt noticed this issue before, so i did some tests:

LAVF: (+ FFDShow DXVA decoder r.3863)
.22 - no crash but choppy playback
.23 - introduce black screen
.28 - no more blackscreen, but crash in graphstudio

[skingery]

Update: I was able to get LAV working with Live TV via WMC today. No issues at all with the new release.
Interesting. So are you seeing an improvement of some kind that make it worth the effort?

[Aleksoid1978]

Can you make a Debug version.

P.S. - sorry, find :)

[nevcairiel]

I wrote it here, becouse i didnt noticed this issue before, so i did some tests:

LAVF: (+ FFDShow DXVA decoder r.3863)
.22 - no crash but choppy playback
.23 - introduce black screen
.28 - no more blackscreen, but crash in graphstudio

The changes in 0.28 are necessary to fix exactly that black-screen problem on alot of files. ffdshow is buggy, always was and always will be (especially its DXVA mode, which is a bad copy of MPC-HCs DXVA, introducing new bugs and skipping functionality)

The MPC-HC splitter for example will fail completely on alot of files that LAVSpitter will play, and due to supporting more "weird" files, the output on such weird files is slightly different. If a decoder doesn't cope with that, its not my responsibility to fix it.
I do have some plans for the H264 parser, but i cannot say if it'll help, and i don't know when i'll work on that.

Last i heard, the MPC-HC DXVA decoder can deal just fine with it.

[jmone]

nevcairiel, I've been testing subtitle handling for MC16 latest beta changes with V0.26 of LAVSplitter (you opinions in that thread would be welcome) using Avatar to test Forced Subs. I notice that at the 47:00 minute mark is the forced sub "Why me? That's not fair...", and this one hangs around for a very very long time (eg till the next forced sub at around 53:45). I know you touched on the logic for this previously...

Forgot to report back (sorry) - Forced Subs are now being cleaned of much more promptly, eg while the forced sub "Why me? That's not fair..." at the 47:00 minute mark in Avatar still overruns the dialog it is now removed after about 10 seconds instead of 7 minutes!

[nevcairiel]

You know, that specific sub is working just fine for me, and already did before the change. :p

[jmone]

I'm sure it is down to ffdshow then (and another reason for LAVVideo)! Out of interest (and not wanting to stir any pot), what is the story with ffmpeg vs libav and which "fork" are you following (or have I got this wrong)?

[nevcairiel]

I actually tested with both MPC-HCs internal renderer and ffdshow.

ffmpeg is the original ffmpeg, no fork or anything. Earlier this year a group of developers tried to take over ffmpeg by forcefully removing the project leader and placing themself in that position. However, they failed as eventually the original project creator confirmed the ffmpeg project leader and gave him the rights to keep the ffmpeg name. After that the developers that tried to take over split from the project and forked it as "libav". It did however seem like they just couldn't be the losers in a pissing contest, so they tried to take their grief out on ffmpeg, taking away all their resources without the ability to properly backup and move them to a new server, which caused a disruption in the mailing list and other services.

I'm following ffmpeg, for the simple reason that i think they actually behave like i would except a FOSS project to behave, and not react with insults, bans, threats and legal claims like libav did. In addition, ffmpeg is actually merging the improvements from libav (as much as possible anyway, the truly broken changes won't be merged, of course), that way i can be sure to get the best of both. Personally i think Michael Niedermayer (the project leader of ffmpeg) is doing an outstanding job keeping ffmpeg stable, despite the plainly broken or wrong changes coming in from the libav side (as demonstrated by the recent changes to swscale, which severly broke ffdshow for a while as clsid just merged libav changes)

This is only my opinion on the whole situation, of course, and this is absolutely not the place to debate which side was "right", so if anyone feels strongly about the whole thing, get your own thread to debate it.

[BelowSky]

Since you have dvbviewer can you open it and watch any channel and then open my uploaded file with another player and wait for dtx code to kick in (around the 16th second).
If that didn't affect the sound in your system, then it will mean that the problem MUST be related to my "USB SPEAKERS".

To answer my own question, yes, without a doubt, the culprit of that system silence is my "below the average" USB speakers. :)

[CruNcher]

just for the Info Cyberlink fixed those crashes with the Canal HD .m2ts sample from v0lt it was indeed their Decoder issue, it works now with lav splitter ;)
I gone again through the whole test sample database .ts not 1 problem :) (only the missing program stream selection, missing feature in that sense no bug) but @ least playback wise it crunched through everything without sync issues just amazing (no option changed), ok their is a issue but it's not really a issue in itself of lav splitter but more of lav audio with damaged latm streams in .ts you wont hear any audio it will skip it (playing the video norma) while with other decoders supporting damaged latm (Mainconcept,DivX,Cyberlink) will skip the video parts and advice the decoder to jump to the latm timestamp parts that are ok (looks funny your video being forced to skip ahead) visible in the audioframedamaged.ts sample ;) :)
Ah yeah and the Encoding issue which is a little sad not be able to use lav splitter efficient for .ts transcoding yet, @ least for now with the tested 30fpslock .ts stream provided :(


First Play Experience (straight after load no interaction):

.ts Mpeg-2/VC-1/H.264 = 100% (DXVA/ON off switching works 4:2:2/4:2:0, flag based deinterlacing,soft ivtc works)
.m2ts Mpeg-2/VC-1/H.264 = 100%

[nevcairiel]

lav audio with damaged latm streams in .ts you wont hear any audio it will skip it (playing the video norma) while with other decoders supporting damaged latm (Mainconcept,DivX,Cyberlink) will skip the video parts and advice the decoder to jump to the latm timestamp parts that are ok (looks funny your video being forced to skip ahead) visible in the audioframedamaged.ts sample ;) :)

Just because the audio is broken, the video can still be shown, skipping it is weird. I rather prefer this method.

Anyhow, judging from log messages, that sample also uses AAC SSR, which is not supported in ffmpeg yet. Nothing i can do.

[CruNcher]

Hmm could you force this non skipping behavior maybe from the supporting audio decoders ;) ?

[nevcairiel]

no, that depends entirely on how the decoder handles the situation.

[BloodySword]

To answer my own question, yes, without a doubt, the culprit of that system silence is my "below the average" USB speakers. :)

Oh I didn't know that you have USB-Speakers. I think there are problems with resampling by software for such speakers. Means you can not play back more than one audio with same samplingrate or different samplingrate, who knows... xD

I recommend you to boy a cheap mini amplifier for home usage and 2 cheap speakers with ok values 50 - 20000 Hz 40W are very cheap, and they are wooden! :>

[Andy o]

If it's a resampling issue, have you tried forcing upsampling to something reasonable with ReClock? If your system is being muted because a very low sampling rate PCM is being sent to the audio device, then that could solve it. Also, are you using WASAPI exclusive? Cause the Windows mixer should take care of the resampling issue, unless you're using exclusive mode.

[BelowSky]

Calm down people... I'm much simpler than that. I just stick the speakers USB cables and listen, that's it.
I actually enjoy listening with these cheap speakers as much as I enjoy watching videos with my cheap screen.
My only purpose with LAV - and audio/video in general - is just that I want it to JUST PLAY.

[CruNcher]

@nev
http://www.mediafire.com/download.php?8w2ecz4jkqqrkcc http://www.mediafire.com/download.php?uaa942eka05iku2 <- No Audio (Latm) both play fine with sound in mplayer i missed those as they had the wrong extension .mpg
Guess the problem is that lav splitter detects them as mp2 and mp1 in the streams and not AAC

[jimwhite]

So what's the consensus on the Arcsoft files required? Which are the best versions of dtsdecoderdll.dll, ASAudioHD.ax, and DdpDec71.dll to use for full HD audio support with no sound degradation from downsampling, etc.

:confused:

[Andy o]

I think nev has said that the dtsdecoder.dll from TMT2 works best (1.1.0.0 IIRC). For DD+, it's highly unlikely that you'll need to decode 7.1 any time soon, so I'd say just use LAV audio to decode that. Nev said he's not having trouble using the ASAudioHD.ax from TMT5, but I am, so you can test yourself. But I think that Arcsoft ASAudioHD.ax downsamples (I might be wrong bout that). So, you either are limited to 16/48, or you have problems after seeking into the file, when using the older .ax. Me, if I need DD+ 7.1, I can just bitstream (for the first time with ATI cards, with LAV audio).

[nevcairiel]

E-AC3 is limited to 48kHz anyway.
I'm not sure if ffmpeg supports decoding 7.1 E-AC3. I have never seen such a track. Only 7.1 E-AC3 i have are those Blu-ray tracks, which are a whole different beast, and not used in real-life.

For DTS
- 1.1.0.0 works fine for all formats
- 1.1.0.7 is broken on Mono and 6.0
- 1.1.0.8 is only broken on 6.0

So 1.1.0.0 is the best, however 6.0 is a really rare format (and 6.1 works fine again), so in 99% of the cases, 1.1.0.8 will be fine as well.
1.1.0.7 is from TMT3.185, 1.1.0.8 comes with TMT5, 1.1.0.0 is from TMT2, i think.

Note that 1.1.0.0 needs some special setup if you're using Win7, as it requires the 2003 edition of the MS C runtime, which does not have an installer for Win7, for some reason (msvcr71.dll and some other dll)

[Andy o]

oh, I'm on Win 7 and I put 1.1.0.0 in the LAV folder, but haven't tested it yet, just assumed it worked right off the bat.

[nevcairiel]

You need to get msvcr71.dll and msvcp71.dll and put them in the System32/SysWOW64 folder (on 32-bit or 64-bit systems respectively). You can google for those, multiple places to obtain them should show up.

[Andy o]

thanks, will try that.

[Andy o]

Hmm for some reason I already had those files in the SysWOW64 folder, I think PowerDVD and/or TMT installs them.

[nevcairiel]

Yeah some software obviously installs them there. Having only TMT5 installed didn't provide them, though.

[jmone]

nevcairiel - IMHO it looks LAVSplitter and LAVAudio have pretty well reached a stable V1 stage with "Planned Features" down to:
- Improved BluRay support (Title selection from within the player) : this would be great and I presume requires both the exposing of the runtime IF for players to use (as with LAVAudio)
- Audio Up/Downmixing : I know others are keen but for some (eg MC'ers) we already have the DSP functions we can use

If you are feeling "bored" and looking for suggestions ;) to put all that talent too I'm sure a similar clean approach for a ffmpeg LAVVideo style directshow filter would be welcomed by all! I know you have hinted at such a possible project but I certainly know that getting Video Decoding with good De-Interlacing and Subs is still the bane for many.

Thanks again for all your work on LAVFilters to date they really have made a great leap forward in usability for high quality HTPC use.

[nevcairiel]

Video Decoding with good De-Interlacing and Subs is still the bane for many.

As i've mentioned before, subtitles is not something the video decoder should be doing. In a perfect world, the renderer would blend the subs on top of the rendered video in its full target resolution and color space. So any Video Decoder i would write will most likely not support subtitles.

For deinterlacing - implementing something like yadif would certainly be nice, but we'll have to see if that is in the cards.

[jmone]

So in theory, say there was a LAVVideo + Deinterlacing (and YADIF if the best of the de-interlacers offered in FFDSHOW IMO) what filter (if any) existis that could be used to render the Subs or would this be something that would need to be added to an existing Video Renderer (eg say by madshi to madVR for example)?

[nevcairiel]

There is two approaches to subtitles.
Either, you have a dedicated subtitle parser/renderer, which renders the subs onto a plain surface, and the video renderer just takes that surface and then blends it over the final video image. This is how its basically done in MPC-HC when using the internal renderer, in combination with EVR-CP, madVR and some other compatible renderers. But sadly, the MPC-HC implementation is not really portable, and you cannot make it a stand-alone.

The second approach would be that the video renderer takes care of subtitles completely, this however would mean it only works with that specific renderer.

[jmone]

Thanks - posted on the JR MC beta forum to see if they have any interest in adding this to MC

[SamuriHL]

Man I hope so. Cause right now with the state of MC16 I don't have subs working at all. :( ffdshow raw video filter causes errors as I don't think it got properly added to the whitelist. Since I use LAV CUVID on my nVidia machine, I'm pretty much resigned to not using MC16 until it's fixed.

[RobertinoM]

I think nev has said that the dtsdecoder.dll from TMT2 works best (1.1.0.0 IIRC). For DD+, it's highly unlikely that you'll need to decode 7.1 any time soon, so I'd say just use LAV audio to decode that. Nev said he's not having trouble using the ASAudioHD.ax from TMT5, but I am, so you can test yourself. But I think that Arcsoft ASAudioHD.ax downsamples (I might be wrong bout that). So, you either are limited to 16/48, or you have problems after seeking into the file, when using the older .ax. Me, if I need DD+ 7.1, I can just bitstream (for the first time with ATI cards, with LAV audio).

Just to figure out clearly

in other posts it was always mentioned "just drop dtsdecoder.dll into LAV install folder" but this sounds more like

take the Arcsoft-files, register ASAudioHD.ax and add. drop the

dtsdecoder.dll into the folder....

can sombody please make a clear statement about needed files??

regards robertino

[hoborg]

Just to figure out clearly

in other posts it was always mentioned "just drop dtsdecoder.dll into LAV install folder" but this sounds more like

take the Arcsoft-files, register ASAudioHD.ax and add. drop the

dtsdecoder.dll into the folder....

can sombody please make a clear statement about needed files??

regards robertino

just drop dtsdecoder.dll into LAV install folder :D

[SamuriHL]

To clear this up once and for all....is everyone paying attention? :)

FOR DTS-HD MA decoding, ALL you need to do is drop the dtsdecoderdll.dll in the LAV folder. PERIOD! No AX, no registering, no fruitcake settings.

FOR ANY OTHER DECODING SUCH AS EAC3, you must register the AX file and use that in your player of choice. You will need to disable EAC3 decoding/bitstreaming in LAV Audio (assuming you use LAV Audio at all in that scenario). And you will need to add fruitcake settings.

But as has been said before, what's the point of that? What does doing the fruitcake settings and all that nonsense gain for you? Nothing. Use LAV Audio and the dtsdecoderdll.dll and you're covered in every real world case you're going to encounter. No hassle.

[RobertinoM]

As I prefer a system with only a few necessary codecs

I LOVE LAV Filter as actually evereything on my system is working like a charm..

so again a big THANK YOU NEV for this gift.

Robertino

[jimwhite]

Hopefully the last question about the Arcsoft files.... what are "fruitcake settings" ????

:confused:

[SamuriHL]

If you don't know what they are, then you don't need to know. :) For those that do, it refers to the idea that you have to screw with win.ini to make the channel setting stick when using the registered ax filter. I call it fruitcake settings cause it's really an ugly hack. It works, but, it's insane. Nev's solution doesn't require that nonsense.

[noee]

.... what are "fruitcake settings" ????

:confused:

win.ini :devil:

[BloodySword]

As i've mentioned before, subtitles is not something the video decoder should be doing. In a perfect world, the renderer would blend the subs on top of the rendered video in its full target resolution and color space. So any Video Decoder i would write will most likely not support subtitles.

For deinterlacing - implementing something like yadif would certainly be nice, but we'll have to see if that is in the cards.

ffdshow already has Yadif included. If you place the Subtitle filter after the deinterlacer, you're happy!
Even if it is not in any card.

[Carpo]

Would it be possible nevcairiel, in a future release of LAV to allow it to find the required arcsoft dlls from a path in the PATH option under the system variables, i only ask this as eac3to complains i don't have the required files in its folder, and when i do put them there and register the .ax file there, LAV doesn't see it.

Unless i am doing it wrong ;)

[SamuriHL]

Would it be possible nevcairiel, in a future release of LAV to allow it to find the required arcsoft dlls from a path in the PATH option under the system variables, i only ask this as eac3to complains i don't have the required files in its folder, and when i do put them there and register the .ax file there, LAV doesn't see it.

Unless i am doing it wrong ;)

You are. :) If it's not in the LAV directory, it'll search the system path for it. So, add your eac3to folder to the path and it SHOULD, in theory, find it. If it doesn't, there could be a bug.

[Carpo]

eac3to is already there, i will test about a bit more, installed TMT5 earlier and then copied the whole codecs folder out, why have the whole thing installed when you just need a few things from it :p

[SamuriHL]

Is it in a SUB folder of the eac3to folder or at the same level as the exe? Cause the path would require it to be in the directory that's in the path, not a sub folder under it. If that makes any sense.

[Carpo]

nope, its int eh same folder as the .exe - prob something i have done, have been cleaning up the pc lately :sly:

[SamuriHL]

LOL! :) Well, it should be picking it up then afaik. I just took the one dll and dropped it into the LAV folder and called it good. I use TMT5 on episode discs anyway, so, it's not a big deal for me. I suppose I could see if it found it in the TMT dir. :D

[robpdotcom]

Could you not just copy the dll, and place it in both directories?

[SamuriHL]

Well that's what I do, but, I think he's trying to avoid having it in multiple places.

[Carpo]

when i did that LAV wouldn't pick it up, i had to register the .ax files too (as i dont need to have TMT5 installed as i use XBMC, although i own TMT5, i just bought it for the files ;) ), and if i did that in both places, one would pick it up but the other wouldn't, so i did just that, re-registered and its finding it in the eac3to folder now as its saying dca under status, guess all the copying about i got confused or did something else wrong.

All is well now :)

[SamuriHL]

You do NOT NEED TO REGISTER THE AX for LAV to use the dll. It only needs to find the DLL in the path and it uses it. It's as simple as that. You confused things by registering it all over the place. :)

[nevcairiel]

Here is the logic how pathes are searched, in this order: (from MSDN)


1. The directory from which the application loaded.
2. The directory where LAVAudio.ax resides
3. The system directory. (The name of this directory is System32.)
4. The 16-bit system directory. (The name of this directory is System.)
5. The Windows directory.
6. The directories that are listed in the PATH environment variable.

[Carpo]

Seems all is ok since i removed all the left overs from previous installs and tests, i have a clean image of my windows 7 drive so i may run that and start from fresh :)

[SamuriHL]

Hey Nev...TiVo files. Is there any chance that LAV Splitter could add support for them? I don't know much about the format, but, when I try to use LAV Splitter it doesn't work. In fact it crashes MPC-HC and with MC16 (it tries to force LAV Splitter at the moment) it just gives me garbled video no matter what decoder I use. My TiVo files contain MPEG2 video and AC3 audio and are very likely a sort of TS container. If you need a sample, let me know. Thanks!

EDIT:

Ok, I just learned a bit more about this whole crazy TiVo thing. They have a TiVo DirectShow Source Filter that MUST be used to open the TiVo file. That's because they're encrypted. Once that happens, feeding the output from TiVo DirectShow Source Filter into LAV Splitter works fine and you can go from there. It does crash MPC-HC, though, to try to use the TiVo filter. No idea on that one. Anyway, I think JR will fix MC16 given the info I gave them so that it'll use the TiVo filter to open and decrypt the file and then feed it into LAV Splitter. I think we're good. If you want any more info on this let me know.

[Carpo]

Slightly OT, when using reclock with LAV splitter+audio (and LAV CUVID) is there anything i need to change in reclock or is it best to leave it on the default settings?

[iSeries]

Hi,

I'm experiencing some odd behaviour with the latest LAV Audio and Arcsoft 1.1.0.8. LAV Audio properties say input is 6 channels but output is 7 channels? Where is this extra channel coming from?

[SamuriHL]

Check the LAV Audio properties page. There's an option to expand 6 channels to 7. Make sure it's not checked if you don't want that behavior.

[iSeries]

Hi,

That option is not checked. It seems a movie will start out correctly, 6 channels in and 6 out, but at some point during playback an extra channel is getting added to the output.

EDIT: Although there is a 7th channel being introduced during playback, it seems to be silent?

[SamuriHL]

Interesting. I'd say give Nev as much information about the situation as possible to work with and see what he or someone else says.

[iSeries]

Here's the input pin:

Filter : LAV Audio Decoder - CLSID : {E8E73B6B-4CB3-44A4-BE99-4F7BCB96E491}

- Connected to:

CLSID: {171252A0-8820-4AFE-9DF8-5C92B2D66B04}
Filter: LAV Splitter
Pin: Audio

- Connection media type:

Audio: DTS 48000Hz 6ch 1536kbps

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: Unknown GUID Name {00002001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 256000
cbFormat: 18

WAVEFORMATEX:
wFormatTag: 0x2001
nChannels: 6
nSamplesPerSec: 48000
nAvgBytesPerSec: 192000
nBlockAlign: 12
wBitsPerSample: 16
cbSize: 0 (extra bytes)

pbFormat:
0000: 01 20 06 00 80 bb 00 00 00 ee 02 00 0c 00 10 00 . ..€»...î......
0010: 00 00 ..

And here's the output pin:

Filter : LAV Audio Decoder - CLSID : {E8E73B6B-4CB3-44A4-BE99-4F7BCB96E491}

- Connected to:

CLSID: {18C16B08-6497-420E-AD14-22D21C2CEAB7}
Filter: Audio Switcher
Pin: Audio

- Connection media type:

Audio: WAVE_FORMAT_EXTENSIBLE 48000Hz 7ch 5376kbps

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 14
cbFormat: 40

WAVEFORMATEX:
wFormatTag: 0xfffe
nChannels: 7
nSamplesPerSec: 48000
nAvgBytesPerSec: 672000
nBlockAlign: 14
wBitsPerSample: 16
cbSize: 22 (extra bytes)

WAVEFORMATEXTENSIBLE:
wValidBitsPerSample: 16
dwChannelMask: 0x0000013f
SubFormat: {00000001-0000-0010-8000-00AA00389B71}

pbFormat:
0000: fe ff 07 00 80 bb 00 00 00 41 0a 00 0e 00 10 00 þÿ..€»...A......
0010: 16 00 10 00 3f 01 00 00 01 00 00 00 00 00 10 00 ....?...........
0020: 80 00 00 aa 00 38 9b 71 €..ª.8›q

- Enumerated media type 0:

Audio: WAVE_FORMAT_EXTENSIBLE 48000Hz 6ch 6912kbps

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: MEDIASUBTYPE_PCM {00000001-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 18
cbFormat: 40

WAVEFORMATEX:
wFormatTag: 0xfffe
nChannels: 6
nSamplesPerSec: 48000
nAvgBytesPerSec: 864000
nBlockAlign: 18
wBitsPerSample: 24
cbSize: 22 (extra bytes)

WAVEFORMATEXTENSIBLE:
wValidBitsPerSample: 24
dwChannelMask: 0x0000003f
SubFormat: {00000001-0000-0010-8000-00AA00389B71}

pbFormat:
0000: fe ff 06 00 80 bb 00 00 00 2f 0d 00 12 00 18 00 þÿ..€».../......
0010: 16 00 18 00 3f 00 00 00 01 00 00 00 00 00 10 00 ....?...........
0020: 80 00 00 aa 00 38 9b 71 €..ª.8›q

[Andy o]

Check the LAV Audio properties page. There's an option to expand 6 channels to 7. Make sure it's not checked if you don't want that behavior.

That's for the LAV decoder, not the Arcsoft one. Arcsoft will expand 6.1 into 7.1, duplicating the back channel, same as it does with TMT.

[SamuriHL]

That's for the LAV decoder, not the Arcsoft one. Arcsoft will expand 6.1 into 7.1, duplicating the back channel, same as it does with TMT.

Oh, right, you're right. :stupid: This is what happens when I make replies when I'm not feeling well. :D

[Andy o]

Here's the input pin:
q

Is this a 6.0 stream? Nev has said that 1.1.0.8 has a problem with 6.0 streams (6.1 is good though). Use 1.1.0.0 (from TMT2) and the dlls he mentioned one or two pages back and see if that helps.

[nevcairiel]

That's for the LAV decoder, not the Arcsoft one. Arcsoft will expand 6.1 into 7.1, duplicating the back channel, same as it does with TMT.

Thats not true anymore. If you don't want that to happen (iow you don't tick Expand 6.1 to 7.1), it won't do that anymore.

On the original issue, for some reason it introduced a back center channel. Are you sure the file is actually 5.1 and not 6.0 ?

[Jasch]

So far new Version is working great, seeking VC1, x264 MKV HD content works like a charm.
Only downside, i still getting some Problems with Media FPS on some Files.
Lav reporting 23FPS should be 25.(MPC reports 25).
I have uploaded a small sample, maybe its helping you sorting this out.

http://www.megaupload.com/?d=C9XZM47Z


THX Alex

[iSeries]

Hi,

It's definitely 5.1 - and there is definitely LFE output, and like I said playback starts out correctly - 5.1 in and 5.1 out. However at some point during playback a 7th channel is introduced, however this channel appears to be completely silent (and after more testing seems to be happening with everything I've played with a DTS-HD soundtrack). I'll try with 1.1.0.0. EDIT: Tried with 1.1.0.0 - same problem!

EDIT 2 - Solved by unticking 'Convert output to standard channel layouts'. Now correctly plays 5.1 in and 5.1 out.

ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
Muxing mode : Stream extension
Duration : 1h 31mn
Bit rate mode : Variable
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits

[nevcairiel]

EDIT 2 - Solved by unticking 'Convert output to standard channel layouts'. Now correctly plays 5.1 in and 5.1 out.


That doesn't solve the problem, you just avoid it.
You should produce a log file, run the debug build of LAV Audio that comes in the zip.

[iSeries]

Yes you are correct, avoiding a problem isn't the same as solving it :-) So it could be a problem with that particular option?

I'll get you a log file.

[blahism]

Nev,

Are you still planning to introduce Media Foundation splitting? I would love to ditch the DivX mkv splitter as soon as possible! (Required to play back on Xbox 360 media center extenders without transcoding/remuxing back to avi..)

[nevcairiel]

Planning, sure, for some distant future. I have no real need for it, and its basically a new splitter, not simply some added feature.

[CruNcher]

Also what are the real benefits of Media Foundation compared to Dshow, except that Dshow hell has been reduced for the AVG consumer i see none that makes it interesting, also applications learned to live in that problematic space encapsulating themselves from the Public Graph Building to unique closed Private Graph Building that made them very unproblematic over the last Dshow decade for Avg consumer. :)

[BloodySword]

Also what are the real benefits of Media Foundation compared to Dshow, except that Dshow hell has been reduced for the AVG consumer i see none that makes it interesting, also applications learned to live in that problematic space encapsulating themselves from the Public Graph Building to unique closed Private Graph Building that made them very unproblematic over the last Dshow decade for Avg consumer. :)

I don't understand you, could you please clarify what you want to say?

DirectShow is 32 bit only, Media Foundation is 64 bit ASAK.

[nevcairiel]

Both DirectShow and Media Foundation can be used in both 32 and 64 bit.

They are just different APIs. Media Foundation is the new and clean API, well designed and clearly meant for media playback, while DirectShow is a remnant from days long gone by, with all the cruft that comes with it.
I would love to have everything in Media Foundation, just because its so much nicer to work with, however, I don't have unlimited time, and i wouldn't even use a Media Foundation splitter at this time, because there are no players, no renderers, no video decoders, no nothing.
So, i rather spent my time improving the DirectShow things, because those i actually use.

Maybe some day we can make the switch to Media Foundation. Once other developers get motivated to switch, i'll be among the first to write Media Foundation things, until then .. DirectShow works just fine.

[SamuriHL]

Yea, that's the real trick right there. You need to get a band of developers together that are willing to all work on the respective parts to make it all work. Splitter, video/audio decoder, renderer (with sub support built in), and finally a player to use it all. The current code out there would not be reusable for MF stuff, so, you're looking at creating everything from scratch. And then you'd have people b*tching that it doesn't work with their 10 year old OS and 15 year old hardware, etc. :) But then you could get rid of the hack-nature of current DShow implementations. So, it would be nice, but, as Nev said, time constraints of supporting existing stuff makes this venture EXTREMELY unlikely in the near term.

[blahism]

Planning, sure, for some distant future. I have no real need for it, and its basically a new splitter, not simply some added feature.

Cool, i'd be happy to help if there is anything you need to get it kick started :)

[blahism]

Yea, that's the real trick right there. You need to get a band of developers together that are willing to all work on the respective parts to make it all work. Splitter, video/audio decoder, renderer (with sub support built in), and finally a player to use it all. The current code out there would not be reusable for MF stuff, so, you're looking at creating everything from scratch. And then you'd have people b*tching that it doesn't work with their 10 year old OS and 15 year old hardware, etc. :) But then you could get rid of the hack-nature of current DShow implementations. So, it would be nice, but, as Nev said, time constraints of supporting existing stuff makes this venture EXTREMELY unlikely in the near term.

I don't think its THAT un-portable. The beauty of starting from scratch is to simply not bring over bad habits, but for the most part they didn't make it impossible to migrate from Directshow to media foundation :)

MSDN Article: http://msdn.microsoft.com/en-us/library/Aa468614

Is there an appropriate place here or elsewhere to start a brainstorming session on this?

I was at a windows 8 demo and directshow wasn't spoken with any word, i doubt its gone but even on released videos published to youtube they're even using Media Foundation in marketing materials now :)

[andyvt]

Yea, that's the real trick right there. You need to get a band of developers together that are willing to all work on the respective parts to make it all work. Splitter, video/audio decoder, renderer (with sub support built in), and finally a player to use it all. The current code out there would not be reusable for MF stuff, so, you're looking at creating everything from scratch. And then you'd have people b*tching that it doesn't work with their 10 year old OS and 15 year old hardware, etc. :) But then you could get rid of the hack-nature of current DShow implementations. So, it would be nice, but, as Nev said, time constraints of supporting existing stuff makes this venture EXTREMELY unlikely in the near term.

I've thought about writing a managed MF based player; there are some really interesting features that are available OOTB. Unfortunately, no useful splitters or decoders make it academic :)

[SamuriHL]

I've thought about writing a managed MF based player; there are some really interesting features that are available OOTB. Unfortunately, no useful splitters or decoders make it academic :)

Chicken....meet egg. :D

[nevcairiel]

I wrote a pretty basic MF player to test the architecture, it was really an academic exercise by design, but i also noticed that it won't progress any further, because as soon as playback started it was obvious how crappy the default EVR on MF is, and without a replacement video renderer, nothing is going anywhere.

@blahism:
Very few parts of the logic can be re-used, the main parts of my filters are interaction with DirectShow, as all the heavy lifting is done by ffmpeg. Sure, it would be way faster then writing a new filter completely from scratch, but its still a new filter, which will require more work then what its currently worth.

[SamuriHL]

I don't think its THAT un-portable. The beauty of starting from scratch is to simply not bring over bad habits, but for the most part they didn't make it impossible to migrate from Directshow to media foundation :)

MSDN Article: http://msdn.microsoft.com/en-us/library/Aa468614

Is there an appropriate place here or elsewhere to start a brainstorming session on this?

I was at a windows 8 demo and directshow wasn't spoken with any word, i doubt its gone but even on released videos published to youtube they're even using Media Foundation in marketing materials now :)

I for one will be adopting W8 when it's in beta, so, if it doesn't have DShow I'll be screwed and surprised. Interesting article. I wonder what the real effort would be to migrate LAV Splitter, LAV Audio, LAV Video (hehehe, ya like how I snuck that in? :D), and madVR to MFC. Even if it's just a straight migration instead of a complete rewrite it'd be a beginning. Things could then be improved upon over time. I'd be all for it, but, I doubt you'll get the devs to agree to do this. :)

[SamuriHL]

I wrote a pretty basic MF player to test the architecture, it was really an academic exercise by design, but i also noticed that it won't progress any further, because as soon as playback started it was obvious how crappy the default EVR on MF is, and without a replacement video renderer, nothing is going anywhere.

@blahism:
Very few parts of the logic can be re-used, the main parts of my filters are interaction with DirectShow, as all the heavy lifting is done by ffmpeg. Sure, it would be way faster then writing a new filter completely from scratch, but its still a new filter, which will require more work then what its currently worth.

Darn. That's no fun. EVENTUALLY it'll have to be done. I mean, when XP is 20, MAYBE people will FINALLY be able to let go of the damn thing. :D

[madshi]

DirectShow will not be gone. Microsoft has a habit of maintaining compatability as much as they possibly can.

[nevcairiel]

The main problem with "migrating" to MF is that you cannot just abandon DirectShow, or many people will come crying. So you have to maintain both, for a very long time. Its just not worth it right now. I wish it was, but it just isn't.

[SamuriHL]

If there's no incentive to move to MF, then we're stuck with DShow for years to come. There's only one MF player I know of and if it weren't for it's crappy UI it'd be a really good player. And that's WinDVD 2010. Of course, they chose to use MF so they could deal with content protection without having to write it themselves like Cyberlink and ArcSoft chose to do. In the end, though, even their content protection is better than the other 2 as it's in reality a bit more lenient. Anyway, if we're stuck with DShow, then, I guess we can forget about MF completely. As I said, when XP is 20 we can revisit the topic of maybe moving on. God I hate backwards compatibility sometimes. It truly holds back innovation and real progress.

[jimwhite]

If you had a voice in fixing the "Crappy UI" in WinDVD 11 what would you say?

[blahism]

If you had a voice in fixing the "Crappy UI" in WinDVD 11 what would you say?

This little birdy wants to chirp chirp chirp about a nice Windows Media Center interface for WinDVD :) The price for windvd is nice, but TMT5 integrates in MCE better

I typically play full screen with a remote so i want the gui out of the way

[BloodySword]

Both DirectShow and Media Foundation can be used in both 32 and 64 bit.

They are just different APIs. Media Foundation is the new and clean API, well designed and clearly meant for media playback, while DirectShow is a remnant from days long gone by, with all the cruft that comes with it.
I would love to have everything in Media Foundation, just because its so much nicer to work with, however, I don't have unlimited time, and i wouldn't even use a Media Foundation splitter at this time, because there are no players, no renderers, no video decoders, no nothing.
So, i rather spent my time improving the DirectShow things, because those i actually use.

Maybe some day we can make the switch to Media Foundation. Once other developers get motivated to switch, i'll be among the first to write Media Foundation things, until then .. DirectShow works just fine.

Interesting.

I wonder why even WinTV7 uses DirectShow instead of Media Foundation. The Application is written in .NET. No problem to detect windows version and use DirectShow in XP and Media Foundation in 7.

DirectShow works like a charme there, the graph builder they implemented is very good and stable. But with ffdshow audio decoder I sometimes have no sound (the decoder does not get any data) and with LAV audio "Slaving" is active, wich leads to buffer underruns VERY often.

TV source is DVB-T.

[nevcairiel]

I wonder why even WinTV7 uses DirectShow instead of Media Foundation. The Application is written in .NET. No problem to detect windows version and use DirectShow in XP and Media Foundation in 7.

It would be double effort for no real-world gain, so why bother.

[BloodySword]

But why does LAV audio jumps into slaving mode. What is slaving? It displays a werid samlingrate and some other strange info in the renderer.

[Mark_A_W]

Hi,

It's definitely 5.1 - and there is definitely LFE output, and like I said playback starts out correctly - 5.1 in and 5.1 out. However at some point during playback a 7th channel is introduced, however this channel appears to be completely silent (and after more testing seems to be happening with everything I've played with a DTS-HD soundtrack). I'll try with 1.1.0.0. EDIT: Tried with 1.1.0.0 - same problem!

EDIT 2 - Solved by unticking 'Convert output to standard channel layouts'. Now correctly plays 5.1 in and 5.1 out.

ID : 4352 (0x1100)
Menu ID : 1 (0x1)
Format : DTS
Format/Info : Digital Theater Systems
Format profile : MA
Muxing mode : Stream extension
Duration : 1h 31mn
Bit rate mode : Variable
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits

I can confirm this, with a 5.1 DTS-MA disc.

Not only do I get 7 channels, instead of 5.1, but the left surround is completely silent, in fact I think the left surround is mixed with the right (a plane went left off the screen and the sound came out the right surround). I only have a 7.1 DTS-MA test tone...grrr...

It's the Convert to Standard Channel Layouts that does it, there is definitely something fishy going on.

[nevcairiel]

I still couldn't confirm the problem. Do you have any samples that directly show the issue? Or at least some log files?
Or just, which Blu-ray does it happen on, maybe i have it.

Or just what exactly the LAV Audio status page says for the two channel fields (the full fields, not just the channel count)

You arent exactly helping the issue. :p

Edit:
Nvm i found a sample to reproduce it, fixing.

[nevcairiel]

Ok, try this version. It should fix the DTS-HD MA thing, i hope.

http://files.1f0.de/lavf/LAVFilters-0.28-6-g151e36c.zip

[noee]

ha, never noticed it before (only have 2 DTS-HD MA sources), but yes, it appears to fixed.

[Mark_A_W]

Ok, try this version. It should fix the DTS-HD MA thing, i hope.

http://files.1f0.de/lavf/LAVFilters-0.28-6-g151e36c.zip


Yep, that's fixed it. Channels are both 6 / 0x3f now, before the output was different.

The service around here is terrible...
That fix took almost an hour :eek:
Another 9 minutes and you'd have had to deliver it for free ;)


Thanks nev!

(I did have a couple of strange audio sync issues in the movie tonight - it mysteriously glitched and went out of sync by few seconds. It could be something else in my convoluted playback chain, I'll keep an ear on it....if it happens again I'll hunt down how to generate a log, I missed that bit.)

[CruNcher]

@nev
here is the output of mplayer with that Portuguese .ts sample :)


MPlayer Sherpya-SVN-r33488-4.2.5 (C) 2000-2011 MPlayer Team
Setting process priority: abovenormal

163 audio & 368 video codecs

Playing C:\SAMPLE~3.TS.
TS file format detected.
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
VIDEO H264(pid=273) AUDIO AAC LATM(pid=274) NO SUBS (yet)! PROGRAM N. 59200
ID_VIDEO_ID=273
ID_AUDIO_ID=274
ID_VIDEO_ID=529
ID_AUDIO_ID=275
PROGRAM_ID=59200 (0xE740), PMT_PID: 257(0x101)
PROGRAM_ID=59224 (0xE758), PMT_PID: 8136(0x1FC8)
ID_AUDIO_ID=530
FPS seems to be: 29.970030
Load subtitles in C:\
ID_FILENAME=C:\SAMPLE~3.TS
ID_DEMUXER=mpegts
ID_VIDEO_FORMAT=0x10000005
ID_VIDEO_BITRATE=0
ID_VIDEO_WIDTH=1920
ID_VIDEO_HEIGHT=1088
ID_VIDEO_FPS=29.970
ID_VIDEO_ASPECT=0.0000
ID_AUDIO_FORMAT=MP4L
ID_AUDIO_BITRATE=0
ID_AUDIO_RATE=0
ID_AUDIO_NCH=0
ID_START_TIME=169.47
ID_LENGTH=0.00
ID_SEEKABLE=1
ID_CHAPTERS=0
Opening video filter: [ass auto=1]
[ass] auto-open
Opening video filter: [screenshot]
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264)
==========================================================================
ID_VIDEO_CODEC=ffh264
==========================================================================
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
AUDIO: 48000 Hz, 2 ch, s16le, 0.0 kbit/0.00% (ratio: 0->192000)
ID_AUDIO_BITRATE=0
ID_AUDIO_RATE=48000
ID_AUDIO_NCH=2
Selected audio codec: [fflatm] afm: ffmpeg (FFmpeg AAC in LATM)
==========================================================================
AO: [dsound] 48000Hz 2ch s16le (2 bytes per sample)
ID_AUDIO_CODEC=fflatm
Starting playback...


no issues, but yeah pretty much every other demuxer has issues with this (including lav splitter + lav audio decoder no sound) ;)

Another stream which shows the audiodamaged.ts issue of fast playing back ghostly (DivX AAC Decoder) but this time audio seems to be not damaged @ all ? (this one also works with mplayer)

Upload in progress...

http://www.megaupload.com/?d=BPADC18B

[pirlouy]

My question was missed some days ago:
I'd like to bitstream DTS-HD, but not DTS. Is it possible ? Will it be possible ? Or it is not recommended ?

Why do I ask this ? I prefer to send PCM rather than non-decoded signals to my preamp. And as you know, it's possible to decode DTS by software, whereas DTS-HD, it's quite impossible (TMT hard manipulation which does not work well).

[nevcairiel]

Its not possible, and I have no concrete plans to make it possible. I'll let someone else explain that using the TMT decoder is actually super easy.

[SamuriHL]

You mean the "drop the dtsdecoderdll.dll from a tmt installation into the LAV filter folder" instructions? Right, um, that's very difficult. :D You know, with the copying, and pasting and the automagic way in which LAV Audio then just decides to use it without any configuration or effort on the user's part at all. What a PITA! :D

[robpdotcom]

"drop the dtsdecoderdll.dll from a tmt installation into the LAV filter folder"


Maybe this should be added to the first post of this thread? I have a feeling it's just going to come up over and over and......

[GTPVHD]

http://www.youtube.com/watch?v=LSFdgwZZVOE

Windows 8 running on quad-core Nvidia Tegra Kal-El SoC decoding HD video using Media Foundation API. It would be nice if LAV Splitter has MF support on both x86 and ARM since Kal-El has a decoder that can decode High Profile 1440p. WMP is after all the first player to ever support MF, long before WinDVD 2010.

http://www.anandtech.com/show/4181/nvidias-project-kalel-quadcore-a9s-coming-to-smartphonestablets-this-year/3

NVIDIA's video decoder gets an upgrade in Kal-El to support H.264 at 40Mbps sustained (60Mbps peak) at a resolution of 2560 x 1440. This meets the bandwidth requirements for full Blu-ray disc playback. NVIDIA didn't just make the claim however, it showed us a 50Mbps 1440p H.264 stream decoded and output to two screens simultaneously: a 2560 x 1600 30" desktop PC monitor and a 1366 x 768 tablet display.

[CruNcher]

http://www.youtube.com/watch?v=LSFdgwZZVOE

Windows 8 running on quad-core Nvidia Tegra Kal-El SoC decoding HD video using Media Foundation API. It would be nice if LAV Splitter has MF support on both x86 and ARM since Kal-El has a decoder that can decode High Profile 1440p. WMP is after all the first player to ever support MF, long before WinDVD 2010.

http://www.anandtech.com/show/4181/nvidias-project-kalel-quadcore-a9s-coming-to-smartphonestablets-this-year/3

Hehe so maybe the next Boxxe then, though Annandtech was also very friendly with Tegra ;) and later the disaster happened, though Nvidia can't risk such a epic fail again vs Intel ;)

[nevcairiel]

This doesn't say that it won't support DirectShow.

Anyhow, how often do i have to repeat it:
You cannot just "add Media Foundation support"! On the splitters side, Media Foundation is fundamentally different to DirectShow, so its a whole new component to be developed!
The same is more or less true for decoders, btw.

There is DirectShow, and there is Media Foundation. There is no overlap! (except DMO decoders, those can be used by both)

[nevcairiel]

Maybe this should be added to the first post of this thread? I have a feeling it's just going to come up over and over and......

I'm setting up some actual page on my webserver right now, which will get some sort of mini-manual for this.

LAV Filters and LAV CUVID have been gaining alot of traction the last few weeks, and its only going to get "worse", so i wanted some resources other then a forum thread for people to read.

You can checkout my WIP on http://1f0.de/

If someone wants to contribute some content, or has some ideas what i should put up there, you're welcome to it.
I kinda need a new banner for the page, stupid default thing. :D

I need to post some more recent News, though. :p

[Stomp]

Hello,

I had a problem with LAVFilters 0.28 on Windows XP SP3 32-bit. It seems that LAVFilters breaks playback of MPEG-1 (mpg) files with the MCI API. When I tried to play a mpg file with mplay32.exe the player just hanged. I don't need the player itself, but MCI is used by other applications and that's important. After uninstalling LAVFilters mplay32 could play the mpg file.

[nevcairiel]

MCI has been deprecated for a decade, if you still require such ancient stuff to work, you should probably not mix it with modern software.

Anyhow, if you want anything fixed, you need to provide more then "it doesnt work". A sample file maybe, and if that file works in WMP, and stuff like that.

[Stomp]

I have tried many mpg files and mplay32 always hanged. It's the playback of mpg files that is broken. All of the files work Windows Media Player. And they work in mplay32 when LAVFilters is uninstalled.

[pirlouy]

Its not possible, and I have no concrete plans to make it possible. I'll let someone else explain that using the TMT decoder is actually super easy.
Ok.
I've tried the Arcsoft's dll solution (1.1.0.7), but it did not work (output 48kHz, 16 bits - 8 channels yet ) and if it worked, I would need Reclock for WASAPI, as I don't output 96kHz, 24 bits when using DirectSound.
So I prefer to keep the bitstream solution which automatically triggers WASAPI.

For DTS, I would have preferred Directshow. But I understand it's not important to you, and I don't ask you to implement it. It's just my point why I wanted DTS-HD bitstream only.

Keep up with your very good work.

[Andy o]

So what's the reason to prefer decoding DTS, if you like bitstreaming DTS-HD?

[SamuriHL]

Yea, I'm confused, too. If you're converting DTS to PCM, yet bitstreaming DTS-HD MA, why not just bitstream both?

[pirlouy]

DTS is 16 bits and <48kHz, I can use directsound volume (windows volume).
Yes, I've settled Windows to through out 48kHz and 16 bits, it avoids problems with my preamp (AV need some delay to sync/catch 24 bits or 92 kHz).

So I let all 24bits or 92kHz stuff (Blu-Ray or DVD-Audio in general) to WASAPI.

But of course, I can bitstream DTS to my preamp... That's just that I don't have Windows volume control...

[SamuriHL]

Fair enough. But if you decode DTS-HD MA with the ArcSoft decoder using Nev's LAV Audio, it should also get sent out through Windows Audio. Meaning it will downsample to 48/16 and whatever speaker config you have set. That's not a decoder issue. That's Windows Audio doing that.

[Andy o]

Is there a reason you don't wanna use ReClock? It would solve your issues, plus give you perfectly matched video playback. Even when using WASAPI exclusive, ReClock lets you control volume with the player's volume (not the Windows mixer, but it's practically the same). Of course this only works when you decode and not bitstream, but you should look into why dtsdecoderdll.dll is not working right for you anyway. With MPC-HC it's easy to map volume to your remote, or your keyboard.

BTW, doesn't ffdshow let you decode DTS and bitstream DTS-HD? I think so.

[pirlouy]

Indeed. That's why I said, if I'd manage to decode DTS-HD properly, I'd need WASAPI (what is the interest of using DTS-HD if downsample to 48kHz 16 bits ?), and AFAIK I'll have to use Reclock to have WASAPI, and I don't want/need to use Reclock.

Like ffdshow manages to do it, I though it was possible, but maybe it's a dirty hack and nevcairiel does not want (I respect that)...

[pirlouy]

Is there a reason you don't wanna use ReClock?
I prefer to avoid what can be avoid. :-)
My TV and GPU are 23,976 ready (madVR report 1 frame repeat all 45 minutes, and I never noticed anything when watching video) and I'm not bothered by 24 -> 23,976 audio speed down.


Even when using WASAPI exclusive, ReClock lets you control volume with the player's volume (not the Windows mixer, but it's practically the same).
Ah ? You can use MPC-HC volume in these condition ? Apart if Reclock hooks MPC volume, I think it's not possible; by default, MPC Volume is ignored when WASAPI...

For dtsdecoderdll.dll, I'll test again in some weeks, after reading other attemps. :P

[SamuriHL]

Also, if you don't want ReClock's video abilities, you can shut that off and just use it for the WASAPI exclusive mode. Basically, at that point you just set MPC-HC to use ReClock as the audio renderer and configure it only to use WASAPI exclusive and disable the video functionality. Works quite well, actually.

[Andy o]

I prefer to avoid what can be avoid. :-)
My TV and GPU are 23,976 ready (madVR report 1 frame repeat all 45 minutes, and I never noticed anything when watching video) and I'm not bothered by 24 -> 23,976 audio speed down.
It's not like ReClock is superfluous. It provides features that you need, and there are practically no negatives, so why not use it?

Ah ? You can use MPC-HC volume in these condition ? Apart if Reclock hooks MPC volume, I think it's not possible; by default, MPC Volume is ignored when WASAPI...
Yes, it can do it, and IMO it's a pretty nice feature.

For dtsdecoderdll.dll, I'll test again in some weeks, after reading other attemps. :P
Others' attempts are pretty well documented already, and I'm pretty sure it just works for most.

[Plutotype]

Hi Nevcairel,

I use 0.28 LAVaudio and have this option "Output only "standard" channel layouts" enabled. Today, at only one movie, I had to disable it, because if enabled, the center channel was (probably) muted or replaced with silence.
It was 6 channel DTS-HD MA, outputted via Realtek chip to 5.1 analog output and per cinch to my Logitech 5.1 set. Sample:
http://www.mediafire.com/?669vteuam2xnms0
Thanks for clarification
Pluto

[robpdotcom]

Also, if you don't want ReClock's video abilities, you can shut that off and just use it for the WASAPI exclusive mode. Basically, at that point you just set MPC-HC to use ReClock as the audio renderer and configure it only to use WASAPI exclusive and disable the video functionality. Works quite well, actually.

That's what I do as well. Madvr reports 1 frame drops every 1.XX days, so I don't bother with Reclock's other features.

[SamuriHL]

That's what I do as well. Madvr reports 1 frame drops every 1.XX days, so I don't bother with Reclock's other features.

Yup. madVR keeps the peace quite nicely so I only use ReClock as an audio renderer, as well.

[robpdotcom]

What about the MPC-HC Audio Renderer? Doesn't it have WASAPI mode?

A simple WASAPI mode audio renderer would be nice. Something that just sends everything through HDMI as is, whether bitstreaming or decoding......



LAVAudio Renderer? madAR? :D

[SamuriHL]

Not to my knowledge, but, yea, a nice WASAPI audio renderer would be sweet. J River's MC16 has one built in. Works very well.

[ney2x]

LAVAudio Renderer? madAR? :D

Sounds good! :D

[Carpo]

How do you go about just using reclock for audio?, just untick everything under video?

[groen]

Lav filters works perfectly as a splitter and the audio filter works great but I can not get the included DTS to work with ffmpeg. I tried several combinations of DTS codecs, arcsoft, ac3filter. So i have had to disable the DTS completely on the lav audio filter. Please tell me if you know what I have done incorrect? This is what I currently have set.

mpc-hc 1.5.1.2903

http://img848.imageshack.us/img848/3958/mpchcaudioswitcher.th.jpg (http://img848.imageshack.us/i/mpchcaudioswitcher.jpg/)
http://img834.imageshack.us/img834/1589/mpchcinternalfilters.th.jpg (http://img834.imageshack.us/i/mpchcinternalfilters.jpg/)
http://img854.imageshack.us/img854/254/mpchcrenderoutput.th.jpg (http://img854.imageshack.us/i/mpchcrenderoutput.jpg/)

Then i have ffdshow-tryouts for video and disable the spdif passthrough on the audio settings output page. Why does the ffdshow-tryouts dts and dts hd from ffdshow not work with lav filter?

http://img215.imageshack.us/img215/417/ffdshowaudios.th.jpg (http://img215.imageshack.us/i/ffdshowaudios.jpg/)
http://img233.imageshack.us/img233/9756/ffdshowvideo.th.jpg (http://img233.imageshack.us/i/ffdshowvideo.jpg/)

Then on the lav audio page I disable DTS passthrough. So to get DTS working but not dts hd, I used the microsoft dts passthrough that works automatically in windows 7 or you can install ac3filter. You suggest using the arcsoft dts codecs but I tried those codecs and had the same results when bitstreaming DTS.

The problem when bitstreaming DTS with lav audio filter+ffmpeg is that i get no sound and the video slows down to slow motion, but the AV receiver does come up with DTS. Occasionally I might hear a slight sound for a split second.

[Mercury_22]

LavSplitter wasn't supposed to respect the Audio "Preferred Language" priority in index.bdmv / m2ts files too ? Cause for me it's working only in mkvs

[nevcairiel]

It doesn't care what type of file it is, as long as there is language info, and for me it works fine on Blu-rays.

[Mercury_22]

It doesn't care what type of file it is, as long as there is language info, and for me it works fine on Blu-rays.
Should I upload as sample the mpls + m2ts files or the mpls + index.bdmv +m2ts?

P.S. Do you have e.g. THE FIFTH ELEMENT REMASTERED ?

[nevcairiel]

If the stream selector shows languages, then it'll also use those for the stream choice. If it doesn't show languages, then the disc just doesn't have any.

I have one fifth element BD, not sure if its the same.

[Mercury_22]

If the stream selector shows languages, then it'll also use those for the stream choice. If it doesn't show languages, then the disc just doesn't have any.

I have one fifth element BD, not sure if its the same.

Tested with multiple BD and the stream selector shows the languages but it prefers the default language instead of the one in the "Preferred Language"

P.S. Your fifth element has French beside English audio?

But if it's working for you then it must be something with my system, but what ?

Edit I've reset the settings and unregistered the splitter multiple times

[nevcairiel]

I just tested on one BD (Iron Man 2 german release), no config selects the first audio stream, which is german, adding "eng" to the preferred list of languages selects the second, english, track.

You did put the languages as their 3 letter code, comma seperated - right? :p

[Mercury_22]

I just tested on one BD (Iron Man 2 german release), no config selects the first audio stream, which is german, adding "eng" to the preferred list of languages selects the second, english, track.

You did put the languages as their 3 letter code, comma seperated - right? :p

:)Yes!
Is fre for french no?

EDIT one idea: could be that the "better" format e.g. dts-hd ma has priority over e.g. ac3 language ?

[SamuriHL]

Quantum of Solace chose the chinese subtitles in MC16. I have eng/eng set, so, that was a bit....odd. I would have thought it would have chosen the forced sub track by default so I'm thinking it may be a bug in MC16. Very strange. :)

[nevcairiel]

MC16 overrides all stream decisions, at least for subtitles, maybe audio too.

[nevcairiel]

:)Yes!
Is fre for french no?

EDIT one idea cold be that the "better" format e.g. dts-ma has priority e.g. ac3 over language ?

French can be both "fre" or "fra", i've seen both used in files, not sure what BDs use.

The language has the highest selection priority, quality is only a factor after that.

[SamuriHL]

MC16 overrides all stream decisions, at least for subtitles, maybe audio too.

Alright. I'll file that with them a bit later.

[nevcairiel]

French can be both "fre" or "fra", i've seen both used in files, not sure what BDs use.

The language has the highest selection priority, quality is only a factor after that.

I just checked some BD, and french is "fra" there. I changed it so that the language code is always shown in addition to the full word.

[Mercury_22]

I just checked some BD, and french is "fra" there. I changed it so that the language code is always shown in addition to the full word.
Working with "fra" :thanks:but I've just discover one BD that it's NOT working with ger (see the sample with DTS EXPRESS stream)

[nevcairiel]

German also has two variants, "ger" and "deu" .. go figure. :p
Like i said, next version will show the language code on the stream chooser as well as the full name, to identify such problems.

[ney2x]

Is there a way to show mkv thumbnails like Haali? I had tried clsid's reg but nothing, I ended up installing divx codecs :(

[Carpo]

would be great if xbmc could use your splitter and audio for its player :)

[nevcairiel]

Is there a way to show mkv thumbnails like Haali? I had tried clsid's reg but nothing, I ended up installing divx codecs :(

Creating thumbnails is some wicked magic, and if using the WMP component isn't working for you, well, not much i can do.

Its not the splitters job to produce thumbnails, there is some extra shell component that does that.


would be great if xbmc could use your splitter and audio for its player :)
Since XBMC does not use DirectShow, no chance. :p
But XBMC uses ffmpeg anyway, so the end result should be pretty similar.

[SamuriHL]

would be great if xbmc could use your splitter and audio for its player :)

Would be great if xbmc were DShow compatible. :)

[Carpo]

I know, they dont use it so it can be used on all platforms, would be good though because Navs filters give me sweet play back :) , plus i can get it to use MPC-HC which can use those filters, so all is good :)

[robpdotcom]

There was a DirectShow player for XBMC a while back. It was actually quite nice imo, and there were lots of ways you could configure the way it used filters. Too bad development stalled.

[Sebastiii]

Hey :) Mediaportal can use it (lav) i have to see why sometimes crash happen but it works really nice + BDHandler (to manage BD Title Selection).

[Boltron]

I use XBMC exclusively but have configured it to use MPC-HC as an external player (exclusive mode). I get the awesome media management of XBMC with Aeon skin and the best playback with MPC-HC/LAV/madVR. There are a few gotchas but nothing serious.

I used to use XBMC DSplayer (for DirectShow) and have tried the new DScodec version but have settled on the external player route.

[Mercury_22]

There is a discrepancy between LAVSplitter and LAVAudio for 7.1 DTS-HD MA LAVSplitter reports the files as 48000Hz 5.1(incorrect) and LAVAudio as 96000Hz 8 channels (correct)
The files are playing correct (8 channels) so is no big deal :) but maybe you can fix this in the next version too

Do you need a sample?

[nevcairiel]

That won't be fixed soon, unless ffmpeg supports DTS-HD one day.

LAV Audio uses the ArcSoft decoder to figure this out, LAV Splitter does not have this ability, and only sees what ffmpeg tells it (the DTS core data)

[Blight]

nevcairiel:
I read your post on the ZP forum regarding your suggested stream switching interface.
If I remember correctly, your suggestion works in a similar way to the 'IAMGraphBuilderCallback' interface, so it shouldn't be too much trouble just to implement the initial interface.

[nevcairiel]

Its somewhat similar, indeed.
Except that for a filter that was created its called for the output Pins of the splitter, so the player can do what needs to be done (if it wants to) - and of course only called on stream changes, not on the initial graph build.

I'll publish the interface soon, sadly my plans got delayed by some other work that had to be done.

BTW, while you're here, i don't know if you saw this post: http://forum.inmatrix.com/index.php?showtopic=13053&view=findpost&p=45954
Would be great if that gets fixed, so i can enable ITrackInfo again - or i'll have to detect if its ZP loading LAV Splitter and disable that interface in that case.

[hoborg]

nevcairiel:
Still no plans to make LAVF audio decoder working with MS DVD navigator? :/

[nevcairiel]

Plans? Sure.
Concrete ETA? Not yet.

[hoborg]

Plans? Sure.
Concrete ETA? Not yet.

:thanks:

[nevcairiel]

I put it on the todo list on the second post, i'll try to get it done "soon". Will have to investigate what exactly is required for that, first.

[hoborg]

I put it on the todo list on the second post, i'll try to get it done "soon". Will have to investigate what exactly is required for that, first.

2x:thanks:

If LAVF audio decoder will be working on DVD playback, i (and many other HTPC users) can say good bay to FFDshow audio decoder :)

[Sebastiii]

nevcairiel:
I read your post on the ZP forum regarding your suggested stream switching interface.
If I remember correctly, your suggestion works in a similar way to the 'IAMGraphBuilderCallback' interface, so it shouldn't be too much trouble just to implement the initial interface.

BTW, while you're here, i don't know if you saw this post: http://forum.inmatrix.com/index.php?...ndpost&p=45954
Would be great if that gets fixed, so i can enable ITrackInfo again - or i'll have to detect if its ZP loading LAV Splitter and disable that interface in that case.

Blight, I will ask you the same question that Nevcariel about ITrackinfo :)

Hope you can fix it on ZP :)

[Thunderbolt8]

can subtitles also be automatically loaded as well when you remux a movie to .mkv and have the .sup file with the same filename in the same folder? or does that only work within the .m2ts structure? if so, would it be possible to implement this, because properly OCRing subs can be a pain in the ass and sometimes easily take up to an hour, depending on the kind of type (e.g. those 3 lined sub files which have the lines for each speaker located near him on screen and after OCRing this then gets all mashed to the middle of the screen without indicating which speaker said what line).

[nevcairiel]

Loading of external subs is not the splitters job, ffdshow can do this for you, or MPC-HC itself.

[Thunderbolt8]

but do they exactly do it in the way the subs are supposed to appear on screen like when playing the BD in a standalone player? thats the thing, otherwise I wont get round of OCRing.

but anyway, guess that subs loading is due to mpc-hc then and not due to LAVfilters which I thought would be the case.

[nevcairiel]

The only reason to OCR is if you want to change the font or increase the quality by re-rendering from a high quality font. The PGS subtitles on Blu-rays (and MKVs ripped from Blu-rays) work no matter what.

[yesgrey]

Feature request:
Would it be possible to add a convert to stereo option to LAV Audio?

Yesterday I installed LAV Splitter on a friend's computer and needed to install ffdshow so that the 5.1 audio would be mixed to 2.0. He only has 2 speakers. With LAV Audio he didn't get any speech (center channel)...

[SamuriHL]

Nev has a mixer on his todo list but there's no ETA. I use ffdshow's audio processor for now just to do the mixing.

[nevcairiel]

Such an option is planned. I'm not sure yet how extensive, maybe in several steps, first a downmix to stereo option, and possibly upmix to 5.1, and maybe some day later a full matrix for custom thingys.

[yesgrey]

OK, thanks. No pressure. :)

[Andy o]

Doesn't Windows just mix the audio? I just checked on my laptop, and 5.1 content comes out the 2 speakers just fine, using LAV filters.

[ontherocks]

Facing some problems with DD-THD bitstreaming. The receiver seems to keep on looking for audio stream. HDMI light and DD light keeps on blinking with no audio. DTS-HD/DTS-HDMA works fine.

DD-THD files obtained from http://www.demo-world.eu/trailers/high-definition-trailers.php
Receiver HT-R380

[RealSnoopyDog]

Hi nevcairiel, just thank you for this software. Works like a charm in DVBViewer. I use it for Blu-Ray and mkv playback. It took me a while to find out that i need reclock to get rid of choppy playback with some material ;)

[Thunderbolt8]

regarding subs, what is the default flag for at the muxing stage in mkvmerge? does it only refer to the default subtitle track or also defines whether the subs should be automatically activated or disabled at the start?

or is the latter purely a matter of the splitter settings (LAVsplitter here)?

[yesgrey]

Doesn't Windows just mix the audio? I just checked on my laptop, and 5.1 content comes out the 2 speakers just fine, using LAV filters.
What's your OS?
I think i worked like that on XP, but now with 7 it doesn't... at least with my friend's computer onboard sound card...

[Superb]

What's your OS?
I think i worked like that on XP, but now with 7 it doesn't... at least with my friend's computer onboard sound card...Windows' mixer works just fine in 7. Just make sure you use proper audio drivers and set the Sound settings correctly (Control Panel-> Sound -> Choose device -> Configure -> pick the correct speaker setup for your system).

[Andy o]

Yeah, Windows 7 here. The mixer should downmix even before audio gets to the device's driver.

[robpdotcom]

regarding subs, what is the default flag for at the muxing stage in mkvmerge? does it only refer to the default subtitle track or also defines whether the subs should be automatically activated or disabled at the start?

or is the latter purely a matter of the splitter settings (LAVsplitter here)?

If LAVSplitter is set to load "Full Subtitles", it will load the default subtitle track, or the first track if none are flagged as default. If it is set to load "Only Forced Subtitles", it will only load subtitle tracks flagged as forced.

[Thunderbolt8]

so its not possible to make a choice whether a track shouldnt be loaded automatically right from the start at the muxing stage? its only either all tracks are loaded automatically or all forced tracks are loaded or none at all?

[robpdotcom]

If you want a subtitle track to be shown automatically, set the forced flag as "yes" in mmg. If you don't want it to be shown automatically, set it as "no".

Then, just make sure you have LAVSplitter set to only load forced subs. You can still load them manually.

[someone`]

AR bug :
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L5.1
Format settings, CABAC : Yes
Format settings, ReFrames : 8 frames
Format settings, GOP : M=3, N=8
Codec ID : V_MPEG4/ISO/AVC
Duration : 23mn 41s
Bit rate : 1 292 Kbps
Nominal bit rate : 1 627 Kbps
Width : 720 pixels
Height : 480 pixels
Display aspect ratio : 16:9
Original display aspect ratio : 3:2
Frame rate : 29.970 fps
Standard : NTSC
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Haali 16:9
LAV 3:2

[glorp]

If LAVSplitter is set to load "Full Subtitles", it will load the default subtitle track, or the first track if none are flagged as default. If it is set to load "Only Forced Subtitles", it will only load subtitle tracks flagged as forced.

I really wish there were also an option like:
"Don't load subtitles if matching language found"

and then that option would take preference over all the others. That way all subs could be turned off automatically by the splitter if a matching preferred langauge track exists but enable them if no matching language is present and then follow the other rules for sub language selection. Haali can do this by it's options that allow you to preferentially select audio,sub pairs (which also includes allowing wildcards in the matches along with the keyword "off") but I can't find a similar way to do that with LAV. Or is there a way to accomplish this with the options that exist that I haven't realized?

[Thunderbolt8]

If you want a subtitle track to be shown automatically, set the forced flag as "yes" in mmg. If you don't want it to be shown automatically, set it as "no".

Then, just make sure you have LAVSplitter set to only load forced subs. You can still load them manually.but what if there are forced subs inside that .sup track? would they be shown then? (which I wouldnt want as long as I dont want to load the track automatically in general).

what would be helpful is some kind of table of results for mkvmerge flags + LAVfilter settings.

e.g. mkvmerge default no, forced no + LAVfilter: show full titles=?
mkvmerge default no, forced yes + LAVfilter: only forced subs=?
.
.
.

and this for all combinations and options there are in mkvmerge and LAVfilter, because it looks to me, as rather noobish user in that area, theres no way to see which option seems to take priority over the other one.

[nevcairiel]

The "Default" flag has no influence IF a subtitle is chosen, it just influences WHICH. If there are two of the same language, one is default, then that one will be used.

If you enable "Show full subtitles" and have "Only with matching language" NOT checked, there will always be a substream activated, based on your language choices, or if no language matches, just the first stream.
If you enable "Show full subtitles" and have "Only with matching language" checked, there will always be a sub stream activated IF it matches one of your preferred languages, if there is no match, then there wont be subs.

The same goes for "Only show forced subs", except that it additionally ONLY works on sub streams with the forced flag, streams without that flag are never used.

[glorp]

I really wish there were also an option like:
"Don't load subtitles if matching language found"


I should have written the option as:
"Don't load subtitles if matching audio language found"

Is there a way to accomplish this short of remuxing every file to set the forced flag on subs that one wants displayed automatically while not showing any subs if a suitable audio language exists?

[robpdotcom]

Is there a way to accomplish this short of remuxing every file to set the forced flag on subs that one wants displayed automatically while not showing any subs if a suitable audio language exists?

I can't comment on the matching language part, but...

You do not need to remux to set a track as forced - just use mmg's header editor, which takes just a few seconds to do.

[nevcairiel]

I might add such an option in the future.

[ney2x]

Got this from madflac thread. A wishlist :)

Or just use LAV Audio decoder, it can decode pretty much everything.

Even though it's based on ffmpeg, which is (or was) limited to 16 bit in ffdshow for flac, I think nevcariel fixed that issue.


However, for guaranteed compatibility, it would be nice if madshi and nevcariel would access madflac in LAV Audio directly, the way he has with the Arcsoft DTS decoder.

Then we would have one complete audio decoder for everything. And stream switching, even for different audio types, should just work.

[nevcairiel]

Will not happen.

[glorp]

You do not need to remux to set a track as forced - just use mmg's header editor, which takes just a few seconds to do.

Good point that I hadn't considered but I really wanted to avoid having to force a specific sub permanently as much as avoid all the remuxing time. Thanks though.

I might add such an option in the future.

Thanks. I appreciate the consideration. The way Haali implements that one option for setting (audio,sub) pairs preferences and also allows for matches with wildcards and keywords in it gives a lot of flexibility in selecting subs on/off automatically. I find I rarely need to change flags on anything when I use that option correctly and always get correct subs w/o having choose or enable them.

[yesgrey]

Windows' mixer works just fine in 7.
Yeah, Windows 7 here. The mixer should downmix even before audio gets to the device's driver.
Are you using reclock? Maybe not...

I'm using reclock in WASAPI mode to bypass windows mixer, hence why the down mixing might not have any effect...

[Superb]

Are you using reclock? Maybe not...

I'm using reclock in WASAPI mode to bypass windows mixer, hence why the down mixing might not have any effect...Not using ReClock nor any other WASAPI-capable filter.
So the mixer in Windows 7 is working... you just choose to bypass it (=not use it). I guess you do need mixing capabilities in LAV Audio for that...

In the meantime (until such feature will be coded), one could probably be using ffdshow audio decoder to mix LAV Audio's output (enable ffdshow for uncompressed audio) and feed it to ReClock, no?

[SamuriHL]

Not using ReClock nor any other WASAPI-capable filter.
So the mixer in Windows 7 is working... you just choose to bypass it (=not use it). I guess you do need mixing capabilities in LAV Audio for that...

In the meantime (until such feature will be coded), one could probably be using ffdshow audio decoder to mix LAV Audio's output (enable ffdshow for uncompressed audio) and feed it to ReClock, no?

Yup. Just add the ffdshow audio processor to the chain. Works fine. I use it on my laptop.

[Andy o]

Are you using reclock? Maybe not...

I'm using reclock in WASAPI mode to bypass windows mixer, hence why the down mixing might not have any effect...

That might be it, although then it could be the drivers, cause the drivers are telling ReClock that they accept 5.1 channels.

But, why use WASAPI exclusive in this situation, especially when you're using ReClock?

[ontherocks]

My question was missed some days ago:
I'd like to bitstream DTS-HD, but not DTS. Is it possible ? Will it be possible ? Or it is not recommended ?

Its not possible, and I have no concrete plans to make it possible. I'll let someone else explain that using the TMT decoder is actually super easy.

When I play any DTS-HD file with MPC-HC & LAV, I am able to bitstream DTS-HD to my receiver. Is it not supposed to do so?

[nevcairiel]

You misunderstood the original question. He wanted to enable bitstreaming for DTS-HD, but NOT for DTS, but the configuration doesn't allow this. Its not about the actual bit streaming capability.

[Monamona]

For mp4, LAV splitter seems not to handle EDTS correctly like previous Haali.
Thus, movie which contains several b-frames (by x264.exe) cannot be seeked accurately or may have desynchronization.

[ney2x]

Can you give us a hint on your next release? Does it include up/down-mixing option in LAV Audio. Is there a plan for LAV Audio Renderer? Does LAV Filters have an icon :D
Honestly, I'm so excited! :X

[nevcairiel]

The next release won't include many user visible enhancements, only some playback improvements to MPEG-TS and alot of under-the-hoods things.

Honestly, development will be slowing down a bit for the next month or so, there is vacation coming up. :p

[ontherocks]

Facing some problems with DD-THD bitstreaming. The receiver seems to keep on looking for audio stream. HDMI light and DD light keeps on blinking with no audio. DTS-HD/DTS-HDMA works fine.

DD-THD files obtained from http://www.demo-world.eu/trailers/high-definition-trailers.php
Receiver HT-R380
Any ideas on this? It happens only with LAV.

[nevcairiel]

Sounds like you're using a splitter that does not split AC3 out of the TrueHD before sending it. Use LAV Splitter, and it should work.

[yesgrey]

In the meantime (until such feature will be coded), one could probably be using ffdshow audio decoder to mix LAV Audio's output
That's what I'm doing.

But, why use WASAPI exclusive in this situation, especially when you're using ReClock?
Because I don't want another resampling step in the chain...;)

[ontherocks]

Sounds like you're using a splitter that does not split AC3 out of the TrueHD before sending it. Use LAV Splitter, and it should work.

I have installed both the decoder as well as splitter.

http://img197.imageshack.us/img197/4924/capture08062011230512.png

http://img534.imageshack.us/img534/6183/capture08062011230620.png

http://img690.imageshack.us/img690/3692/capture08062011230640.png

http://img809.imageshack.us/img809/8800/capture08062011230754.png

[Mercury_22]

I have installed both the decoder as well as splitter.

http://img197.imageshack.us/img197/4924/capture08062011230512.png

http://img534.imageshack.us/img534/6183/capture08062011230620.png

http://img690.imageshack.us/img690/3692/capture08062011230640.png

http://img809.imageshack.us/img809/8800/capture08062011230754.png

Have uncheck MPC-HC's mpeg splitter cause even if you add LAV to external filters you still need to disable the internal mpeg splitter to be able to use an external (LAV) splitter ?

[e-t172]

But, why use WASAPI exclusive in this situation, especially when you're using ReClock?

Because I don't want another resampling step in the chain...;)

If the sampling rate of what you're playing is the same as the sampling rate set in the Windows Audio properties for your audio device, then no resampling will take place after Reclock.

[Andy o]

^To add a bit, you might even wanna resample with ReClock, for smooth video playback, and just set the output at the same rate as the Windows mixer. You won't have to change any settings depending on content sample rate. Nothing will be resampled by the Windows mixer.

[yesgrey]

XP's kmixer resampled even when the sample rate was the same, hence why I thought that the same might happen in Win7... If it's not, then that should be OK.

Thanks.

[n3w813]

The next release won't include many user visible enhancements, only some playback improvements to MPEG-TS and alot of under-the-hoods things.

Honestly, development will be slowing down a bit for the next month or so, there is vacation coming up. :p

Will audio delay be in the next release? (crossing fingers) :)

[Nevilne]

XP's kmixer resampled even when the sample rate was the same, hence why I thought that the same might happen in Win7... If it's not, then that should be OK.

Thanks.

This has me confused, is it better not to resample with reclock on xp then?

[yesgrey]

This has me confused, is it better not to resample with reclock on xp then?
If you want reclock to perform its magic you have to let it resample. However, if you want to bypass kmixer you need to use Kernel Streaming, or then your sound would be resampled once again...

[sneaker_ger]

I really wish there were also an option like:
"Don't load subtitles if matching language found"

and then that option would take preference over all the others. That way all subs could be turned off automatically by the splitter if a matching preferred langauge track exists but enable them if no matching language is present and then follow the other rules for sub language selection. Haali can do this by it's options that allow you to preferentially select audio,sub pairs (which also includes allowing wildcards in the matches along with the keyword "off") but I can't find a similar way to do that with LAV. Or is there a way to accomplish this with the options that exist that I haven't realized?

I can only second this request, the option in Haali's splitter is a very good solution.

[Sebastiii]

Hi Nevcairiel,

I have somes questions, i have see 2 cases :
1) In last few build, i can't on when id do right click on ffdshow icon going to "LAV Splitter Source" properties, it's working for LAV Audio or LAV Splitter or all other directshow filter.
Maybe it's related to the change about name of splitter ? and i assume it's the same on madvr icon.

2) I try to implement the selection of BD subs with Mediaportal BDHandler plugin (I do the selection on matching the stream language name) and i get something strange :
I have tested on 3 BD : Avatar, 127H and Micheal Clayton.

2.1) In graphstudio on LAV Splitter Filter (righ click) :
I get in the list "Forced Subtitles (auto)" with french so here all seems correct :
But when i analyse LCID of the forced new stream, on Avatar and 127H LCID is detected as English 1033 and Micheal Clayton as French 1036.

On my test when LCID is detected as 1033 (English) for forced sub (but display with french in subtitle list on LAV Splitter), the forced subtitle in AVATAR is correctly display as french.
Is it possible to add correct LCID to match the language :) ?

Ps : setting in LAV Splitter was : fre for subtitle and "enable auto forced subtitle stream" ticked

Thanks for help :)
Seb.

[Sebastiii]

See previous post, i hope it's more clear than previous one.

[Snowknight26]

Switching from an LPCM track to a TrueHD track in LAV Splitter causes the file to stop being played if MPC-HC's internal LPCM decoder is used (works with ffdshow).

Sample: http://stfcc.org/misc/lavsplitter.m2ts

[nevcairiel]

Thats a bug in MPC-HC. I'll fix it.

BTW, if you use LAV Splitter, i recommend you also use LAV Audio. ;)

[nevcairiel]

LAV Filters 0.29

LAV Splitter
- The language code is now always shown in the stream changer (as a hint to easily know what to put into the preferred languages fields)
- Enabled ITrackInfo
- Improved H264 in MPEG-TS playback (including an option to try to fix broken HD-PVR recordings)
- Added a programmatic configuration interface for players to configure LAV Splitter

LAV Audio
- Added support for decoding encrypted DVD audio
- Fix decoding of some DTS-HD streams with the ArcSoft decoder
- Added a programmatic configuration interface for players to configure LAV Audio


Download: Installer (both x86/x64) (http://files.1f0.de/lavf/LAVFilters-0.29.exe) -- Zips: 32-bit (http://files.1f0.de/lavf/LAVFilters-0.29.zip) & 64-bit (http://files.1f0.de/lavf/LAVFilters-0.29-x64.zip)

Some quick notes about this release:
LAV Audio can now be used for DVD playback, as well as some PES streams created by some Live TV source filters. Hooray!

On the LAV Splitter side, the most notable change is more improvements to H264 in MPEG-TS playback. If you notice that any file that previously worked is now broken, please try turning off the “Try to fix broken HD-PVR recordings” option, and see if it solves the issue. In any case, please report this!

For anyone that didn't know yet, you can also follow releases on my Blog now, as well as find all Downloads (and a Donation button, for those that insist):
http://1f0.de

Have fun!

PS:
ITrackInfo is enabled now, except for ZoomPlayer, because it broke the player there. Heh.

[Sebastiii]

Thanks Nev :) Great release :)

[ontherocks]

Have uncheck MPC-HC's mpeg splitter cause even if you add LAV to external filters you still need to disable the internal mpeg splitter to be able to use an external (LAV) splitter ?
Yep, that did the trick. Thanks a lot.

I finally found a HD-DVD demo file (HD-DVD Demo 1080p VC-1 DDPlus 5.1 (http://www.techpowerup.com/downloads/530/.html)) which has the standard DD+ (E-AC3) stream (without any AC3 core).
LAV bitstreamed it to my receiver and DD+ light showed up for the first time.
The only combination to bitstream all supported audio is MPC-HC with LAV Splitter & Decoder.......guaranteed.
Thanks nevcairiel :thanks:

[renethx]

Nev

Can LAV Source Spitter force ffdshow raw video filter to be loaded by option? Like Haali Media Splitter can always load VSFilter by option. The main usage is

LAV Source Splitter + LAV CUVID Decoder + ffdshow raw video filter inside WMC internal player

for dual purpose:

1. Subtitles (CUVID is preferred instead of ffdshow Video Decoder because of hardware deinterlacing)
2. SVP (frame interpolation)

Somehow ffdshow raw video filter (or DirectVobSub) is never loaded inside WMC internal player however high its merit value is. On the other hand, if Haali is used and "Autoload VSFilter" is enabled, DirectVobSub is loaded fine in WMC. Yeah, I know it's the player's role what filter is loaded, but WMC is crappy and quite a few people want to use it nevertheless.

[Gleb Egorych]

nevcairiel, thanks for the nice update! :)

[jmone]

Thanks nevcairiel for the release! You really worked quickly getting the DVD Audio out :)

Added a programmatic configuration interface for players to configure LAV

FYI - this is a key feature for Players (like is now being implemented in J Rivers Media Center) where the player can make temporary config changes during commencement of playback. The idea on this is for the "average" user to be able to take advantage of a collection of exposed Filter properties and features without having the first idea on how to manually config the Filters at all (JR is calling it "Red October" in MC). nevcairiel has been working with the MC Devs on the concept with LAV Splitter a part of the first release and now LAV Audio is being added. The result so far is pretty good, especially due to the quick turnaround on issues/ideas/features from nevcairiel!

Thanks
Nathan

[Jay_UK]

Hi there,

Forgive me if this has already been posted, I tried searching (I am a noobie) :D

I currently use ffdshow audio with profiles to control volume levels (I have different sources that present at different levels). I use filter name to determine which profile to use (eg FLV content for me is loud, so needs a volume reduction. I use *FLV Splitter* as a DS filter name match).

Does LAVF support audio profiles?

Also, I can not use ffdshow with LAVF (as a splitter) as the source is always the same name for different sources?

Can anyone help?

Thanks,

J.

[hoborg]

Thanks for .29!

[Dogway]

I have a mkv that doesn't get splitted, audio plays, video screen is black:
mkvmerge v1.5.0
libebml v0.7.5 + libmatroska v0.7.7

[Plutotype]

Hi Nevcairel,

I use 0.28 LAVaudio and have this option "Output only "standard" channel layouts" enabled. Today, at only one movie, I had to disable it, because if enabled, the center channel was (probably) muted or replaced with silence.
It was 6 channel DTS-HD MA, outputted via Realtek chip to 5.1 analog output and per cinch to my Logitech 5.1 set. Sample:
http://www.mediafire.com/?669vteuam2xnms0
Thanks for clarification
Pluto

fixed with 0.29 - thanks Nev!

[mark0077]

OMG thanks for these new versions. Had great fun watching a Queen DVD (vob files) in 24bit 96khz dts (using my tmt 5 dll in mpc dir), deinterlaced using hardware, all the while sent to madVR for output, and flawless. Thanks for this great splitter and decoder combo which helped make it all happen so easily.

Also for the first time played back an old m2ts recording of Sleeping Beauty for a few nieces of mine with 0 frame drops (thanks to the new filter option to fix broken recordings).... I assume it just fixes up bad timestamps or something like that? Playback of this particular file used to be a mess. In anycase, absolutely delighted, thanks again!!! Keep up the good work!

[nevcairiel]

I have a mkv that doesn't get splitted, audio plays, video screen is black:
mkvmerge v1.5.0
libebml v0.7.5 + libmatroska v0.7.7

Some more details would be great. What video codec, what audio codec, oh and a short sample! :)


Does LAVF support audio profiles?

No, and probably never will. Well, it doesn't have a volume filter anyway.

Also, I can not use ffdshow with LAVF (as a splitter) as the source is always the same name for different sources?
Doesn't ffdshow also have the ability to do this based on file name? That sounds much more reliably, matching for "*.flv" instead of "FLV Splitter".


Can LAV Source Spitter force ffdshow raw video filter to be loaded by option? Like Haali Media Splitter can always load VSFilter by option.

It could, i'll think about it some if i like the idea at all.

[jmone]

nevcairiel, what tool do you use to examine the timecode info on the video frames in a M2TS (x264, AC3) video clip. I've got some issues with how PowerDirector9 is rendering transitions on some 1080/50p material and I wnat to "see" what it is doing.
Thanks
Nathan

[nevcairiel]

nevcairiel, what tool do you use to examine the timecode info on the video frames in a M2TS (x264, AC3) video clip.

None, if i'm debugging a issue i usually just add debug lines to LAV Splitter that output the timestamps.

[hoborg]

Nev

Can LAV Source Spitter force ffdshow raw video filter to be loaded by option? Like Haali Media Splitter can always load VSFilter by option. The main usage is

LAV Source Splitter + LAV CUVID Decoder + ffdshow raw video filter inside WMC internal player

for dual purpose:

1. Subtitles (CUVID is preferred instead of ffdshow Video Decoder because of hardware deinterlacing)
2. SVP (frame interpolation)

Somehow ffdshow raw video filter (or DirectVobSub) is never loaded inside WMC internal player however high its merit value is. On the other hand, if Haali is used and "Autoload VSFilter" is enabled, DirectVobSub is loaded fine in WMC. Yeah, I know it's the player's role what filter is loaded, but WMC is crappy and quite a few people want to use it nevertheless.

+1
This will be usefull.

[adam777]

Thank for the new version :)
Live TV PES stream audio decoding still a no go in here, though.

[nevcairiel]

Did you re-register LAV Audio?

Also, Pin information would be useful. Does it connect LAV Audio and decoding fails, or does it not even use LAV Audio to begin with?

[adam777]

OK.
First of all, I renamed my default AAC decoder (for TV viewing, that is) and also FFDShow, to make sure they won't load, and than launched the application.
Judging by Process Explorer, LAV Audio seems to load - http://i.imgur.com/zEUVA.png
I also connected to the graph in Graph Studio utilizing DirectShow Filter Graph Spy and got the following:
http://i.imgur.com/NZTsQ.png
http://i.imgur.com/8SPrh.png
http://i.imgur.com/XqRhs.png
http://i.imgur.com/Mrb6Y.png
http://i.imgur.com/5YCvd.png
Let me know what else is needed and thanks again.

[nevcairiel]

The graph seems rather broken, it didnt actually connect any decoder to the source.

Which Player is that, is it freely available (maybe as trial or somethign) for testing?
I hope one of my TV channels actually uses AAC audio, i think its really uncommon here.

A shot of the Audio Output Pin of the Source would still be helpful!

[nevcairiel]

For mp4, LAV splitter seems not to handle EDTS correctly like previous Haali.
Thus, movie which contains several b-frames (by x264.exe) cannot be seeked accurately or may have desynchronization.

This is all in the hands of ffmpeg. There have however been some fixes coming in since the last time i updated ffmpeg, and i'll be merging those fixes for the next version.

Do you have a sample file which shows the problem, so i can see if things improved?

[adam777]

The graph seems rather broken, it didnt actually connect any decoder to the source.

Which Player is that, is it freely available (maybe as trial or somethign) for testing?
I hope one of my TV channels actually uses AAC audio, i think its really uncommon here.

A shot of the Audio Output Pin of the Source would still be helpful!

It's available for free here - http://pvr.sichbo.ca/
The audio isn't really AAC but HE-AACv2, which is decoded fine when dumping an MPEG-TS from within the program and playing it directly in MPC-HC using LAV Splitter and LAV Audio (I think it even decodes it as stereo which is good, obviously).
Sample (not even sure if it's helpful, but no harm in that) - http://www.multiupload.com/LDFM8YY1VP
And the audio output pin:
http://i.imgur.com/8rpah.png
http://i.imgur.com/wVjaI.png
:thanks:

[nevcairiel]

Ah, its LATM AAC of course, not sure if that can be decoded without proper parsing. I'll do some thinking on this.

You wouldn't happen to have some other TV stations that use MPEG or AC3 Audio, and test if those work? :D

[hoborg]

Ah, its LATM AAC of course, not sure if that can be decoded without proper parsing. I'll do some thinking on this.

You wouldn't happen to have some other TV stations that use MPEG or AC3 Audio, and test if those work? :D

That will be great.
A lot of HTPC users are forced to use Monogram AAC or DivX AAC decoder to be able watch LiveTV/Recordings with LATM/HE-AAC.

[nevcairiel]

The problem with LATM is that part of the parsing is done in LAV Splitter, so i would need to move that over to LAV Audio instead.. we'll see how it turns out.

Too bad i don't have a HE-AAC tv channel that i know of, i'll probably throw some test builds at you later. :p

[ney2x]

Thanks! As usual it's working great! For me/us, last feature that I'm/we're waiting is audio up/down-mixing, and it's perfect! :D

[CruNcher]

Ah, its LATM AAC of course, not sure if that can be decoded without proper parsing. I'll do some thinking on this.

You wouldn't happen to have some other TV stations that use MPEG or AC3 Audio, and test if those work? :D

Hehe samples i guess you have enough (LATM) and yeah that Mplayer works in most Latm situation and also uses ffmlatm and lav audio doesn't is strange (except scalable aac) ;)

[nevcairiel]

File samples are irrelevant, as this is about decoding Live TV Audio.

[Mark_A_W]

Thanks! As usual it's working great! For me/us, last feature that I'm/we're waiting is audio up/down-mixing, and it's perfect! :D

Up/down mixing is pretty useless without delay.

I'd also like to be able to force 32bit FP output.


Then I could replace ffdshow audio with LAV Audio completely.

[e-t172]

Up/down mixing is pretty useless without delay.

I'm confused. Can you give more details as to why is delay useful here?

[nevcairiel]

I'd also like to be able to force 32bit FP output.

Why? That would not have any benefits.
It is however a planned feature, i would just like to know why people want to use it.

[Dogway]

Some more details would be great. What video codec, what audio codec, oh and a short sample!
I tried to split the .mkv to upload a short sample, but the small samples work. So its more likely what I already said, version of mkvmerge.
mkvmerge v1.5.0
libebml v0.7.5 + libmatroska v0.7.7
video: AVC High@L5.1 x264 core 54 rev 600
audio1:libVorbis 1.0 (UTC 2002-07-17) 2ch
audio2:libVorbis 1.0 (UTC 2002-07-17) 2ch
subs:S_TEXT/SSA

Or you tell me how to split .mkv without modifying the header...

[adam777]

The problem with LATM is that part of the parsing is done in LAV Splitter, so i would need to move that over to LAV Audio instead.. we'll see how it turns out.

Too bad i don't have a HE-AAC tv channel that i know of, i'll probably throw some test builds at you later. :p

Oh, joy!
Thanks :p

[nevcairiel]

This build enables AAC LATM parsing in the audio decoder. Let me know if it did anything. :p

http://files.1f0.de/lavf/LAVFilters-0.29-3-g94f4a34.zip

That PVR app you're using is sadly not compatible with my DVB-C/T hardware, so i cannot test it. :(

Note that i also just updated ffmpeg in this build, and have not yet extensively tested since the update. :)

[BloodySword]

@nevcairiel:

Could you please help me develop my new WPF-Based player?
In my first tests I use QuartzTypeLib via Interop. Is this correct?

I need a full list of Classes and DLL Imports I must use for full functionality (eg AspectRatio control of renderers etc.)

Thanks

[hoborg]

This build enables AAC LATM parsing in the audio decoder. Let me know if it did anything. :p

http://files.1f0.de/lavf/LAVFilters-0.29-3-g94f4a34.zip

That PVR app you're using is sadly not compatible with my DVB-C/T hardware, so i cannot test it. :(

Note that i also just updated ffmpeg in this build, and have not yet extensively tested since the update. :)

Just tested in NPVR and LAVF audio decoder refuse connection to MS demultiplexer.

Divx AAC decoder working.

Tested on "Psychoville 2_20110505_22002230.ts" sample.

[madshi]

- Added support for decoding encrypted DVD audio

[...]

LAV Audio can now be used for DVD playback
Can you provide a few details on the technical solution? You know, madVR still doesn't properly do DVD playback on newer OSs, due to the stupid MS navigator making problems. Do you have any clue how to work around that Macrovision crap I'm having problems with?

Thx.

[andyvt]

@nevcairiel:

Could you please help me develop my new WPF-Based player?
In my first tests I use QuartzTypeLib via Interop. Is this correct?

I need a full list of Classes and DLL Imports I must use for full functionality (eg AspectRatio control of renderers etc.)

Thanks

DirectShow.NET (http://directshownet.sourceforge.net/)

[Mark_A_W]

I'm confused. Can you give more details as to why is delay useful here?

Sorry, perhaps I over stated my case. You need both delay and channel mixing (at least) to be a complete audio processor. Without both, I'd still need to use ffdshow.

Why? That would not have any benefits.
It is however a planned feature, i would just like to know why people want to use it.


Ah, because the filter I use after the audio decoder, Convolver Wrapper, only works with 16 bit or 32 bit data.

If it gets 24 bit then the audio is garbled. And as most of my movies have 24 bit audio, I don't want to send out 16 bit. Again, without being about to force 32bit FP, I'd be using ffdshow in the chain to force it.

Compatibility is the reason. Reclock will do the same thing, but at the end of the chain - which is too late for me.


Cheers

Mark

PS. I just donated - it's on the Downloads page on Nev's blog. Get to it you bunch of tightarses, buy nev a beer :p

[jmone]

PS. I just donated - it's on the Downloads page on Nev's blog. Get to it you bunch of tightarses, buy nev a beer :p

+1 - it's at http://1f0.de/downloads/ if you have trouble finding it.

[SamuriHL]

Yup, I just donated, as well. :)

[Skinleech]

This build enables AAC LATM parsing in the audio decoder. Let me know if it did anything. :p

http://files.1f0.de/lavf/LAVFilters-0.29-3-g94f4a34.zip

That PVR app you're using is sadly not compatible with my DVB-C/T hardware, so i cannot test it. :(

Note that i also just updated ffmpeg in this build, and have not yet extensively tested since the update. :)

I'll test this later with DVBViewer & UK Freeview HD, I'm just about to go out for a few hours.

I mentioned this several pages back but knew it was low on your priority list - nice to see it getting some love, thank you :)

[Jay_UK]

Hi Nevcairiel,

Thanks for responding to my questions about audio profiles and the issue about using ffdshow (and profiles) with your splitter.

The problem I have, is that my source is a stream and not a file. As such I am not sure if its possible to use your splitter (lavf splitter) with ffdshow profiles to detect just FLV content?

Thanks,

J.

ps - From what I saw when testing, it looked great! Keep up the good work :)

[adam777]

This build enables AAC LATM parsing in the audio decoder. Let me know if it did anything. :p

http://files.1f0.de/lavf/LAVFilters-0.29-3-g94f4a34.zip

That PVR app you're using is sadly not compatible with my DVB-C/T hardware, so i cannot test it. :(

Note that i also just updated ffmpeg in this build, and have not yet extensively tested since the update. :)

Hi nevcairiel,
Tried the build - still not getting any sound.
Graph studio, pin information etc. exactly the same as last time.
Adam.

[CruNcher]

Yep no sound in many samples (also the extreme wrong muxed audio ones, signaling mp1 or mp2 instead of latm (aac)) still mplayer is the only one playing them all (though the corrupted ones better then the remuxed ones, also funny :D).

http://www.megaupload.com/?d=BPADC18B <- remuxed one no sound @ all except with DivX AAC Decoder (but to fast playback,video is faster then audio :D) stream even accepted by videolan (video plays, doesn't accept the original) mplayer (video plays but not smooth anymore, audio also doesn't play anymore like it does with the corrupted original :D)

Though with these samples lav audio decoder connects but for the corrupted streams shows mp1/mp2float and for the remuxed one aac_latm but still no audio.

Audio works (to fast Video Playback):

Filter : DivX AAC Decoder - CLSID : {2CCC9657-58A9-41AC-AA39-451202B98FAF}

- Connected to:

CLSID: {171252A0-8820-4AFE-9DF8-5C92B2D66B04}
Filter: LAV Splitter
Pin: Audio

- Connection media type:

Audio: AAC(LATM) 24000Hz stereo

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: Unknown GUID Name {000001FF-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 256000
cbFormat: 20

WAVEFORMATEX:
wFormatTag: 0x01ff
nChannels: 2
nSamplesPerSec: 24000
nAvgBytesPerSec: 0
nBlockAlign: 1
wBitsPerSample: 0
cbSize: 2 (extra bytes)

pbFormat:
0000: ff 01 02 00 c0 5d 00 00 00 00 00 00 01 00 00 00 ÿ...À]..........
0010: 02 00|14 90 ...�


Audio Fails (Video Playback ok):

Filter : LAV Audio Decoder - CLSID : {E8E73B6B-4CB3-44A4-BE99-4F7BCB96E491}

- Connected to:

CLSID: {171252A0-8820-4AFE-9DF8-5C92B2D66B04}
Filter: LAV Splitter
Pin: Audio

- Connection media type:

Audio: AAC(LATM) 24000Hz stereo

AM_MEDIA_TYPE:
majortype: MEDIATYPE_Audio {73647561-0000-0010-8000-00AA00389B71}
subtype: Unknown GUID Name {000001FF-0000-0010-8000-00AA00389B71}
formattype: FORMAT_WaveFormatEx {05589F81-C356-11CE-BF01-00AA0055595A}
bFixedSizeSamples: 1
bTemporalCompression: 0
lSampleSize: 256000
cbFormat: 20

WAVEFORMATEX:
wFormatTag: 0x01ff
nChannels: 2
nSamplesPerSec: 24000
nAvgBytesPerSec: 0
nBlockAlign: 1
wBitsPerSample: 0
cbSize: 2 (extra bytes)

pbFormat:
0000: ff 01 02 00 c0 5d 00 00 00 00 00 00 01 00 00 00 ÿ...À]..........
0010: 02 00|14 90 ...�

[Boltron]

Finally, a Donate button! Your awesome Nev, I am happy to support you.

[CiNcH]

@ nevcairiel,
can you please point me to the definition of the ITrackInfo interface?

[nevcairiel]

Thanks to everyone who donated, you guys are really awesome. :)

can you please point me to the definition of the ITrackInfo interface?

http://git.1f0.de/gitweb?p=lavfsplitter.git;a=blob;f=common/includes/ITrackInfo.h;hb=HEAD


Can you provide a few details on the technical solution? You know, madVR still doesn't properly do DVD playback on newer OSs, due to the stupid MS navigator making problems. Do you have any clue how to work around that Macrovision crap I'm having problems with?

I'm not sure why that Macrovision failure happens. I don't know if the renderer needs to implement something special to indicate that it supports Macrovision.
All i basically implement is a IKsPropertySet in the Input Pin which accepts a bunch of DVD propertys from the Naviagtor (these -> http://msdn.microsoft.com/en-us/library/dd388585(v=vs.85).aspx)

Everything else is basically just using DeCSS to decrypt, which the renderer doesn't have any say in.

Implementation is here: http://git.1f0.de/gitweb?p=lavfsplitter.git;a=blob;f=common/DSUtilLite/DeCSS/DeCSSInputPin.cpp;hb=HEAD

[blahism]

how much of a donation would i have to make to get media foundation splitter reconsidered? :) i unfortunately don't have the time to review the code to make code contributions but i wouldn't mind donating money if that inspires any movement there

[nevcairiel]

Since my time is quite limited, it'll probably not be moved up in the time table.
Its not that i dont want to do it, its just that i have alot other things to do that serve a real world purpose for me. :p

[Skinleech]

Tried the LATM build with no joy. I have LAV Audio selected in both audio filters sub menus in Dvbviewer but it always defaults to the MS DTV Audio decoder.

[nevcairiel]

DVBViewer works just fine for me, however i have no AAC channels, so i can only test AC3 and MPEG Audio. I would need some Pin informations for that.

[CiNcH]

I am attaching a small LOAS/LATM packaged AAC-HE sample.

I have also got samples from Japan where there is a 2.0 -> 5.1 change and vice-versa mid-stream.

[tormento]

I have totally switched from Haali+CoreAVC(registered and paid)+FFDAnythingelse to LAV "suite".

If I try to play a video only mkv output from Megui, MediaPlayerHC is unable to seek when clicking on the progress bar and jumps back to 00:00.

Bad explanation, please try yourself ;)

The same video works ok with haali or internal mphc splitter.

[fps]

- Fix decoding of some DTS-HD streams with the ArcSoft decoder

Thanks for this bugfix, this seems to have solved some issues, where the channel output switched from 5.1 to 6.1 after some time, which caused a lot of audio dropouts. I've only noticed this problems with DTS-HD tracks (using the arcsoft decoder, didn't have time to check the internal one), AC3 playback was fine all the time.

However, I've encountered another problem today. At a specific scene of the Die Hard 4.0 Blu-Ray (english DTS-HD track) I either get lots of audio dropouts or the audio is kind of played in the wrong order, some parts from some time after the current get played back and so forth. This is pretty much always reproducible with the internal dts decoder, it doesn't happen all the time with the arcsoft one. Enabling bitstreaming shows the same problem. With libtds from ffdshow audio playback is fine.
Setup: Win7/32, madVR, CoreAVC 2.55, mpcHC, LAVFilters 0.29

Do you have this Blu-Ray or would a sample/log help?

[nevcairiel]

I have it on Blu-ray, but who knows if its the same version. What time in the movie does the problem occur?

[endpoint122]

how make this work at my mpc-hc
i downloaded and setup then i added it make it my perfer but when i see filters it's not in them why?

[fps]

I have it on Blu-ray, but who knows if its the same version. What time in the movie does the problem occur?
My Blu-Ray has a runtime of 2h08m44s. The scene is at about 15 minutes, where the terminator figure hits the delete key and the appartment is blown up. After that you see the sniper looking below, that's where the problem starts. You hear him zipping the case and the other guys caughing, while he's still looking at the appartment. Might take a few runs to reproduce the problem, but I'd say it's there about 80% of the time.

[Andy o]

what do you see in filters?

[fps]

Do you mean the filters menu of MPCHC, or the status page of LAVAudio?
In MPCHC it's just the usual, Reclock Audio Renderer, madVR Renderer, CoreAVC, LAV Audio and LAV Splitter.

The status page of LAVAudio shows

Input:
Codec: dca
Channel: 6/0x3f
Sample Rate 48000
Format: 32 bit float

Output:
Codec: PCM
Channel: 6/0x3f
Sample Rate 48000
Format: 32 bit float

So it looks like normal I guess. With v0.28 it was 7 channels at the output page, which I described two posts before.
But with 0.29 it correctly shows 6 channels even after the problem.

[endpoint122]

right click on the movie then filters to see which one is use

[fps]

That's what i posted above ;).

[Ap3 n1nja]

Hey guys, I don't know if this is what it's suppose to be, but does anyone else see "C#¼¯" as an audio codec when playing files with vorbis audio? (I'm using Potplayer btw)

[Andy o]

sorry, my question above was for endpoint122

[Skinleech]

DVBViewer works just fine for me, however i have no AAC channels, so i can only test AC3 and MPEG Audio. I would need some Pin informations for that.

No problem - assuming you can let me know how to access the info in DVBViewer? Freeview HD only broadcasts in AAC LATM, and similar to what CiNcH has mentioned, it likes to switch from 2.0 to 5.1 mid-stream.

Thanks :)

[pankov]

Guys,
if you use the DVBViewer Recording Service ... or even the NetStreeming plugin in DVBViewer you can let nevcairiel access your live video stream.
This way he can debug the AAC LATM problem and fix it.
Even if your internet connection doesn't have enough upload speed I think he'll be able to see the actual data.
If you need help with configuring this send me PM.

[Gleb Egorych]

Bug in LAV Audio interacting with dtsdecoderdll.dll 1.1.0.0. Here is small sample (http://www.mediafire.com/?u7difyid5s1eqer). In the middle of the file sound mutes. If right after file open you seek to 6 sec position then there is no sound mute.
Version 1.1.0.1 is also affected, versions 1.1.0.7/8 are unaffected. ASAudioHD.ax with 1.1.0.0 plays fine without mute.

BTW there is a discussion (http://forum.doom9.org/showthread.php?p=1504294#post1504294) in eac3to topic regarding best arcsoft version, seems there is no best version unfortunately.

[nevcairiel]

Bug in LAV Audio interacting with dtsdecoderdll.dll 1.1.0.0

Fixed, thanks for the report and sample.

Regarding the "best" version - i'm using 1.1.0.8 these days, because i've never seen an actual 6.0 sample in real life (only layout thats known to be really broken with 1.1.0.8), and overall it seems to be more stable for me.

[Gleb Egorych]

You're welcome :)

You wrote only about 6.0 problem with 1.1.0.8 while in eac3to thread some people wrote versions .7 and .8 couldn't decode DTS-HD MA 6.1 properly (see here (http://forum.doom9.org/showthread.php?p=1459063#post1459063) and here (http://forum.doom9.org/showthread.php?p=1461999#post1461999)). Does it work for you?

Sumup of ArcSoft versions is here (http://forum.doom9.org/showthread.php?p=1461841#post1461841).

[nevcairiel]

6.1 works for me, indeed. Might've been something in eac3to, because it was developed for 1.1.0.0 initially.

[Gleb Egorych]

Sorry for bothering, 6.1 works without artifacts (ringing, distortions like reported with eac3to), right?

[nevcairiel]

Go try it, its not like its some super magic to exchange that one dll.

[Gleb Egorych]

I have only two samples of DTS 6.1, one with noise (http://www.mediafire.com/?k77pldzems9stn3) to test channel mapping and another (http://www.mediafire.com/?jcie332d4231g8r) with voice. With both of them it's hard to differ decoder versions so I used dump.ax to compare results. Outputs of 1.1.0.1, 1.1.0.7 and 1.1.0.8 are bit-identical and differ from output of 1.1.0.0.

[cho-gde-kto]

TrueHD (Core AC3 5.1 640) passthrough over SPDIF....
- * Splitter + LAV Audio = PCM (stereo) or no sound
- * Splitter + Cyberlink Audio = PCM (stereo)
- CLDemuxer2 + LAV Audio = no sound
- CLDemuxer2 + Cyberlink Audio = Dolby (5.1)

P.S. with any DTS* - no problem

[nevcairiel]

TrueHD does not work over SPDIF, and never will.

On m2ts files, LAV Splitter will create a separate AC3 track for backwards compat on TrueHD files.

[cho-gde-kto]

TrueHD does not work over SPDIF, and never will.

On m2ts files, LAV Splitter will create a separate AC3 track for backwards compat on TrueHD files.

yes, but this track (AC3 track for backwards compat on TrueHD files) is wrong passed through spdif with LAV* (PCM Stereo)

with PDVD, TMT, Splash, DAUM - Dolby 5.1 (AC3)

E-AC3 - only with PDVD, TMT I hear DOLBY (AC3)
(7/64, ASUS P5B-E, MicroLab H-500D)

http://www.multiupload.com/KEW7L2LUZG
http://www.multiupload.com/YFK2UDUWOK


MPC-HC > Cyberlink Denuxer2 + Cyberlink Audio Decoder + Cyberlink Video Decoder (11) = perfect :cool: (bitstream any HD-audio tracks on m2ts over SPDIF (DTS-MA/HRA with DTS CORE, TrueHD(with embedded AC3/CORE), and even E-AC3 ! :eek: , ...))

My microlab shows - "DTS" and "Dolby Digital" :cool: , but with LAV* (or with ffdshow, mpadecfilter,...) - "DTS" on DTS* tracks and "PCM Stereo" on TrueHD/E-AC3 :(
Why???

[jj666]

Tested my Akira BD remux to .TS which is rather a peculiar one with 5.1 TrueHD and 2.0 core for the Japanese, and 5.1/5.1 for the English, that worked perfectly fine (with correct cores being played back on both files). Manually selecting the core of course.

I don't have TrueHD bitstreaming ticked, but can confirm that it's playing 2.0 PCM (which of course is all the SPDIF is capable of) automatically. Is there a way to preselect the core as happens with DTS-HD, if bitstreaming TrueHD isn't selected?

Cheers,

-jj-

[fps]

I have it on Blu-ray, but who knows if its the same version. What time in the movie does the problem occur?

My Blu-Ray has a runtime of 2h08m44s. The scene is at about 15 minutes, where the terminator figure hits the delete key and the appartment is blown up. After that you see the sniper looking below, that's where the problem starts. You hear him zipping the case and the other guys caughing, while he's still looking at the appartment. Might take a few runs to reproduce the problem, but I'd say it's there about 80% of the time.
Were you able to reproduce the problem?

[bjd]

@nevcairel

I installed v0.29 splitter/audio by clicking on the .bat files in the folder and then reset the media type lists for the audio decoder in Mpc-Hc. I noticed MEDIATYPE_DVD_ENCRYPTED_PACK does not appear in the allowed types, so I can only get DVD audio to work by manaully adding it. Have I missed something ?

[nevcairiel]

Were you able to reproduce the problem?

Didn't try yet.

@nevcairel

I installed v0.29 splitter/audio by clicking on the .bat files in the folder and then reset the media type lists for the audio decoder in Mpc-Hc. I noticed MEDIATYPE_DVD_ENCRYPTED_PACK does not appear in the allowed types, so I can only get DVD audio to work by manaully adding it. Have I missed something ?

You must've missed something, that shows up just fine here.

---
As a info for everyone else, i'll be on vacation, so don't expect answers for the next 2 weeks. :p

[SamuriHL]

As a info for everyone else, i'll be on vacation, so don't expect answers for the next 2 weeks. :p

Good thing I'll be drunk on a beach all next week then. :D

[Gleb Egorych]

I updated my previous post (http://forum.doom9.org/showthread.php?p=1507687#post1507687) with correct sample links. I compared different version of arcsoft decoders (1.1.0.0, .1, .5, .7, .8) and can say that DTS 6.1 problems of .5+ versions with eac3to are clearly eac3to problems, not decoder. Moreover, .0 and .1 often output 24 bit when source is 16 bit, .5+ output correct bitdepth. Guess it's the reason why eac3to fails to decode DTS 6.1 properly with the latter versions.

Even if output bitdepth is identical output pcm is different (tested with dump.ax). Version .0 gives one version, .1 gives the second one, .5+ the third. Don't know which is better but I assume .5+ versions are better.

Version .5 is very similar to .7, same mono bug. Don't have a sample to check 6.0 bug but I assume it's here like in .7 and .8.

i'll be on vacation, so don't expect answers for the next 2 weeks. :p
Have a nice trip :)

[nevcairiel]

Here (http://files.1f0.de/samples/330.dts) is a 6.0 channel id sample

[bjd]

I installed v0.29 splitter/audio by clicking on the .bat files in the folder and then reset the media type lists for the audio decoder in Mpc-Hc. I noticed MEDIATYPE_DVD_ENCRYPTED_PACK does not appear in the allowed types, so I can only get DVD audio to work by manaully adding it. Have I missed something ?


You must've missed something, that shows up just fine here.

Yes - had to remove the filter from the list and re-add to get the new media types, just resetting does nothing. All sorted now.

[nevcairiel]

Oh yeah, i noticed that before. Sorry, thought you meant that with "reset".

[pirlouy]

Option "expand mono to stereo" is nice; could you create an option "expand stereo to rear/surround" ? That would be great (and so, no need for a complete matrix in my case).

[clsid]

LAVAudio does not decode Vorbis in MKV with Haali as splitter. Is it just a missing mediatype?

Can you let LAVAudio store the format settings in a similar way as LAVSplitter? That is easier to use by external config apps.

[Mercury_22]

Option "expand mono to stereo" is nice; could you create an option "expand stereo to rear/surround" ? That would be great (and so, no need for a complete matrix in my case).

+1 for REAR:)

[Gleb Egorych]

Here (http://files.1f0.de/samples/330.dts) is a 6.0 channel id sample
Thanks, nev. Contrary to my assumption version 1.1.0.5 plays this file correctly. In fact it is the only version of dtsdecoderdll.dll which plays the file with proper channel number and proper bitdepth. Sadly it has mono bug like 1.1.0.7 does.

[JarrettH]

Can I have LAV Audio downmix to 2.0 channels? I used to use AC3 filter for doing this :cool:

[nevcairiel]

Yes - had to remove the filter from the list and re-add to get the new media types, just resetting does nothing. All sorted now.

I fixed the behaviour of that reset button in MPC-HC 3222.

Can I have LAV Audio downmix to 2.0 channels? I used to use AC3 filter for doing this :cool:

Thats a planned feature, but not yet supported.

[Gleb Egorych]

About arcsoft and mono: tested DTS-HD MA 1.0 24/48 file, 1.1.0.0 and 1.1.0.1 produce identical output, 1.1.0.8 produces a different output. Nice behaviour for a lossless decoder.

[JarrettH]

Ok thanks, not a big deal, actually possibly, but I'll just reinstall AC3 filter to do that part. :p

[nevcairiel]

About arcsoft and mono: tested DTS-HD MA 1.0 24/48 file, 1.1.0.0 and 1.1.0.1 produce identical output, 1.1.0.8 produces a different output. Nice behaviour for a lossless decoder.

The thing about 1.1.0.8 is that it produces Stereo output for Mono files, and i take that output and reduce it back to one channel, and i'm not 100% sure if thats a lossless operation.

I can run a comparison myself and see if i can tweak the conversion algorithm a bit.

[Sebastiii]

Hi Nevcairiel,

What do you think about talking : http://forum.team-mediaportal.com/756617-post607.html ?
Thank you.

[ney2x]

FYI.

I asked shark007 (http://shark007.net/) to include LAV CUVID and LAV Filters in his codec pack and he says he will in his next release. And also Win7DSFilterTweaker v4.2 (http://www.codecguide.com/windows7_preferred_filter_tweaker.htm) now supports LAV CUVID (thanks to clsid) and Codec Tweak Tool v4.9.5 (http://www.free-codecs.com/download/codec_tweak_tool.htm) supports LAV Filters. These utilities are useful to those who used WMP12/WMC as their main player.

[BloodySword]

Found a small bug with TTA, the last buffer seems to be dropped at the end of playback. Don't know if this is a avformat problem or even avcodec problem. This problem is reproduceable with ANY .tta file I have. They were encoded with official ttaenc.exe, latest version.

[ney2x]

Off-topic:

Is there a way to show subtitles like ffdshow subtitle placement/letterbox/vertical position 98% (bottom)

http://img861.imageshack.us/img861/2369/ffdshowc.jpg

And not like this
DirectVobSub
http://img855.imageshack.us/img855/733/directvobsub.jpg

Note: Sorry, I don't know the right word for what I'm trying to achieve in DirectVobSub, is it video overlay??? :D

Already tried to asked support in other thread but no one answering me. I decided to ask here cause people here are accommodating :)

[BloodySword]

Is the black letterbox encoded in the video? If not, I think it is only possible with EVR.

[pirlouy]

I think it's a job for subtitle renderer. So if you use MPC-HC, you have to play with MPC subtitles options.

[Johan76]

I have been starting to test the LAVF filters recently but have an odd problem with the audio.

All decoded audio gets distorted. I can hear the actual audio but with heavy distortion ontop.
Bitstreaming SPDIF for AC3 and DTS works perfectly but not if I turn off bitstreaming and decode.

I have so far only tested AC3 and AAC format.

I have an old mainbord with Nforce 220 chipset and onboard Nforce MGP audio.
FFDshow audio and other codecs have always been working so something seems to be up with my system and LAV Audio.
LAVF splitter and FFDshow audio works so it must be the audio part where something strange occurs.

What do you need from me to be possible to check this out? I have no clue even where to start looking.

[nevcairiel]

Both AC3 and AAC get decoded to floating point, some older audio drivers don't handle that properly. In a future version, you may be able to configure the desired output format.

You can confirm that this is the problem by simply going into the ffdshow audio config, and on the Output tab, unselect every format except 32-bit float. If you get the same distortions, thats the problem.

[lboregard]

nevcairiel,

i have some blu-rays with multi-channel lpcm audio that my receiver (denon avr-1909) interprets as stereo ...

mpchc + lavsplliter + lavaudio

i had this working with reclock before (receiver showing multi channel something)

in the status tab for lavaudio, i can see it sending 6 channels as output

any idea what might be wrong ?

[SamuriHL]

Your Windows Audio settings are wrong. To do it with just LAV Audio you need to match the Windows Audio settings with the number of channels. ReClock gave you channel matching with exclusive mode, but, without exclusive mode the channels must match. Most likely you have it set to stereo.

[lboregard]

Your Windows Audio settings are wrong. To do it with just LAV Audio you need to match the Windows Audio settings with the number of channels. ReClock gave you channel matching with exclusive mode, but, without exclusive mode the channels must match. Most likely you have it set to stereo.

excellent ! i had it set to stereo .. changed it to 5.1 and voila ! MULTI CH IN input bliss on my receiver :)

thanks SamuriHL !

[SamuriHL]

excellent ! i had it set to stereo .. changed it to 5.1 and voila ! MULTI CH IN input bliss on my receiver :)

thanks SamuriHL !

No problem. Glad I was able to help.

[CiNcH]

I think that I found a potential issue.

I am using LAV Splitter to play BD inside the DVBViewer (it detects the main movie mpls by itself and opens it with LAV). When stopping playback, the DVBViewer remembers the position and will continue from there when restarting playback. This seems to cause problems with certain disks, e.g. "Angels and Daemons" (Illuminati). The DVBViewer just crashes with a C++ exception. I found out that the main movie is scattered among many m2ts files. I think that this happens when playback is stopped and resumed after the first m2ts file.

May this be a seeking bug with seamless branching BD's?

[BloodySword]

Found a small bug with TTA, the last buffer seems to be dropped at the end of playback. Don't know if this is a avformat problem or even avcodec problem. This problem is reproduceable with ANY .tta file I have. They were encoded with official ttaenc.exe, latest version.

[RealSnoopyDog]

Hi CiNcH! I don't have this problem here, just checked it with Illuminati. Do you use one of the latest ffdshow builds where nevcairiel provided a patch for crashes in H.264 DXVA playback? Even build 3878 was instable, it always crashed at the same playback position of a special movie. With build 3882 this was gone. (and i compiled my own build right after nev's fix, this was also stable)

I only have one issue with .mpls playback: when i watch a movie (.mpls / .m2ts) and select another mpls playlist (e.g. bonus material) while the current one is still playing back, then LAV starts accessing the disk but never ends. I have to kill the host program with the task manager. Sometimes the first change between .mpls files works but at the latest the third or fourth change causes this behaviour with all movies that i own. I start a new playback while another playback is still running. The original filter in DVBViewer does not show this behaviour.

[Johan76]

Both AC3 and AAC get decoded to floating point, some older audio drivers don't handle that properly. In a future version, you may be able to configure the desired output format.

You can confirm that this is the problem by simply going into the ffdshow audio config, and on the Output tab, unselect every format except 32-bit float. If you get the same distortions, thats the problem.

You are very right. Tested and confirmed. The only format not working was 32 bit floating. I have never thought about those options since it has just been working.

Are selectable output any near in the future? I can unfortunately not use LAV audio until this is possible :/

Many thanks for pointing this out. Now I at least know why! :)

[pankov]

Off-topic:

Is there a way to show subtitles like ffdshow subtitle placement/letterbox/vertical position 98% (bottom)

http://img861.imageshack.us/img861/2369/ffdshowc.jpg

And not like this
DirectVobSub
http://img855.imageshack.us/img855/733/directvobsub.jpg

Note: Sorry, I don't know the right word for what I'm trying to achieve in DirectVobSub, is it video overlay??? :D

Already tried to asked support in other thread but no one answering me. I decided to ask here cause people here are accommodating :)
ney2x,
if you do insist on using DirectVobSub instead of FFDShow subtitles (either in the decoder or in the separate filter) you can try the option "Vertical padding" in VobSub's General tab. I remember I used it a long time ago but I think it's broken in the latest versions - not sure.

[BloodySword]

You are very right. Tested and confirmed. The only format not working was 32 bit floating. I have never thought about those options since it has just been working.

Are selectable output any near in the future? I can unfortunately not use LAV audio until this is possible :/

Many thanks for pointing this out. Now I at least know why! :)

You could try to update your audio drivers. Wich OS are you running? Since Vista, audio conversions are done by CPU, so windows will convert the formats. In XP a driver update might also add CPU conversion of float to integer.

[e-t172]

You could try to update your audio drivers. Wich OS are you running? Since Vista, audio conversions are done by CPU, so windows will convert the formats. In XP a driver update might also add CPU conversion of float to integer.

You could also use Reclock to convert from floating point to integer.

[CiNcH]

Hi CiNcH! I don't have this problem here, just checked it with Illuminati. Do you use one of the latest ffdshow builds where nevcairiel provided a patch for crashes in H.264 DXVA playback? Even build 3878 was instable, it always crashed at the same playback position of a special movie. With build 3882 this was gone. (and i compiled my own build right after nev's fix, this was also stable)
I see. I am using latest ICL build 3853 by clsid. Think I will switch to the generic builds then. I just had the problem that DVBViewer always crashed when trying to resume the movie. Only after resetting position to 0 it worked again. Thought this could be due to seamless branching and a seeking error. Guess I will have to do more serious testing on this issue then, this time with latest ffdshow build however.

[Johan76]

You could try to update your audio drivers. Wich OS are you running? Since Vista, audio conversions are done by CPU, so windows will convert the formats. In XP a driver update might also add CPU conversion of float to integer.

My HTPC is on WinXP.
Not sure if there are any updated drivers.
I can look but it is always hard to find good drivers for such old hardware.

But since I have a "new" GPU which handles even bluray material I see no point in buying new hardware just because... :)

I will look for newer driver.

BloodySword> Thanks for reclock tips. I will consider. However I cannot use this if it will mess with the bitstreaming.

[ney2x]

ney2x,
if you do insist on using DirectVobSub instead of FFDShow subtitles (either in the decoder or in the separate filter) you can try the option "Vertical padding" in VobSub's General tab. I remember I used it a long time ago but I think it's broken in the latest versions - not sure.

Yeah, I'm completely getting rid of ffdshow and I'll be using LAV Suite (CUVID/Filter). Anyways, my problem solved today, by adjusting the vertical placement and unchecking "Positions subtitle relative to the video frame"

http://img864.imageshack.us/img864/659/mpcj.th.jpg (http://imageshack.us/photo/my-images/864/mpcj.jpg/)

http://img838.imageshack.us/img838/2518/mpc2.th.jpg (http://imageshack.us/photo/my-images/838/mpc2.jpg/)

And also I am using DirectVobSub v2.40.3164 (http://xhmikosr.1f0.de/index.php?folder=bXBjLWhjL21wYy1oY19hcHBzL3ZzZmlsdGVy) . Oh! The reason I used separate subtitle filter is because I also used Windows Media Center (Windows 7 x64).

[Mercury_22]

Yeah, I'm completely getting rid of ffdshow and I'll be using LAV Suite (CUVID/Filter). Anyways, my problem solved today, by adjusting the vertical placement and unchecking "Positions subtitle relative to the video frame"

http://img864.imageshack.us/img864/659/mpcj.th.jpg (http://imageshack.us/photo/my-images/864/mpcj.jpg/)

http://img838.imageshack.us/img838/2518/mpc2.th.jpg (http://imageshack.us/photo/my-images/838/mpc2.jpg/)

And also I am using DirectVobSub v2.40.3164 (http://xhmikosr.1f0.de/index.php?folder=bXBjLWhjL21wYy1oY19hcHBzL3ZzZmlsdGVy) . Oh! The reason I used separate subtitle filter is because I also used Windows Media Center (Windows 7 x64).

You can also use MPC-HC as default player in "Windows Media Center (Windows 7 x64)" (http://mikinho.com/wmc/open-with/) and you'll not need any other filters :)

[clsid]

if you do insist on using DirectVobSub instead of FFDShow subtitles (either in the decoder or in the separate filter) you can try the option "Vertical padding" in VobSub's General tab. I remember I used it a long time ago but I think it's broken in the latest versions - not sure.Vertical padding has been broken for quite some time now. If someone can narrow down the range of revisions in which it got broken, it would increase the chance of a quick fix. So if anyone is bored and wants to do some testing, please do!

[ney2x]

You can also use MPC-HC as default player in "Windows Media Center (Windows 7 x64)" (http://mikinho.com/wmc/open-with/) and you'll not need any other filters :)

Oh well! I ask for 1 help and here you come you help me more... :D That's why I love doom9 :p

Vertical padding has been broken for quite some time now. If someone can narrow down the range of revisions in which it got broken, it would increase the chance of a quick fix. So if anyone is bored and wants to do some testing, please do!

Do you know a site to where I can download different revisions of VSFilter.dll only? Cause I can't parallel download here (http://www.xvidvideo.ru)

Edit: Upon thorough testing different revisions I managed to find the last working "vertical padding" which is DirectVobSub v2.40.2.3104_x86 (http://www.mediafire.com/?1el194y25jbut95) and from DirectVobSub v2.40.2.3112 onwards are broken. FYI.

Question:
Is CCCP Project (http://www.cccp-project.net/beta/) related to MPC-HC Team? Cause I just found out awhile ago that their revisions of VSFilter are not broken.

[clsid]

Are you sure 3104 is working? Because afaik it was broken long before that.

CCCP probably has some custom code. Perhaps they have already located the bug.

[Johan76]

Tried with reclock to convert to 16 bit integer. Kind of work. Sound is good but I get other problems. Easier for me to stick with ffdshow for now.

Sent from my HTC Desire using Tapatalk

[ney2x]

Are you sure 3104 is working? Because afaik it was broken long before that.

CCCP probably has some custom code. Perhaps they have already located the bug.

http://img545.imageshack.us/img545/1623/vobsub.th.jpg (http://imageshack.us/photo/my-images/545/vobsub.jpg/)

But the problem with 3104 is it stays in my system tray and consuming memory even after exiting the player.

And also I want to report the latest revision DirectVobSub v2.40.3237 (http://xhmikosr.1f0.de/index.php?folder=bXBjLWhjL21wYy1oY19hcHBzL3ZzZmlsdGVy) still broken. I thought someone fixed it but still a no go...

This maybe my last post regarding DirectVobSub cause this is OFF-TOPIC. Thanks for your time.

[Gleb Egorych]

*OFFTOPIC*
I'm using Zoom Player and neither 3042 nor 3104 works as expected with vertical padding. Version from CCCP works but there is a problem with aspect ratio on files wider than 16:9 (I use 16:9 extension).

EDIT: problem with aspect ratio was pure ZP problem, solved. As I understand CCCP custom patches only revert particular MPC-HC revisions, 339 and 1096 to be exact.
Sorry for offtopic.
*OFFTOPIC*

[Portioli]

may i ask you sth?
if i have LAV FILTERS in wmp & w7mc(using codec tweak tool) the only way to have external subs is using ffdhow?

[Thunderbolt8]

DirectVobSub v2.40.3237 still broken. I thought someone fixed it but still a no go...OT, but +1.
has already been like that with .2995 (and maybe also before).

[Rectal Prolapse]

Hello! I would like to confirm if LAV Audio + ARCSoft decoder will properly decode DTS-HD Master Audio. Will it do this losslessly or will it only decode the DTS core? I tried searching the thread but could not confirm.

Thanks!

[SamuriHL]

Lossless.

[Portioli]

the best way to have lossless DTS-HD MA with the exact same sample size & sample rate happens with lav audio and the dtsdecoderdll.dll using reclock in wasapi exclusive

http://i.imgur.com/56TOf.jpg

[Gleb Egorych]

Portioli, your screenshot shows that you don't have lossless and exact sample rate because ReClock definitely resamples on your screenshot. It sends bit exact resampled audio not source audio.

[Gleb Egorych]

I tested 5 different versions of dtsdecoderdll.dll along with LAV Audio, here are results.

Versions tested: 1.1.0.0, .1, .5, .7, .8.

Samples tested:
1) Lossy DTS 2.0, 2.1, 5.1, 6.0, 6.1 configs, 16/48, 24/48 and 24/96 bitdepth/sample rate;
2) DTS-MA 1.0, 2.0, 2.1, 5.1, 6.1, 7.1 configs including 7.1 "strange setup" config (eac3to term), 16/48, 24/48, 24/96 variants.
My results differ from results that some people posted in eac3to thread, I guess there may be some eac3to specific problems.

lossy DTS
1) 1.1.0.0 and 1.1.0.1 always decode lossy DTS as 24 bit, and their decoding results differ from each other.
2) 1.1.0.5 and up decode in proper bitdepth.
3) Versions 1.1.0.5 and up decode lossy DTS identically.
4) 1.1.0.1 and 1.1.0.5 decode 24 bit lossy DTS identically.
5) Decoding of 16 bit lossy DTS was changed from 1.1.0.0 to 1.1.0.1 and from 1.1.0.1 to 1.1.0.5.
6) 1.1.0.7 and 1.1.0.8 decode 6.0 without back center channel.

The conclusion for lossy: I assume the best version for lossy DTS is 1.1.0.5. It decodes all configurations I've tested, in proper bitdepth, and decoding algorithm didn't change since this version (except 6.0 bug in .7 and .8).

DTS-MA
All versions output identical streams except mono case. Mono decoding was broken in 1.1.0.5 and partially fixed in 1.1.0.8 (outputs as stereo). Note that MA 7.1 16/48 "strange setup" is decoded as 24/48 by all versions.

The conclusion for lossless: the best versions are 1.1.0.0 and 1.1.0.1.

Of course, to be sure that decoding is done properly some "reference proper" decoder is needed. I don't have it.

EDIT: Forgot to mention about 2.1 bug. None of the versions can decode LFE channel in 2.1 files.

[nevcairiel]

"Lossy" codecs do not have a "proper" bitdepth, in a perfect world it would output floating point data as the ffmpeg decoder does.
Because the process is lossy, there won't be an "accurate" result, different decoders produce different results (most likely), and most decoders also decode to floating point .. ArcSoft is most likely just cutting/rounding to the bitdepth of the source material.

Unless you mean that its outputting 24-bit samples which only have 16bit filled? (Which i've seen before as well - but i may be able to influence that from the outside actually)

Regarding the "reference proper", over in the eac3to thread, some people compared output to the nero and sonic decoders, and apparently only the newer version of the ArcSoft decoder match the output of those two (which is otherwise identical) - on lossless DTS-HD MA only, of course, comparing bit-exactness of lossy is pointless.

[Portioli]

Dts hd - ma is a codec. As long as we have a non hd audio capable receiver we cannot bitstream.
So our goal is the player ( pc+ audio filter ) to decode this codec and send multi-Chanel PCM to our receiver if it has hdmi ( or analog)
if these PCM tracks have the same sample size and same samplerate why this process is not lossless as long as PCM Is set : wasapi exclusive & PCM output's sampling rate is set same as input.
I am talking only for lossless and not bit perfect.
What's the difference of using wasapi for flac files?

[e-t172]

Resampling with Reclock is not lossless. It is very close to being lossless (so close the difference is probably inaudible) but it is not lossless.

If you want lossless bit-perfect sound you need to disable Reclock's time correction ("slave" option). But, obviously, the downside is you'll probably have some frame drops/repeats if you do that. So it's a compromise. Most people prefer to let Reclock resample because the effects on the audio are negligeable, whereas a frame drop/repeat can be quite noticeable.

Also, according to some people, speeding up and resampling from 23.97 fps to 24 fps is, in fact, the Right Thing to Do.

[dbone1026]

OK, just did a clean reinstall of W7 on my PC. Installed LAV splitter, I can't remember, how do I get into the options for LAV? I have the LAV Splitter set up on MPC HC as my external filter (preferred), just can't remember how I used to get into the LAV settings (i.e. to enable forced subs)

[SamuriHL]

LOL. Run a video, right click go to filters, and you should be able to get to the properties there. :)

[dbone1026]

LOL. Run a video, right click go to filters, and you should be able to get to the properties there. :)

yeah that is what i thought. i will have to try again, my head has been in the clouds of late!

[Andy o]

If upsampling makes audio not "lossless", then the audio is never lossless, because the DACs upsample anyway. You need to define what you're "losing" here. Lossless codecs remain without "loss" until the decoding part. Then people usually have no qualms in messing with it via EQ, DSP, room correction, etc. If you don't then your speakers and room will inevitably mess with it for the worse. I see ReClock as just one more of those processing steps. You're not losing sound quality, so does it matter? You're gaining smooth playback (if you have Intel, I would say ReClock is essential), it's not like you're just doing it for kicks.

[robpdotcom]

yeah that is what i thought. i will have to try again, my head has been in the clouds of late!

You can also double click LAVSplitter in the MPC-HC external filters list.

[SamuriHL]

You can also double click LAVSplitter in the MPC-HC external filters list.

Hey quiet you! I was making him suffer a little. :p :D

[Snowknight26]

One of the most recent versions of the splitter has introduced a large delay when opening files (up to 5 seconds). I ran Process Monitor and saw that the splitter was reading the first 20MB or so of the file (in 4kB chunks!) before a graph was rendered (then started back from byte 0 after it was rendered). Switching to MPC-HC's internal (TS) splitter removed the delay.

The issue seems to be most prevalent with MPEG-TS files.
Process Monitor logs: http://stfcc.org/misc/mpchc_splitters.zip

[nevcairiel]

The underlying IO system should do read-buffering, therefor the read-size is irrelevant. And yes, on many files its actually required to read this much to find all the codec parameters. Stupid MPEG-TS doesn't define them properly.
If it takes 5s to read 20MB on your drive, you should get a faster drive.

Anyway, off to vacationing!

[Gleb Egorych]

"Lossy" codecs do not have a "proper" bitdepth, in a perfect world it would output floating point data as the ffmpeg decoder does.
Because the process is lossy, there won't be an "accurate" result, different decoders produce different results (most likely), and most decoders also decode to floating point .. ArcSoft is most likely just cutting/rounding to the bitdepth of the source material.
I understand that, bitdepth in DTS header is clearly source wavs bitdepth. Anyway generally you need to convert output bitdepth to integer, you can do in decoder or you may rely on windows mixer or ReClock. If it's done in a decoder and done properly it's OK for me. The only question is whether this is done properly in ArcSoft decoder :)

Unless you mean that its outputting 24-bit samples which only have 16bit filled? (Which i've seen before as well - but i may be able to influence that from the outside actually)
I don't know whether high-order bits are zero, I do know that 1.1.0.0 and 1.1.0.1 output different 24 bit streams and it looks suspicious in my opinion.

Regarding the "reference proper", over in the eac3to thread, some people compared output to the nero and sonic decoders, and apparently only the newer version of the ArcSoft decoder match the output of those two (which is otherwise identical) - on lossless DTS-HD MA only, of course, comparing bit-exactness of lossy is pointless.
AFAIK there is DTS-HD StreamPlayer which can be considered as a reference decoder. It's interesting how it decodes lossy and lossless tracks. Unfortunately I don't have it.

About eac3to: people in that thread discussed that MA 7.1 16/48 "strange setup" is not properly decoded with 1.1.0.0, and 1.1.0.8 is needed. Such track was found on "Black Hawk Down" blu-ray. I do have this blu-ray and I reproduced the problem, with 1.1.0.0 there is clearly audible noise. But it's eac3to problem, not ArcSoft! Noise appears only when decoding is done using eac3to, no noise when using 1.1.0.0 through LAV Audio. I dumped outputs of LAV Audio+1.1.0.0 and LAV Audio+1.1.0.8 and they were identical.

[madshi]

About eac3to: people in that thread discussed that MA 7.1 16/48 "strange setup" is not properly decoded with 1.1.0.0, and 1.1.0.8 is needed. Such track was found on "Black Hawk Down" blu-ray. I do have this blu-ray and I reproduced the problem, with 1.1.0.0 there is clearly audible noise. But it's eac3to problem, not ArcSoft! Noise appears only when decoding is done using eac3to, no noise when using 1.1.0.0 through LAV Audio. I dumped outputs of LAV Audio+1.1.0.0 and LAV Audio+1.1.0.8 and they were identical.
Can you upload a sample of the MA 7.1 strange setup track where that noise in audible with eac3to?

[Gleb Egorych]

madshi, sample sent. There is also a problem (http://forum.doom9.org/showthread.php?p=1502158#post1502158) with 6.1 decoding, bad decoding with 1.1.0.8, 1.1.0.0 is needed. Again, it's eac3to specific problem, LAV Audio is OK.

[dbone1026]

You can also double click LAVSplitter in the MPC-HC external filters list.

Ah yeah, that is how I was doing it previously

:thanks:

[madshi]

There is also a problem (http://forum.doom9.org/showthread.php?p=1502158#post1502158) with 6.1 decoding, bad decoding with 1.1.0.8, 1.1.0.0 is needed. Again, it's eac3to specific problem, LAV Audio is OK.
Sample? :)

[Gleb Egorych]

Sample? :)
Sample (http://www.mediafire.com/?jcie332d4231g8r). The problem appears with DTS-ES 6.1 and DTS-MA 6.1.

EDIT: sample link fixed, sorry.

[madshi]

Sample (http://forum.doom9.org/showthread.php?p=1502158#post1502158). The problem appears with DTS-ES 6.1 and DTS-MA 6.1.
There is no sample at that link.

[Snowknight26]

The underlying IO system should do read-buffering, therefor the read-size is irrelevant. And yes, on many files its actually required to read this much to find all the codec parameters. Stupid MPEG-TS doesn't define them properly.
If it takes 5s to read 20MB on your drive, you should get a faster drive.

The file was fully cached in RAM so no actual reading was done from the HDD.

I also found it peculiar that before the graph is rendered, the read-size is 4kB, but after, it's 64kB.

I'll try to get you a better analysis to show you that there is actaully a problem. :p

[CruNcher]

Snowknight26 i reported this some posts back very carefully and indepth and i found even more problems, though most of these stability/performance issues don't really show up in a Playback environment (major goal of lav splitter) but become very apparent when doing Dshow based transcoding with lav splitter (so blocking lav splitter outside of a Player environment is advised for the moment) as the source filter especially for .mpg and .ts so far ;)
And yeah MPCs Splitter doesn't suffer from this, funny though Cyberlinks splitter shares the same behaviour with lav splitter hehe ;)

http://forum.doom9.org/showpost.php?p=1502105&postcount=3261
http://forum.doom9.org/showpost.php?p=1502471&postcount=3309

[Gleb Egorych]

I added in my past post (http://forum.doom9.org/showthread.php?p=1508578#post1508578) information about 2.1 bug.

Also I ran eac3to with all 1.1.0.x versions of arcsoft decoder on lossy DTS files. As I understand eac3to patches streams to mark it 24 bit and to make all arcsoft versions output 24 bit. In this case outputs of all versions except 1.1.0.0 are identical.

So the final conclusion is following: the best version of dtsdecoderdll.dll is 1.1.0.1. It decodes all lossy streams to 24 bit, it doesn't have mono and 6.0 bugs, its decoding algorithm for lossy and lossless content wasn't changed in latter versions (except mono and 6.0 of course). The only problem of 1.1.0.1 (and all arcsoft versions) - silent LFE channel in 2.1 streams.

PS Version 1.1.0.1 came with the original release of TMT3, 3.0.1.120.

[Andy o]

thanks for the extensive testing!

[skingery]

I added in my past post (http://forum.doom9.org/showthread.php?p=1508578#post1508578) information about 2.1 bug.

Also I ran eac3to with all 1.1.0.x versions of arcsoft decoder on lossy DTS files. As I understand eac3to patches streams to mark it 24 bit and to make all arcsoft versions output 24 bit. In this case outputs of all versions except 1.1.0.0 are identical.

So the final conclusion is following: the best version of dtsdecoderdll.dll is 1.1.0.1. It decodes all lossy streams to 24 bit, it doesn't have mono and 6.0 bugs, its decoding algorithm for lossy and lossless content wasn't changed in latter versions (except mono and 6.0 of course). The only problem of 1.1.0.1 (and all arcsoft versions) - silent LFE channel in 2.1 streams.

PS Version 1.1.0.1 came with the original release of TMT3, 3.0.1.120.

I read yesterday that that Arcsoft will soon have a beta update to TMT5. Interesting to see if this changes this dll.

AndyVT has the details about the update over on Missing Remote http://www.missingremote.com/blog/arcsoft-totalmedia-theatre-5-update

[Thunderbolt8]

so for mono and 6.0 its still best to use 1.1.0.0 ?

[zipi]

Is there any reason why LAV Audio Decoder outputs 32 bit float on MP3s (or at all for that matter when connected to a real hardware audio renderer) ?

how can one disable floating point output ?

[CiNcH]

I am using 'ffdshow DXVA Video Decoder' to decode VC-1. It seems that for VC-1 within m2ts, VC-1 timestamp correction has to be enabled, and for VC-1 within MKV, it has to be disabled. Shouldn't this option only apply to m2ts?

[mindbomb]

does it have to be disabled for vc-1 in an mkv? that hasnt been my experience with other dxva filters.

[nevcairiel]

I am using 'ffdshow DXVA Video Decoder' to decode VC-1. It seems that for VC-1 within m2ts, VC-1 timestamp correction has to be enabled, and for VC-1 within MKV, it has to be disabled. Shouldn't this option only apply to m2ts?

VC-1 in MKV has terrible timestamps, and ffdshow DXVA can't deal with it properly, apparently.

[Andy o]

Is there any reason why LAV Audio Decoder outputs 32 bit float on MP3s (or at all for that matter when connected to a real hardware audio renderer) ?

how can one disable floating point output ?

I don't think you can, but what do you mean by "real hardware audio renderer"? Example?

[CiNcH]

VC-1 in MKV has terrible timestamps, and ffdshow DXVA can't deal with it properly, apparently.
It can. As long as 'VC-1 timestamp correction' is disabled. But then m2ts playback is choppy which requires the correction. If 'VC-1 timestamp correction' is enabled, m2ts playback is smooth, but MKV is choppy. Seems as if frames are being swapped in both "error" cases or something like that.

does it have to be disabled for vc-1 in an mkv?
Yes. At least when using 'ffdshow DXVA Video Decoder'. Maybe not with others. Maybe it also depends on the muxer. I sometimes use MakeMKV to place a Blu-Ray on my NAS.

Short summary:
m2ts -> LAV -> ffdshow DXVA Video Decoder: VC-1 timestamp correction has to be enabled, otherwise frames will be swapped
mkv -> LAV -> ffdshow DXVA Video Decoder: VC-1 timestamp correction has to be disabled, otherwise frames will be swapped

I thought that the correction is only necessary for m2ts, depending on the decoder that is being used (e.g. ffdshow vs. CyberLink).

[RealSnoopyDog]

I am using 'ffdshow DXVA Video Decoder' to decode VC-1. It seems that for VC-1 within m2ts, VC-1 timestamp correction has to be enabled, and for VC-1 within MKV, it has to be disabled.
Confirm, i have the same problem.

One of the most recent versions of the splitter has introduced a large delay when opening files (up to 5 seconds). I ran Process Monitor and saw that the splitter was reading the first 20MB or so of the file (in 4kB chunks!) before a graph was rendered (then started back from byte 0 after it was rendered).I also see this problem and i think, the read buffer is too small. I regulary run into problems when i watch movies from slower data sources, e.g. a "real" Blu-Ray in a BD-ROM. From one sudden to the next, LAV starts accessing the disc like hell and playback starts stuttering. This happens incidentially, it may happen, but must not. When i use a different splitter with the same Direct Show filters, the problem does not occur. Unfortunately, i don't have Visual Studio 2010 atm to compile the filter, only 2008. If i could compile it, i would play with BD_READ_BUFFER_SIZE a bit and see, what happens.

The underlying IO system should do read-buffering, therefor the read-size is irrelevant.After long year of expierience, i would not rely on this. :p

But thanks again for this software, it's getting better and better :)

[ryrynz]

I have a video which has some portions of the MP3 audio cut out on my Yamaha receiver when Auto AV Sync correction is enabled.
The audio is sent bitstreamed over HDMI, let me know if you'd like a sample.

[ney2x]

@nevcairiel

Just a quick question. Is there a plan for AC3 (SPDIF Encode mode) like the ffdshow? A friend of mine ask me...

Thanks.

[nevcairiel]

Just a quick question. Is there a plan for AC3 (SPDIF Encode mode) like the ffdshow? A friend of mine ask me...

No, its an audio decoder, not an encoder.

[nevcairiel]

It can. As long as 'VC-1 timestamp correction' is disabled. But then m2ts playback is choppy which requires the correction. If 'VC-1 timestamp correction' is enabled, m2ts playback is smooth, but MKV is choppy. Seems as if frames are being swapped in both "error" cases or something like that.

I'll add ffdshow DXVA to the list of broken codecs then, which will cause the "auto" setting to enable it for m2ts and disable for MKV.

[Andy o]

@nevcairiel

Just a quick question. Is there a plan for AC3 (SPDIF Encode mode) like the ffdshow? A friend of mine ask me...


You can do that with ReClock, or (I think) with ffdshow as audio processor, both of which will still let you use LAV as a decoder.

[zipi]

Is there any reason why LAV Audio Decoder outputs 32 bit float on MP3s (or at all for that matter when connected to a real hardware audio renderer) ?

how can one disable floating point output ?

I don't think you can, but what do you mean by "real hardware audio renderer"? Example?

any real soundcard audio renderer, not an intermediate filter/processor.
With many pro cards,if you feed them 32bit float, you'll get a distortion storm guaranteed to kill your gear & hearing.

nevcairiel?

[nevcairiel]

Your so called "pro cards" should get a driver then that does either not accept float input, or just converts it to integer before sending it over the DACs.

Forcing a specific output format may be possible one day, but since it degrades quality, its not on the top of the list.

[zipi]

I cannot say I am too worried (or understand your logic tbh) about "losing quality", on mp3 decoding, by not using 32 bit float output for mp3s.
Not to mention I have not seen LAV Audio decoder do it with Wavpack for example, using the same cards.

btw about the "so called" part of your reply :):
I actually stumbled upon this problem by accident setting up a system with LAV installed,
As I build audio oriented systems & dealing with a lot of professional cards, I thought it is worth reporting it to you, so whether it gets fixed or not, I have zero interest in it either way ...

Thanks for replying, all the best.

[Monamona]

This is all in the hands of ffmpeg. There have however been some fixes coming in since the last time i updated ffmpeg, and i'll be merging those fixes for the next version.

Do you have a sample file which shows the problem, so i can see if things improved?

I uploaded a sample as follows.

https://rapidshare.com/files/2471309460/sample.mp4
(200MB, sorry...)

I put chapters at the beginning of scenes (like opening main-part...) so that you can see gaps of 2 frames.

[Portioli]

sorry if this is offtopic but i would like to make a question about this hd audio discussion.

as long as lav audio can decode Dolby TrueHD (and DTS-HD MA with the TMT5 trick)
is it possible to send hd audio uncompressed via hdmi ( multi-channel PCM ) to old AV Receivers that do not support HBR decoding (dts-hd ma + DD TrueHD) ?

are the audio drivers/card drivers or the hdmi cable of a pc, capable to send 6 or 8 channels of losseless audio to the receiver?

Are analog outputs the only solution?

thanks in advance

[Andy o]

any real soundcard audio renderer, not an intermediate filter/processor.
With many pro cards,if you feed them 32bit float, you'll get a distortion storm guaranteed to kill your gear & hearing.

nevcairiel?

Why must you do it this way? It can't be latency (for movies/music it shouldn't be an issue). Isn't it easier to just use a proper renderer to output 24-bit or whatever to your card?

It's what LAV audio would do, btw, if you could "force" it to output integer from lossy sources (correct me if I'm wrong, Nev).

[Andy o]

as long as lav audio can decode Dolby TrueHD (and DTS-HD MA with the TMT5 trick)
is it possible to send hd audio uncompressed via hdmi ( multi-channel PCM ) to old AV Receivers that do not support HBR decoding (dts-hd ma + DD TrueHD) ?
Sure, as long as you have a full HDMI audio device. ATI/AMD 4000 and newer, Intel G35, G45 and Arrandale/Clarkdale and newer (with proper mobo support), Nvidia 460GTX and lower 400 series, and their 500 series counterparts. Some older Nvidia integrated chipsets too, like the 9300, 9400, 8200, 8300.

[Portioli]

Thanks for this Andy

[Blight]

Hi,
I have an MOV file which appears to have PCM (uncompressed) audio (download here : http://games.on.net/file/30341/Borderlands_-_Australian_launch_trailer).
When using the 'LAV Source Splitter' filter, it exposes two sub-types:
{34616D69-0000-0010-8000-00AA00389B71} // QDesign Music 2
{AFBC2343-3DCB-4047-9655-E1E62A61B1C5} // E-AC3

Since neither is the actual format, any audio decoder other than the 'LAV Audio Decoder' fails to connect.

[nevcairiel]

The first type is called "IMA4". Its an ADPCM codec, and ffdshow should support it.
The second type is not E-AC3, its a LAV Splitter internal type. Someone mislabeld it in some ZP profile.

[mark0077]

nev, sorry if it was mentioned in an earlier thread but should I be able to seek in mkv's with the current lav splitter. I have alot of mkvs I created from DVDs using MakeMkv in the past.

I can seek using the internal mpc-hc or haali mkv splitter but with lav splitter, when I seek to a certain location, the splitter jumps to the previous / next chapter, rather than the specific times I want to seek to.

[nevcairiel]

I can seek just fine in any MKVs i have.
Try turning off "Fast seek(on key-frame)" in the MPC-HC "Tweaks" options, maybe it helps your case, its sometimes somewhat buggy (MPC-HCs fault)

[JarrettH]

Can you tell me what is more correct for my setup?
http://www.emu.com/products/product.asp?product=10447

It seems LAV Audio outputs most audio streams in 32-bit, so I'll begin with that...

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 24-bit Output (this matches the EMU 0404 24-bit output rate)

OR

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 32-bit Float Output (EMU 0404 converts this to 24-bit output)

I think the first scenario makes more sense, do you? :thanks:

[Mark_A_W]

Can you tell me what is more correct for my setup?
http://www.emu.com/products/product.asp?product=10447

It seems LAV Audio outputs most audio streams in 32-bit, so I'll begin with that...

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 24-bit Output (this matches the EMU 0404 24-bit output rate)

OR

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 32-bit Float Output (EMU 0404 converts this to 24-bit output)

I think the first scenario makes more sense, do you? :thanks:

If both work, then there should be no difference.

[PeQuE]

Hi nev...

I know you are on vacation but... Could you think about the possibility of adding audio delay feature to Lav Audio Decoder? I only miss this to finally get rid of ffdshow... :(

Thanks a lot for considering...

[mindbomb]

Can you tell me what is more correct for my setup?
http://www.emu.com/products/product.asp?product=10447

It seems LAV Audio outputs most audio streams in 32-bit, so I'll begin with that...

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 24-bit Output (this matches the EMU 0404 24-bit output rate)

OR

LAV Audio Output 32-bit --> PCM 32-bit --> AC3 Filter to downmix to 2.0 and using PCM 32-bit Float Output (EMU 0404 converts this to 24-bit output)

I think the first scenario makes more sense, do you? :thanks:


heh, i went through this type of thing before myself.

Firstly, you should use ffdshow instead of ac3 filter, iirc, ac3 filter freaks out when you play things with more than 6 channels.

second, it is better to downmix in 32 float.

[CiNcH]

I'll add ffdshow DXVA to the list of broken codecs then, which will cause the "auto" setting to enable it for m2ts and disable for MKV.

OK.

Just a small addition. Haali + ffdshow DXVA is also fine for both, VC-1 within m2ts and MKV. I don't believe that it checks the downstream filter.

[nevcairiel]

Maybe it always does it for m2ts and never for MKV, only way to confirm is to run it through some other (commercial) decoders which require the opposite settings in LAV Splitter.

[JarrettH]

heh, i went through this type of thing before myself.

Firstly, you should use ffdshow instead of ac3 filter, iirc, ac3 filter freaks out when you play things with more than 6 channels.

second, it is better to downmix in 32 float.

Thanks, actually I have ffdshow, but I always preferred the downmixing in AC3 filter. I'll give ffdshow another try though, I'd love to get rid of that additional application!

Is the Processing tab in ffdshow the input?

[Dogway]

FYI after 2 months I discovered how to demux the file I was having problems with in this post (http://forum.doom9.org/showthread.php?p=1491561#post1491561), avc+aac in a QT muxed ".mov". Mov extension doesn't show up in lavsplitter input formats, so simply rename to mp4 and everything will work as expected. Its a nice workaround until nev gives support :cool:
lavsplitter through graphstudio could automatically detect it as mp4, but through mpc the video screen was just messed freeze so I guess it detected as mov

[mark0077]

I can seek just fine in any MKVs i have.
Try turning off "Fast seek(on key-frame)" in the MPC-HC "Tweaks" options, maybe it helps your case, its sometimes somewhat buggy (MPC-HCs fault)

Thanks nev, that seemed to help yes. However seeking large distances in my mkv files with lav splitter has some issues.

If I seek for example 1 hour forward in an mkv, the image just seems to stop / freeze, and the audio continues in the background. Sometimes if I seek say about 10 minutes forward, I see like about 1000 frames flash past over the space of a second or two, then I'm at my desired location.

Does the splitter actually still send all of the frames from my start to finish location to the decoder, it seems that way at present. I don't get instantly moved to the new location like with haali for example. Am I still hitting more mpc-hc problems or could it be lav splitter?

Try opening a largish, 2hr mkv and seek to almost the end to see if you can reproduce, if you'd like? Definitely something not right, let me know if you need any more info if you're interested in trying to reproduce etc.

EDIT: Actually when I disable the lav cuvid decoder, I can't reproduce any of the issues above, I can seek almost instantly in these files. Maybe its lav cuvid thats causing the issue? Let me know if I should repost over there instead... I have lavcuvid set to decode h264, vc-1, mpeg2, deinterlacing set to None, 50p/60p, Field Order: Auto, Settings are "Use stream aspect ratio" on and "Use HQ DXVA processing" off, with Output Format: YV12

Using GTX 295, Windows 7 64bit, mpc-hc 32bit with lavcuvid and lavsplitter/lav audio.

[nevcairiel]

It doesn't seen all frames, of course, but seeking is sometimes a bit inaccurate so that it ends up at a few seconds earlier (usually never later), and those frames are delivered to the downstream filters completely, causing those flashing frames in LAV CUVID, because it doesn't discard them like other decoders. I should fix that in LAV CUVID, as well as improve seeking accuracy in some file formats.

On your original issue however, your MKVs seem weird. Usually a MKV should have a Cue Point every few seconds, those cue points are used for the "fast seeking" in MPC-HC, if MakeMKV only puts them on chapters, i consider that a bug. Sadly MPC-HC is rather stupid, and it always seeks to cue points when that option is one, even if its time is many seconds or even minutes away.

[mark0077]

ah okie, no problemoz

[RealSnoopyDog]

Hi nevcairiel, i started compiling your splitter and got to the point where i need to compile ffmpeg. This not so easy (compiling ffdshow is more easy) but anyway. I use MinGW and got it working, the only thing is: what parameters should i use for the "configure" script of ffmpeg. I found somewhere that i should use the options "--enable-memalign-hack" if the destination is Windows. So i use "--enable-memalign-hack --enable-shared --cpu=x86". Is this what you also use? You know, MinGW runs sooooo sloooowly..... I get grey hair when i trial-and-error this ;)

[nevcairiel]

There are build scripts in the main source directory.

[Blight]

The first type is called "IMA4". Its an ADPCM codec, and ffdshow should support it.
The second type is not E-AC3, its a LAV Splitter internal type. Someone mislabeld it in some ZP profile.

Using 'ffdshow_rev3892_20110620_clsid' build from 2 days ago, ffdshow isn't able to connect to LAV Splitter for this audio stream (see attached screenshot).

Edit:
1. Is there an official sub-type for E-AC3?
2. Reading the changelog, there's indication of a programmable interface. Please include the headers in the ZIP/Installer.

[HTPC-User]

Hi nevcairiel!

In MPC-HC one can switch to the next audio language and subtitle track by pressing a keyboard shortcut.
Is this also possible with your splitter? If it is possible, what do I have to do?

:thanks:

[hoborg]

Hi nevcairiel!

In MPC-HC one can switch to the next audio language and subtitle track by pressing a keyboard shortcut.
Is this also possible with your splitter? If it is possible, what do I have to do?

:thanks:

You can do this if you use FFDshow video decoder.
Very useful for control audio/subtitles streams switching, turn on/off subtitles, etc... by remote controler in all DS based players.

[Gleb Egorych]

Edit:
1. Is there an official sub-type for E-AC3?
Look here (http://bugs.zoomplayer.com/bug.php?op=show&bugid=91) and here (http://msdn.microsoft.com/en-us/library/dd317599(v=vs.85).aspx).

[RealSnoopyDog]

There are build scripts in the main source directory.

Uups, sorry, it was a bit late that day and i didn't recognize that there is a script. But now i was able to compile LAV - even with Visual Studio 2010 express. Thank you :)

[RealSnoopyDog]

Usually a MKV should have a Cue Point every few seconds, those cue points are used for the "fast seeking" in MPC-HC, if MakeMKV only puts them on chapters, i consider that a bug.

There seems to be a bug in the latest version:
http://www.makemkv.com/forum2/viewtopic.php?f=1&t=3634

One more question related to DTS Express: As far as i understand, DTS express is used for secondary audio tracks which are not designated to be played back standalone. They should mixed over the currently playing primary audio tracks. I found some newer Blu-Rays in my collection which have a DTS express secondary audio stream. You think it's hard to implement a mixing feature like this:When these new features are streamed in, your player will actually mix the audio portion of the “new” content with the audio already on the disc. The new audio is called Secondary Audio on Blu-ray Disc and can be up to 5.1 channels; on HD DVD this is called Sub Audio and can be in 2.0 channels. DTS uses a system called DTS Express to deliver Secondary or Sub Audio to your player. The DTS Express audio is mixed with the Primary Audio, giving you an intensely great interactive surround experience.

To ensure that the audio mix comes out perfectly, DTS-HD employs an innovative feature called Dynamic Mixing. When the movie or concert creators encode the Secondary or Sub audio, they include meta-data that will dynamically mix that audio with the primary audio. With Dynamic Mixing, you’ll continually have the correct sound balance — just as the producers intended.
Reference: http://www.dts.com/DTS_Audio_Formats/DTS-HD_Master_Audio/How_Does_DTS-HD_Work.aspx

But there are not many discs which come with this feature so i think the effort for implementing this is too high (?) and can be used better for other improvements. Or is the Arcsoft DLL already capable of this? For this reason, secondary audio streams should not be listed in the normal IAMStreamSelect togehter with the primary streams, once a mixing feature is available.

It's also working with DVBViewer (see screenshot) where i played back one of my blu-rays. But i manually re-create the contents of the audio menu and i have to append the secondary audio track to the end of the menu.

[Blight]

Gleb:
Thanks, I've updated the profile and closed the bug report :)

[RealSnoopyDog]

For this reason, secondary audio streams should not be listed in the normal IAMStreamSelect togehter with the primary streams, once a mixing feature is available.
OK, not a good idea from my side. It's good as it is right now. If the user selects a secondary audio stream from within IAMStreamSelect then the secondary stream can be played back standalone (as it is right now) or mixed together with the current primary stream (possible future). To go back to the previous state, just select the previous active stream.

I just changed the DVBViewer audio menu: If there are subaudio streams available, then it shows a menu entry at the bottom which is labeled "Secondary audio tracks...". This entry points to a sub menu where you can either select the secondary audio track(s) or "None" which switches back to the previous active audio track (see screenshot). I got it working like this with DVBViewer + LAV.

[nevcairiel]

Not all secondary streams are really required to be mixed onto the main stream, most of the time they contain the commentary for PiP streams - which you can mix, but the original audio would fade out significantly anyway, so can as well play it standalone.

[RealSnoopyDog]

You're right: Pip is secondary video which is mixed togehter with the primary video. The same with secondary audio. I found two Blu-Rays right now with secondary audio. In both cases the secondary audio track is silent most of the time because you should hear the original soundtrack with the lanuage that you selected at that time. Is it hard to realize some kind of mixing with DirectShow? You'll need another audio output pin (?)

[simpleway]

I tested it for mkv with real video (RV40), this video pin can't connect to grabest real video decorder

[HTPC-User]

You can do this if you use FFDshow video decoder.
Very useful for control audio/subtitles streams switching, turn on/off subtitles, etc... by remote controler in all DS based players.

Hi!

I am not quite sure if I can use this in my planned scenario: MPC-HC with ffdshow (for deinterlacing and avisynth scripts), LAV Filters/splitter and madVR.
In order to see subtitles I have to use the MPC-HC subtitle engine with madVR. (As far as I know ffdshow subtitles aren't supported by madVR).

So I still wonder if it is possible to switch to the next subtitle and audio track by pressing a key (and using Nevcairiel's cool splitter).

But perhaps I only misunderstood your suggestion.

:thanks:

[fastplayer]

(As far as I know ffdshow subtitles aren't supported by madVR).
They are supported. Always have been.

[hoborg]

Hi!

I am not quite sure if I can use this in my planned scenario: MPC-HC with ffdshow (for deinterlacing and avisynth scripts), LAV Filters/splitter and madVR.
In order to see subtitles I have to use the MPC-HC subtitle engine with madVR. (As far as I know ffdshow subtitles aren't supported by madVR).

So I still wonder if it is possible to switch to the next subtitle and audio track by pressing a key (and using Nevcairiel's cool splitter).

But perhaps I only misunderstood your suggestion.

:thanks:

FFDshow put subtitles inside the video, so they will apear in any renderer.
It is good to renable resize in ffdshow too to have the same subtitles undepended on video size.

[HTPC-User]

Hi!

FFDshow put subtitles inside the video, so they will apear in any renderer.
It is good to renable resize in ffdshow too to have the same subtitles undepended on video size.

Ok, I was wrong. I tried the ffdshow subs with EVR CP in MPC-HC and also the subtitles didn't work. So I think there is some kind of configuration problem on my side.
By the way I followed for example this tutorial http://www.homecinema-hd.com/ffdshow-video-subtitles_en.html but no subtitles. Any specific setting to be aware of? (Autoload subs in MPC-HC playback is off, subtitles in ffdshow are on, all transform filters in MPC are unchecked)

Anyway, as stated in several postings and threads the MPC-HC subtitle engine seems to be the better choice for subs than ffdshow and therefore I would like to use it.

That leads us to my original question :): Is it somehow possible to switch to the next audio and subtitle language by pressing a keyboard shortcut (and using MPC-HC's subtitle engine, LAVFilter/splitter, ffdshow for deinterlacing and scripts, madVR etc.)?

:thanks:

[edison]

Sometimes, the LAV Splitter detected some 59.94p mkv files as 59.92p.

http://www.gokuai.com/w/1bsbF0G29e4g4A0n/lav_59.940_59.917fps.jpg

and sometimes detected as 60.000p
http://www.gokuai.com/w/5L0mjw4zA3U1gRm0/lav_59.940_60.000fps.jpg

[Selur]

Trying to play a mp4 with multiple test subtitles (http://www.multiupload.com/4VDVDSAZ5B) in MPC-HC using LAV Filters, LAV CUVID and EVR CP.
Video works fine, but the subs don't. (they work fine if I use MPC-HC internal mp4 splitter)
-> is it a bug or just me overlooking something?

Cu Selur

[adam777]

Hello all,
I've read what seems relevant in the thread, but still could not understand the exact details regarding DTS-HD decoding, so if someone could make it simple for me it would be great.
I'm using LAV Splitter under Windows 7 x64 and my system's speaker setup is 5.1.
Also, I'm using ReClock's WASAPI exclusive as audio renderer.
What should I expect under the following:

1. Decoding a 7.1 DTS-HD file using LAV Audio (decoding quality (full? core?), output configuration).

2. Decoding a 7.1 DTS-HD file using Arcsoft decoder (the dll, same parameters).

3. Decoding a 5.1 DTS-HD file using LAV Audio (same).

3. Decoding a 5.1 DTS-HD file using Arcsoft decoder (same).

:thanks:
Adam

[nevcairiel]

LAV Audio alone (using ffmpeg) can only decode the Core, and if with "ArcSoft decoder" you mean using dtsdecoderdll.dll in LAV Audio then the following applies:

So -

1. 5.1 Core
2. Full 7.1 HD
3. 5.1 Core, same as 1)
4. Full 5.1 HD

[adam777]

LAV Audio alone (using ffmpeg) can only decode the Core, and if with "ArcSoft decoder" you mean using dtsdecoderdll.dll in LAV Audio then the following applies:

So -

1. 5.1 Core
2. Full 7.1 HD
3. 5.1 Core, same as 1)
4. Full 5.1 HD

OK, great.
Is there a combination that allows me to get something like full HD 5.1 in the second case?
Meaning, some sort of downmixing done by the arcsoft decoder that converts the 7.1 audio to 5.1, leaving it 24 bit etc.?

[nevcairiel]

No. But since its already decoded to PCM, you can use something like ffdshow for downmixing, without downsamplig.

LAV Audio will support (simple) downmixing eventually, but i don't know when i'll work on that.

[adam777]

Thanks again, it all makes sense now.

Oh, forgot to ask, what is the "correct" way of downmixing? extracting the core from the original stream or decoding at full resolution and let FFDShow handle the downmixing?

* EDIT *
I'll be a bit more specific.
Which way is generally considered "better" - utilizing the built-in 5.1 DTS core, thus receiving perfect sound matrix, but losing the details available in the MA stream.
Or decoding the stream in full resolution, maintaining all details, but losing (?) the sound mix due to the external downmixing.

On a side note, I think I've read something regarding an option to output the 7.1 MA stream at both 7.1,6.1 and 5.1 in a lossless fashion natively, which is obviously ideal.

[nevcairiel]

The ArcSoft decoder offers downmixing functionality, however it is unknow if it does it "properly", that is using the 6 channel downmix matrix in the HD stream (if present), or just does a stupid downmix like ffdshow would. In any case, its currently not possible to use it.

[adam777]

Well, you got me on "stupid", I'll stay on good old DTS core for the time being, most likely my equipment (both electronic and ears) isn't enough to really tell the difference anyway...

[Andy o]

Thanks again, it all makes sense now.

Oh, forgot to ask, what is the "correct" way of downmixing? extracting the core from the original stream or decoding at full resolution and let FFDShow handle the downmixing?
Is your receiver 7.1 capable? Some receivers (Pioneers for instance) have some sort of virtual downmixing, in which the 5.1 surrounds will try to virtualize the rears. This can work pretty well, it would be my preferred method. You would have to run your PC as if you had 7.1 speakers though, and let the receiver downmix everything. I'm not sure how ffdshow does it, though, or the Arcsoft decoder for that matter.

[SamuriHL]

Is your receiver 7.1 capable? Some receivers (Pioneers for instance) have some sort of virtual downmixing, in which the 5.1 surrounds will try to virtualize the rears. This can work pretty well, it would be my preferred method. You would have to run your PC as if you had 7.1 speakers though, and let the receiver downmix everything. I'm not sure how ffdshow does it, though, or the Arcsoft decoder for that matter.

That's what I do on my HTPC.

[adam777]

Well, that is where "most likely my equipment (both electronic and ears) isn't enough to really tell the difference anyway" comes into play.
At the moment it's just a standard notebook audio card connected to a fairly simple Logitech X-530, hence I doubt I'll be able to tell the quality difference anyway.
Will possible get an HTPC "soon", so it was mostly out of curiosity...

[dbone1026]

Finally posted a quick writeup on LAV Filters just talking about basic setup. Thought I would post here if anyone found this helpful or if there were any comments/suggestions on something I missed or should have added:

http://www.mediasmartserver.net/2011/06/30/guide-setting-up-lav-filters-mpc-hc-for-all-your-splitter-and-audio-needs/

[Dogway]

Besides .mov containers stated in this post (http://forum.doom9.org/showthread.php?p=1491545#post1491545) I also couldn't split a transport stream container, .ts file with mpeg2+mp2 streams. I could hand you a sample if I knew how to split these kind of files.

[TheLion]

I have a question regarding the auto A/V sync feature of LAVAudio. I recently switched from madFlac (using eac3to to rip all my Blu-Rays to mkv, encoding all audio to flac) to using LAVaudio for all streams - especially DTS MA tracks with Arcsoft 1.1.0.8 as decoder.

I avoid using reclock and/or the "video clock" feature of JRiver MC because I don't want my audio to be resampled - not even so very slightly. In general those algorithms use the video clock as master - and resample audio accordingly.

I want bit perfect audio decoding with as little jitter as possible (I am using a FIR convolution VST plugin which is very sensible to jitter in the decoded stream). I don't want the audio stream to be resampled in any way.

How is the LAVAudio auto A/V sync working in that regard? (My display works at pretty exactly 23.976, madVR shows a few dropped frames every hour or so)

I ask because switching auto A/V on off results in slightly different sound "quality". I am not sure if the signal gets resampled (=degraded) or if jitter gets reduced (=better quality).

Thank you!

[nevcairiel]

LAV Audio always trys to create smooth timestamps of all audio data being send to the renderer, basically by completely recalculating all audio timestamps based on the duration of the decoded PCM samples.

The A/V sync options whole purpose is to compare the calculated timestamps to the timestamps coming from the source filter, and making sure that there are no constant differences between them (ignoring jitter and other sorts of things).
Most of the time, this is only an issue if your file is broken and has gaps in the audio. On "normal" files, this option should not do anything.

More importantly, it does not do anything to the audio itself, it just corrects possible timing calculation mistakes (which really shouldn't happen), or compensates for gaps in the audio stream.
It does not do anything based on the FPS of the movie, or the refresh rate, or any other external factors. On a "good" file, it does not do anything - its just a safety net for broken files (and live streams, i suppose)

[TheLion]

Thank ypu very much for this reply. It gives me comfort to just leave it enabled.

I understand that with LAV you do anything to deliver an audio stream with as little Jitter as possible to the audio engine (in my case JRiver's 64bit)?! Is thee anything you can think of that would improve it even further?

What happens if your auto sync algorithm detects a sync problem? How does it get the streams back to in-sync playback? By repeating/skipping audio samples?

I hope you are aware that you and Mathias are the greatest assets in the HTPC community atm ;-)

Thank you!

[TheLion]

One more question: One of the great features of my previous eac3to/madFlac setup was to be able to remove all dialog normalization/DRC annoyances from audio streams before re-encoding them.

There are quite many audio tracks with DN/DRC enabled by default. A pretty infamous one is the Blu-Ray TrueHD track of Iron Man. It defaults to DRC enabled. When played back and the decoder has no means to manually override the DRC setting all dynamics are gone...

When I don't "enable" DRC in the LAV audio decoder - does that mean that all tracks all cleansed from Dialog Normalization/DRC? (Just like eac3to does it)

[nevcairiel]

What happens if your auto sync algorithm detects a sync problem? How does it get the streams back to in-sync playback? By repeating/skipping audio samples?

It just adapts the timestamps to match, which will cause in a short gap in the audio, or a drop of a few samples. Nothing will be repeated.



When I don't "enable" DRC in the LAV audio decoder - does that mean that all tracks all cleansed from Dialog Normalization/DRC? (Just like eac3to does it)
That depends on the decoder in ffmpeg, i just tell it to not apply DRC, but i'm not sure if the TrueHD decoder acts on that info. eac3to might be doing something special.

[Mercury_22]

Can't build latest (Nevcairiel-LAVFSplitter-f06c75b libbluray-3fa955d ffmpeg-177f36c) lav = it's taking forever to compile and I've stopped it here Setting environment for using Microsoft Visual Studio 2010 x86 tools.
Makefile:2: config.mak: No such file or directory
Makefile:42: /common.mak: No such file or directory
Makefile:75: /libavutil/Makefile: No such file or directory
Makefile:153: /doc/Makefile: No such file or directory
Makefile:154: /tests/Makefile: No such file or directory
make: *** No rule to make target `/tests/Makefile'. Stop.
install prefix /usr/local
source path .
C compiler gcc
ARCH x86 (generic)
big-endian no
runtime cpu detection yes
yasm yes
MMX enabled yes
MMX2 enabled yes
3DNow! enabled yes
3DNow! extended enabled yes
SSE enabled yes
SSSE3 enabled yes
AVX enabled yes
CMOV enabled no
CMOV is fast no
EBX available yes
EBP available no
debug symbols no
strip symbols yes
optimize for size no
optimizations yes
static no
shared yes
postprocessing support no
new filter support no
network support no
threading support w32threads
SDL support no
Sun medialib support no
libdxva2 enabled no
libva enabled no
libvdpau enabled no
AVISynth enabled no
libcelt enabled no
frei0r enabled no
libdc1394 support no
libdirac enabled no
libfaac enabled no
libgsm enabled no
libmp3lame enabled no
libnut enabled no
libopencore-amrnb support no
libopencore-amrwb support no
libopencv support no
libopenjpeg enabled no
librtmp enabled no
libschroedinger enabled no
libspeex enabled no
libtheora enabled no
libvo-aacenc support no
libvo-amrwbenc support no
libvorbis enabled no
libvpx enabled no
libx264 enabled no
libxavs enabled no
libxvid enabled no
openal enabled no
zlib enabled yes
bzlib enabled yes

Enabled decoders:
aac mp3adufloat
aac_latm mp3floatdio
aasc mp3on4 dsicinvideo
ac3 mp3on4float
adpcm_4xm dvdsub mpc7
adpcm_adx dvvideo mpc8
adpcm_ct dxa mpeg1video
adpcm_ea eac3 mpeg2video
adpcm_eaeacmvs_xa mpeg4
adpcm_ea_r1 eamad mpegvideo
adpcm_ea_r2 eatgq msmpeg4v1
adpcm_ea_r3 eatgv msmpeg4v2
adpcm_ea_xas eatqi msmpeg4v3
adpcm_g722 msrlebps
adpcm_g726 msvideo1_exp
adpcm_ima_amv mszhtsvx_fib
adpcm_ima_dk3 mxpegsvx_raw
adpcm_ima_dk4 nellymoser
adpcm_imffv1_eacs nuv
adpcm_imffvhuffad pam
adpcm_ima_iss flac pbm
adpcm_ima_qt flashsv pcm_alaw
adpcm_imflicjpeg pcm_bluray
adpcm_ima_wav flv pcm_dvd
adpcm_ima_ws fourxm pcm_f32be
adpcm_ms fraps pcm_f32le
adpcm_sbpro_2 frwu pcm_f64be
adpcm_sbpro_3 gif pcm_f64le
adpcm_sbpro_4 gsm pcm_lxf
adpcm_swf gsm_ms pcm_mulaw
adpcm_thp h261 pcm_s16be
adpcm_xa h263 pcm_s16le
adpcm_yamaha h263i pcm_s16le_planar
alac pcm_s24be
als pcm_s24daud
amrnb pcm_s24le
amrwb pcm_s32beff_byterun1
amv pcm_s32leff_ilbm
anm pcm_s8
ansi pcm_u16be
ape pcm_u16le
ass pcm_u24be
asv1 pcm_u24lenterplay_dpcm
asv2 pcm_u32benterplay_video
atrac1 pcm_u32lepeg2000
atrac3 pcm_u8
aura pcm_zork
aura2 pcx1
avs pgmc
bethsoftvid pgmyuvth
bfi pgssub
bink pictor
binkaudio_dct mace6 png
binkaudio_rdft mdec ppm
bmp ptxic
c93 qcelp
cavs qdm2gb
cdgraphics mlp qdraw
cinepak qpegdeo
cljr qtrle motionpixels
cook r10k
cscd r210 mp1float
cyuv ra_144
dca ra_288 mp2float
dfa rawvideo
dnxhd rl2adu
roq vp6
roq_dpcm vp6atexseqvideo
rpza vp6f
rv10 vp8
rv20 vqaehd
rv30 wavpack truemotion1
rv40 wmapro truemotion2
s302m wmav1 truespeech
sgi wmav2
shorten wmavoice
sipr wmv1vq
smackaud txd wmv2
smacker wmv3
smc wnv1
snow ws_snd1
sol_dpcm vb xan_dpcm
sonic xan_wc3
sp5x xan_wc4
srt xl vmdaudio
sunrast xsub vmdvideo
svq1 yopc
svq3 zlibis
targa zmbv
theora vp5

Enabled encoders:

Enabled hwaccels:

Enabled parsers:
aac mpeg4video
aac_latm flac mpegaudio
ac3 mpegvideo
cavsvideo h263 pnm
dca vc14
dirac vp3eg
dnxhd vp8
dvbsub

Enabled demuxers:
aac pcm_u24bengenient
ac3 pcm_u24le
aea pcm_u32be
aiff pcm_u32le
amr pcm_u8
anm pmp
apc pvam4
ape qcp
applehttp m4v r3d
asf rawvideomatroska
ass rl2 microdvd
au rmpeg
avi roq
avs rpl
bethsoftvid mmf rso
bfi segafilm
bink shorten
c93 siff
caf smacker
cavsvideo mpegps sol
cdg soxgts
daud spdif mpegtsraw
dfa srt mpegvideo
dirac str msnwc_tcp
dnxhd swf
dsicin thp
dts tiertexseq
dv tmv
dxa truehd
ea tta
ea_cdata nut tty
eac3 txd
ffm vc1
ffmetadata oma vc1t
filmstrip vmd_alaw
flac voc pcm_f32be
flic vqf pcm_f32le
flv w64 pcm_f64be
fourxm wav pcm_f64le
g722 wc3 pcm_mulaw
gsm wsaud pcm_s16be
gxf wsvqa pcm_s16le
h261 wtv pcm_s24be
h263 wv pcm_s24le
h264 xa pcm_s32be
idcin xwma pcm_s32le
iff yop_s8
image2 yuv4mpegpipeu16be
image2pipe pcm_u16le

Enabled muxers:
spdif

Enabled protocols:
file

Enabled filters:

Enabled bsfs:

Enabled indevs:

Enabled outdevs:

License: GPL version 2 or later
Creating config.mak and config.h...
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'
Makefile:157: warning: overriding commands for target `libavcodec/'
subdir.mak:27: warning: ignoring old commands for target `libavcodec/'
Makefile:157: warning: overriding commands for target `libavformat/'
subdir.mak:27: warning: ignoring old commands for target `libavformat/'
Makefile:157: warning: overriding commands for target `libavutil/'
subdir.mak:27: warning: ignoring old commands for target `libavutil/'

I'm using your build.bat

[nevcairiel]

Thats a bug in ffmpeg, it usually finishes when you give it some time - at least did for me.

[Mercury_22]

Strange until now it took like 5 min to build entire LAV now it's already like (~) 1/2 H and I'm still at this point

EDIT The only difference from the last time now I'm using GCC 4.6.1 Stable Release and before I was using GCC 4.6.1 Pre Release
P.S. MPC-HC it's building as usual (faster then LAV)

[nevcairiel]

Like i said, its a bug in ffmpegs build thingy, and i cba to track it down. Complain to them. :)

[nevcairiel]

Trying to play a mp4 with multiple test subtitles (http://www.multiupload.com/4VDVDSAZ5B) in MPC-HC using LAV Filters, LAV CUVID and EVR CP.
Video works fine, but the subs don't. (they work fine if I use MPC-HC internal mp4 splitter)
-> is it a bug or just me overlooking something?

Cu Selur

Basic support for MPEG-4 Timed Text subtitles (also known as MOV Text or tx3g) was added just now, and will be in 0.30.
Note that it only works for pure text subtitles and style information may be lost.

[nevcairiel]

Strange until now it took like 5 min to build entire LAV now it's already like (~) 1/2 H and I'm still at this point


I found the commit in ffmpeg that caused it and reverted it, it should build fine again now.

[Mercury_22]

I found the commit in ffmpeg that caused it and reverted it, it should build fine again now.

:thanks: working now (under 5 min :) )


Edit : spoke too soon X64 version won't start building ! e.g. using your build.bat I get stuck after building x86 version (successfully)

2>C:\LAV\libbluray\src\util/bits.h(281): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>c:\lav\libbluray\src\libbluray\bdnav\sound_parse.h(38): warning C4200: nonstan
dard extension used : zero-sized array in struct/union
1> arithutil.cpp
2> Cannot generate copy-ctor or copy-assignment operator when UDT conta
ins a zero-sized array
2> bluray.c
2>src\libbluray\bluray.c(345): warning C4244: '=' : conversion from 'int64_t' to
'int', possible loss of data
2>src\libbluray\bluray.c(429): warning C4244: 'argument' : conversion from 'uint
64_t' to 'size_t', possible loss of data
2>src\libbluray\bluray.c(844): warning C4244: 'argument' : conversion from 'uint
64_t' to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(863): warning C4244: 'argument' : conversion from 'uint
64_t' to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(876): warning C4244: 'argument' : conversion from 'uint
64_t' to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(919): warning C4244: 'argument' : conversion from 'uint
64_t' to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(972): warning C4244: '=' : conversion from 'uint64_t' t
o 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(1009): warning C4244: 'argument' : conversion from 'uin
t64_t' to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(1031): warning C4244: '=' : conversion from 'uint64_t'
to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(1051): warning C4244: '=' : conversion from 'uint64_t'
to 'unsigned int', possible loss of data
2>src\libbluray\bluray.c(1195): warning C4244: 'argument' : conversion from 'uin
t64_t' to 'unsigned int', possible loss of data
2>src\libbluray\bluray.c(1350): warning C4244: '=' : conversion from 'uint64_t'
to 'uint32_t', possible loss of data
2>src\libbluray\bluray.c(2211): warning C4018: '>' : signed/unsigned mismatch
2> graphics_controller.c
1> combase.cpp
2>src\libbluray\decoders\graphics_controller.c(945): warning C4018: '>=' : signe
d/unsigned mismatch
2>src\libbluray\decoders\graphics_controller.c(945): warning C4018: '>=' : signe
d/unsigned mismatch
2> graphics_processor.c
2>C:\LAV\libbluray\src\util/bits.h(73): warning C4244: '=' : conversion from 'in
t64_t' to 'off_t', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(75): warning C4244: '=' : conversion from 'in
t64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(151): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(157): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(229): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(281): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2> ig_decode.c
2>C:\LAV\libbluray\src\util/bits.h(73): warning C4244: '=' : conversion from 'in
t64_t' to 'off_t', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(75): warning C4244: '=' : conversion from 'in
t64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(151): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(157): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(229): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(281): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>c:\lav\libbluray\src\libbluray\decoders\../hdmv/mobj_parse.h(61): warning C420
0: nonstandard extension used : zero-sized array in struct/union
2> Cannot generate copy-ctor or copy-assignment operator when UDT conta
ins a zero-sized array
2> m2ts_demux.c
2> pes_buffer.c
2> pg_decode.c
2>C:\LAV\libbluray\src\util/bits.h(73): warning C4244: '=' : conversion from 'in
t64_t' to 'off_t', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(75): warning C4244: '=' : conversion from 'in
t64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(151): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(157): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(229): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(281): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2> hdmv_vm.c
1> cprop.cpp
2>c:\lav\libbluray\src\libbluray\hdmv\mobj_parse.h(61): warning C4200: nonstanda
rd extension used : zero-sized array in struct/union
2> Cannot generate copy-ctor or copy-assignment operator when UDT conta
ins a zero-sized array
1> ctlutil.cpp
2>src\libbluray\hdmv\hdmv_vm.c(875): warning C4244: 'return' : conversion from '
uint64_t' to 'uint32_t', possible loss of data
2>src\libbluray\hdmv\hdmv_vm.c(882): warning C4244: 'return' : conversion from '
uint64_t' to 'uint32_t', possible loss of data
2> mobj_parse.c
2>C:\LAV\libbluray\src\util/bits.h(73): warning C4244: '=' : conversion from 'in
t64_t' to 'off_t', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(75): warning C4244: '=' : conversion from 'in
t64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(151): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(157): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(229): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>C:\LAV\libbluray\src\util/bits.h(281): warning C4244: '=' : conversion from 'i
nt64_t' to 'int', possible loss of data
2>c:\lav\libbluray\src\libbluray\hdmv\mobj_parse.h(61): warning C4200: nonstanda
rd extension used : zero-sized array in struct/union
2> Cannot generate copy-ctor or copy-assignment operator when UDT conta
ins a zero-sized array
2> mobj_print.c
2>c:\lav\libbluray\src\libbluray\hdmv\mobj_parse.h(61): warning C4200: nonstanda
rd extension used : zero-sized array in struct/union
2> Cannot generate copy-ctor or copy-assignment operator when UDT conta
ins a zero-sized array
2> Compiling...
2> register.c
1> ddmm.cpp
1> dllentry.cpp
2> logging.c
2> strutl.c
1> dllsetup.cpp
2>Link:
2> Creating library C:\LAV\bin_Win32\libbluray\libbluray.lib and object C:\L
AV\bin_Win32\libbluray\libbluray.exp
2> Generating code
1> mtype.cpp
1> outputq.cpp
1> perflog.cpp
2> Finished generating code
2> libbluray.vcxproj -> C:\LAV\bin_Win32\libbluray\libbluray.dll
1> pstream.cpp
1> pullpin.cpp
2>PostBuildEvent:
2> Description: Copy .dll/.lib into library path
1> refclock.cpp
1> renbase.cpp
2> 1 file(s) copied.
2> 1 file(s) copied.
2>FinalizeBuildStatus:
2> Deleting file "C:\LAV\bin_Win32\libbluray\libbluray.unsuccessfulbuild".
2> Touching "C:\LAV\bin_Win32\libbluray\libbluray.lastbuildstate".
2>
2>Build succeeded.
2>
2>Time Elapsed 00:00:08.64
1> schedule.cpp
1> seekpt.cpp
1> source.cpp
1> strmctl.cpp
1> sysclock.cpp
1> transfrm.cpp
1> transip.cpp
1> videoctl.cpp
1> vtrans.cpp
1> winctrl.cpp
1> winutil.cpp
1> wxdebug.cpp
1> wxlist.cpp
1> wxutil.cpp
1>Lib:
1> baseclasses.vcxproj -> C:\LAV\bin_Win32\lib\strmbase.lib
1>FinalizeBuildStatus:
1> Deleting file "C:\LAV\bin_Win32\baseclasses\baseclasses.unsuccessfulbuild".
1> Touching "C:\LAV\bin_Win32\baseclasses\baseclasses.lastbuildstate".
1>
1>Build succeeded.
1>
1>Time Elapsed 00:00:13.57
3>------ Rebuild All started: Project: DSUtilLite, Configuration: Release Win32
------
3>Build started 07/02/2011 11:29:25.
3>InitializeBuildStatus:
3> Creating "C:\LAV\bin_Win32\DSUtilLite\DSUtilLite.unsuccessfulbuild" because
"AlwaysCreate" was specified.
3>ClCompile:
3> stdafx.cpp
3> BaseDSPropPage.cpp
3> CSSauth.cpp
3> CSSscramble.cpp
3> DeCSSInputPin.cpp
3> DShowUtil.cpp
3> filterreg.cpp
3> FontInstaller.cpp
3> H264Nalu.cpp
3> locale.cpp
3> registry.cpp
3>Lib:
3> DSUtilLite.vcxproj -> C:\LAV\bin_Win32\lib\dsutil.lib
3>FinalizeBuildStatus:
3> Deleting file "C:\LAV\bin_Win32\DSUtilLite\DSUtilLite.unsuccessfulbuild".
3> Touching "C:\LAV\bin_Win32\DSUtilLite\DSUtilLite.lastbuildstate".
3>
3>Build succeeded.
3>
3>Time Elapsed 00:00:02.97
4>------ Rebuild All started: Project: Demuxers, Configuration: Release Win32 --
----
5>------ Rebuild All started: Project: LAVAudio, Configuration: Release Win32 --
----
4>Build started 07/02/2011 11:29:28.
5>Build started 07/02/2011 11:29:28.
4>InitializeBuildStatus:
4> Creating "C:\LAV\bin_Win32\Demuxers\Demuxers.unsuccessfulbuild" because "Alw
aysCreate" was specified.
5>InitializeBuildStatus:
5> Creating "C:\LAV\bin_Win32\LAVAudio\LAVAudio.unsuccessfulbuild" because "Alw
aysCreate" was specified.
5>ClCompile:
5> stdafx.cpp
4>ClCompile:
4> stdafx.cpp
5>c:\lav\ffmpeg\libavutil\common.h(172): warning C4244: 'return' : conversion fr
om 'int64_t' to 'int32_t', possible loss of data
4>c:\lav\ffmpeg\libavutil\common.h(172): warning C4244: 'return' : conversion fr
om 'int64_t' to 'int32_t', possible loss of data
5> Bitstream.cpp
5> BitstreamParser.cpp
4> BaseDemuxer.cpp
4> BDDemuxer.cpp
5> dllmain.cpp
5> DTSDecoder.cpp
5> LAVAudio.cpp
4> ByteParser.cpp
4> ExtradataParser.cpp
4> LAVFAudioHelper.cpp
4> LAVFDemuxer.cpp
5> AudioSettingsProp.cpp
5> Media.cpp
5> PostProcessor.cpp
4> LAVFInputFormats.cpp
4> LAVFVideoHelper.cpp
4> LAVFStreamInfo.cpp
5> dts.cpp
4> LAVFUtils.cpp
4> StreamInfo.cpp
5>c:\lav\ffmpeg\libavutil\common.h(172): warning C4244: 'return' : conversion fr
om 'int64_t' to 'int32_t', possible loss of data
4>Lib:
4> Demuxers.vcxproj -> C:\LAV\bin_Win32\lib\demuxers.lib
4>FinalizeBuildStatus:
4> Deleting file "C:\LAV\bin_Win32\Demuxers\Demuxers.unsuccessfulbuild".
4> Touching "C:\LAV\bin_Win32\Demuxers\Demuxers.lastbuildstate".
4>
4>Build succeeded.
4>
4>Time Elapsed 00:00:02.66
5>Link:
5> Creating library C:\LAV\bin_Win32\LAVAudio\LAVAudio.lib and object C:\LAV
\bin_Win32\LAVAudio\LAVAudio.exp
5> Generating code
6>------ Rebuild All started: Project: LAVSplitter, Configuration: Release Win32
------
6>Build started 07/02/2011 11:29:32.
6>InitializeBuildStatus:
6> Creating "C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.unsuccessfulbuild" becaus
e "AlwaysCreate" was specified.
5> Finished generating code
6>ClCompile:
6> stdafx.cpp
6>c:\lav\ffmpeg\libavutil\common.h(172): warning C4244: 'return' : conversion fr
om 'int64_t' to 'int32_t', possible loss of data
5> LAVAudio.vcxproj -> C:\LAV\bin_Win32\LAVAudio\LAVAudio.ax
5>FinalizeBuildStatus:
5> Deleting file "C:\LAV\bin_Win32\LAVAudio\LAVAudio.unsuccessfulbuild".
5> Touching "C:\LAV\bin_Win32\LAVAudio\LAVAudio.lastbuildstate".
5>CustomBuildStep:
5> Description: Performing Custom Build Step
5> 1 file(s) copied.
5>
5>Build succeeded.
5>
5>Time Elapsed 00:00:04.29
6> dllmain.cpp
6> InputPin.cpp
6> SettingsProp.cpp
6> PacketQueue.cpp
6> OutputPin.cpp
6> LAVSplitter.cpp
6> StreamParser.cpp
6>Link:
6> Creating library C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.lib and object
C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.exp
6> Generating code
6> Finished generating code
6> LAVSplitter.vcxproj -> C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.ax
6>FinalizeBuildStatus:
6> Deleting file "C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.unsuccessfulbuild".
6> Touching "C:\LAV\bin_Win32\LAVSplitter\LAVSplitter.lastbuildstate".
6>CustomBuildStep:
6> Description: Performing Custom Build Step
6> 1 file(s) copied.
6>
6>Build succeeded.
6>
6>Time Elapsed 00:00:04.96
========== Rebuild All: 6 succeeded, 0 failed, 0 skipped ==========

[nevcairiel]

I just finished building x64, and all is fine. I'm using the full 4.6.0 MSYS package from XhmikosR, and it works just fine without any changes.

[Mercury_22]

I just finished building x64, and all is fine. I'm using the full 4.6.0 MSYS package from XhmikosR, and it works just fine without any changes.

That's strange I'm using the MSYS_MinGW_GCC_461_x86-x64_Full.7z (http://xhmikosr.1f0.de/index.php?folder=dG9vbHM=) from XhmikosR

And again MPC-HC both x64 and x86 are building fine and LAV x64 used too too :) with 4.6.0 MSYS full and 4.6.1 pre from ALEXIN (http://www.xvidvideo.ru/component/docman/cat_view/28-cross-mingwgcc-x86x64.html)

[VipZ]

Yep, x64 is also fine for me, upgraded to MSYS_MinGW_GCC_461_x86-x64_Full.7z from XhmikosR today as well.

EDIT, it seems AAC audio decoding when in MKV or MP4 no longer works on both x86 and x64 (MPC AAC loaded as low merit). This doesn't seem to be MinGW related.

[nevcairiel]

Fixed AAC decoding

[VipZ]

Fixed AAC decoding

Thanks

[Mercury_22]

I'm going mad please help!!!:helpful:
I've re download everything and I'm using the MSYS_MinGW_GCC_460_x86-x64_Fulll now but the x64 still won't build
So is it possible that the zip files I'm downloading directly from https://github.com/Nevcairiel/LAVFSplitter and http://git.1f0.de/gitweb?p=ffmpeg.git;a=summary and http://git.1f0.de/gitweb?p=libbluray.git;a=summary are different ?

P.S. I've just reinstall everything today (windows7 VS2010....) so maybe I've forgot some settings (variable, path or....?:confused:
Again both MPC-HC x64 and x86 are building fine !

Please help:mad:

EDIT:
This has become ridiculous
I've just build a clean VM with just Windows 7 x64 SP1 & C++ (VS 2010) nothing else not even updates
I've download Nevcairiel-LAVFSplitter-0.29-12-g74976d4.zip, ffmpeg-261528c.zip, libbluray-3fa955d.zip and extracted all in C:\LAV
I've download MSYS_MinGW_GCC_460_x86-x64_Full.7z extracted to C:\MSYS and edited "fstab" file in C:\MSYS\etc\ added "C:\MSYS\mingw \mingw"
I've added the following environment variables:
Variable Value
MSYS C:\MSYS
MINGW32 C:\MSYS\mingw
MINGW64 C:\MSYS\mingw
and modified the "Path" environment variable by adding at the end: ;%MINGW32%\bin;%MINGW64%\bin;%MSYS%\bin

Then I've run the build.bat from c:\Lav and ....


Nothing is changed !!!
The x86 it's building successfully but when to start building x64 it's "hanging"
At this moment 20 minutes have past and the building of x64 still hasn't start

What am I doing wrong ? :mad::confused::helpful:

[Gleb Egorych]

About A/V sync: some time ago I reported (http://forum.doom9.org/showthread.php?p=1503907#post1503907) that AVI crash bug and Zoom Player hang bug (http://forum.doom9.org/showthread.php?p=1501013#post1501013) were fixed by the same commits. Since about 7 weeks of usage I can say that the ZP hang is fixed only when A/V sync is disabled (I usually use LAV Audio this way). If A/V sync is enabled then ZP hang occurs sometimes. It's hard to reproduce reliably but it definately relates to the option.

[nevcairiel]

Thats unlikely. The A/V sync option will not do *anything* at all on 99% of all files. Only broken files with a gap in audio (or an overlap) will actually trigger it. And even if its triggered, all it does is slightly adjust the audio timestamps, it doesn't do anything fancy that would cause a hang, unless the audio renderer you're using is completely insane in the membrane

[VipZ]

@ Mercury

Try going for separate MinGW enviroment, http://sourceforge.net/projects/mingw-w64/files/Toolchains%20targetting%20Win64/Personal%20Builds/rubenvb/x86_64-w64-mingw32-gcc-4.6.1-2_rubenvb.zip/download and http://sourceforge.net/projects/mingw-w64/files/Toolchains%20targetting%20Win32/Personal%20Builds/rubenvb/i686-w64-mingw32-gcc-4.6.1-2_rubenvb.zip/download

I used that for a short while when ffdshow broke on my old MinGW env, but I have since moved back to the combined builds from xvidvideo.ru which work for LAV/ffdshow and MPC perfectly for me.

[Mercury_22]

@ Mercury

Try going for separate MinGW enviroment, http://sourceforge.net/projects/mingw-w64/files/Toolchains%20targetting%20Win64/Personal%20Builds/rubenvb/x86_64-w64-mingw32-gcc-4.6.1-2_rubenvb.zip/download and http://sourceforge.net/projects/mingw-w64/files/Toolchains%20targetting%20Win32/Personal%20Builds/rubenvb/i686-w64-mingw32-gcc-4.6.1-2_rubenvb.zip/download

I used that for a short while when ffdshow broke on my old MinGW env, but I have since moved back to the combined builds from xvidvideo.ru which work for LAV/ffdshow and MPC perfectly for me.
I've used a clean "everything" on the VM and since is working for you but not for me....
Do you rebuild everything (are you using the build.bat) ?
Do you download the source from the server or are you using git (TortoiseGit)?

[VipZ]

I've used a clean "everything" on the VM and since is working for you ....
Do you rebuild everything (are you using the build.bat) ?
Do you download the source from the server or are you using git (TortoiseGit)?

For LAV, Nev has made it very simple for us :)

I have pretty much refreshed my environment today.
I download via TortoiseGit, did a fresh download today when I 1st encountered the delayed building, tho it still built within 5-7min or so.
I have also deleted my whole old MSYS folder today and used MSYS_MinGW_GCC_461_x86-x64_Full.7z (I don't bother editing any files even if its mentioned to do so)
I have got my env var's set to how you mentioned.
To build, i just run build.bat and wait around 3min for both x86 and x64 to compile.

If you have issues, you can try one more thing I have found cause build issues with ffdshow when I was playing around with my build scripts, edit build_ff_x64.sh and change the line "make -j8 &&" to "make -j4 &&"

[Mercury_22]

For LAV, Nev has made it very simple for us :)

I have pretty much refreshed my environment today.
I download via TortoiseGit, did a fresh download today when I 1st encountered the delayed building, tho it still built within 5-7min or so.
I have also deleted my whole old MSYS folder today and used MSYS_MinGW_GCC_461_x86-x64_Full.7z (I don't bother editing any files even if its mentioned to do so)
I have got my env var's set to how you mentioned.
To build, i just run build.bat and wait around 3min for both x86 and x64 to compile.

If you have issues, you can try one more thing I have found cause build issues with ffdshow when I was playing around with my build scripts, edit build_ff_x64.sh and change the line "make -j8 &&" to "make -j4 &&"
It didn't work :thanks: anyway
I'm just puzzled cause I can build MPC-HC x64 (and x86 and lav x86) without any problems ! and LAV x64 it's just hanging there without any error or something

[VipZ]

It didn't work :thanks: anyway
I'm just puzzled cause I can build MPC-HC x64 (and x86 and lav x86) without any problems ! and LAV x64 it's just hanging there without any error or something

I find compiling mpc isn't a very good test, what about ffdshow x64?

[Gleb Egorych]

Thats unlikely. The A/V sync option will not do *anything* at all on 99% of all files. Only broken files with a gap in audio (or an overlap) will actually trigger it. And even if its triggered, all it does is slightly adjust the audio timestamps, it doesn't do anything fancy that would cause a hang, unless the audio renderer you're using is completely insane in the membrane
Actually I experienced hangs on DVB files (untouched and remuxed to mkv). I'm not sure whether all of them were broken.

I use DirectShow Audio renderer, "DirectSoud: Speakers (Audigy 2 ZS (WDM))" to be exact.

[Mercury_22]

OK I've build the LAV x64 too !!! (using MSYS_MinGW_GCC_461_x86-x64_Full.7z) but wtf ? I had to install TortoiseGit and get the code that way !
So it seems that I was right ?! = the zip files I've downloaded before are not the same as the code I get with TortoiseGit ?!!

[jmone]

There seems to be an issue connecting LAVSplitter to FFDSHOW for VC-1 streams in an EVO container - http://yabb.jriver.com/interact/index.php?topic=64716.0

[nevcairiel]

Sucks to have jumped onto the HD-DVD bandwagon before it burned, eh!
EVO is a dead format, ffdshow a broken decoder, and my time is limited.

For the record, i already know that ffdshow fails to honor pulldown flags commonly present on VC-1 EVOs, so even if it would try to decode, it would not play properly.

[jmone]

But where would the fun be if we didn't jump on the various bandwagons early! It is not a big issue for me as I don't personally play EVO content directly and will be remuxing what I want (prob to Blu) anyway. I'll copy your response over at MC and see if they want (or could be bothered) to use Haali for EVO as it is fine in the filter chain with ffdshow for this stuff.

[jmone]

...... "ffdshow a broken decoder, and my time is limited" :) One can only hope to read stuff into what future projects you may be considering!

[adam777]

Hello all,
I came across a file with the following MediaInfo properties:

General
Unique ID :
Complete name :
Format : Matroska
File size : 19.9 GiB
Duration : 3h 48mn
Overall bit rate : 12.5 Mbps
Encoded date : UTC 2011-06-28 08:33:46
Writing application : mkvmerge v4.8.0 ('I Got The...') built on May 24 2011 03:12:58
Writing library : libebml v1.2.0 + libmatroska v1.1.0

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L4.1
Format settings, CABAC : Yes
Format settings, ReFrames : 5 frames
Codec ID : V_MPEG4/ISO/AVC
Duration : 3h 48mn
Bit rate : 10.8 Mbps
Width : 1 920 pixels
Height : 800 pixels
Display aspect ratio : 2.40:1
Frame rate : 23.976 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.292
Stream size : 16.7 GiB (84%)
Title : x264
Writing library : x264 core 115 r1995 c1e60b9
Encoding settings : cabac=1 / ref=5 / deblock=1:-3:-3 / analyse=0x3:0x113 / me=umh / subme=10 /
psy=1 / psy_rd=1.10:0.00 / mixed_ref=1 / me_range=64 / chroma_me=1 / trellis=2 / 8x8dct=1 / cqm=0 /
deadzone=21,11 / fast_pskip=0 / chroma_qp_offset=-2 / threads=12 / sliced_threads=0 / nr=0 / decimate=1 /
interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=6 / b_pyramid=2 / b_adapt=2 / b_bias=0 /
direct=3 / weightb=1 / open_gop=0 / weightp=2 / keyint=250 / keyint_min=23 / scenecut=40 / intra_refresh=0 /
rc_lookahead=250 / rc=2pass / mbtree=1 / bitrate=10765 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 /
qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / aq=1:0.90
Language : English

Audio #1
ID : 2
Format : DTS
Format/Info : Digital Theater Systems
Format profile : ES
Codec ID : A_DTS
Duration : 3h 48mn
Bit rate mode : Constant
Bit rate : 1 510 Kbps
Channel(s) : 7 channels / 6 channels
Channel positions : Front: L C R, Side: L R, Back: C, LFE / Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Compression mode : Lossy
Stream size : 2.41 GiB (12%)
Title : DTS-ES 6.1
Language : English

Now, I have a 5.1 speaker setup and managed to apply the needed matrix in FFDShow in order to convert the 6.1 output of LAV Audio to 5.1 (splitting back center to play equally on side left, side right)
However, it seems that a built-in 5.1 track is also available, is there a way to use it instead of the 6.1 track, or is it some sort of MediaInfo error?
Thanks, Adam.

[nevcairiel]

The 5.1 track is just the DTS core, which will simply not contain any of the info added by the extra back center channel.

[adam777]

OK, what I thought.
On a practical view - is there any way to let LAV Audio decode the DTS core, rather than sending the 6.1 signal through FFDShow for downmixing?

[nevcairiel]

No. "Features" that purposely degrade quality are not my targets.

[adam777]

Fair enough :)
:thanks:

[Sebastiii]

Hi :)

Sometimes when i seek in the movie, i lost center channel, a little seek and it's back, it happen rarely lol.
I didn't have log and it's really hard to reproduce.
Thanks :)
Seb.

[Midzuki]

Well, I've just tested LAV Audio again, and I'm glad it deals with Arcsoft's DTS decoder correctly :cool:

:thanks: :thanks: :thanks:

Notwithstanding, it cannot decode MLP in .mkv (not through Haali splitter at least) :confused:

[nevcairiel]

Do you have a sample of a MLP file? I've only ever seen TrueHD, and not MLP. Also, try with LAV Splitter.

[Midzuki]

That's weird :confused: LAV Splitter refuses connecting to ffdshow's audio decoder :confused: however LAV Audio works fine at decoding MLP in Matroska with LAV Splitter, and ffdshow's MLP decoder works only with Haali splitter :confused:

Do you have a sample of a MLP file?

Yes, but I will share them only if you promise :devil: there will be ZERO issues with:

a) channel count ;
b) channel order ;
c) channel groups with half samplerates ;

:)

[nevcairiel]

I promise you one thing: Without a sample, nothing will change! :)

[Midzuki]

I promise you one thing: Without a sample, nothing will change! :)

Fair enough :D

[robpdotcom]

Here's an mlp sample - it contains a 2.0 44.1kHz 24bit track, and 5.1 48kHz 24bit track inside mka:

http://www.megaupload.com/?d=B83YD2L9

It's the same for me as Midzuki - ffdshow will connect to Haali, LAVaudio will connect to LAVsplitter. With MPC-HC's internal Matroska splitter, ffdshow is fine, LAVaudio thinks it's AC3.

Something you might find interesting: If TrueHD bitstreaming is enabled in ffdshow, it will bitstream it as TrueHD. I get the blue light on my AVR with either track - the 5.1 track bitstreams fine, but the 2.0 track plays at what seems like 2x speed (I'm guessing it confuses my AVR because the 2.0 track is 44.1kHz).