FFmpegSource

[qyot27]

Can you make a build based on ffmbc?- some things are different there.
Those patches aren't for the C plugin (more likely GCC in general, actually), and trying to integrate them results in compilation failure(s). So unless it gets adjusted to compile cleanly, no.

And this was after hitting ffmbc with a hammer to get it to compile cleanly. It wasn't compiling libavcodec/timecode.c and not linking it into libavformat, causing undefined references in mov and mxfdec.

[Chirico]

Is there any special configuration options needed to get this to build for and load in vapoursynth? I've compiled this on OS X and am trying to load the libffms2.dylib but I keep getting the error about no entry point found.

[Myrsloik]

Is there any special configuration options needed to get this to build for and load in vapoursynth? I've compiled this on OS X and am trying to load the libffms2.dylib but I keep getting the error about no entry point found.

The build system is a bit of a mess about these things. Wait for the next revision where I'm going to fix it if you want to keep it simple. I think you also need to pass an option to enable vs support.

[Chirico]

Ok, I read through the configuration options quickly and didn't see anything. I'll look through it again.

Edit:
Figured out the issue. Didn't notice that you needed to run the autogen.sh that was mentioned in a previous post. Now it compiled and loaded fine. Again, for anyone else on OS X that doesn't want to compile it themselves you can grab it here (http://www.sendspace.com/file/eb0dq3).

[qyot27]

I've run into an issue trying to cross-compile the trunk to test VapourSynth support with it again.

PKG_CONFIG_PATH=$HOME/win32_build/lib/pkgconfig ./configure --prefix=$HOME/win32_build \
--host=i686-w64-mingw32 --build=i686-w64-mingw32
results in
configure: error: cannot run C compiled programs.
If you meant to cross compile, use `--host'.

Leaving out --build, configure completes, compiles fine. Previously, it wouldn't complain about the presence of both --build and --host, but that was prior to the most recent set of updates, and it might have also been prior to me updating to Ubuntu 12.10, which ships with newer versions of autotools than 12.04 did (and I cannot remember if my attempts to do any of this were after October 18th, or if they were all prior to it). Not sure if this could be contributing to the problem below.



PKG_CONFIG_PATH=$HOME/win32_build/lib/pkgconfig ./configure --prefix=$HOME/win32_build \
--host=i686-w64-mingw32 --enable-shared
configure completes, but won't link to libpthreadGC2.a or libopus.a

The warning/error itself:
*** Warning: This system can not link to static lib archive
/usr/bin/../lib/gcc/i686-w64-mingw32/4.7.2/../../../../i686-w64-mingw32/lib/../lib/libopus.la.
*** I have the capability to make that library automatically link in when
*** you link to this library. But I can only do this if you have a
*** shared version of the library, which you do not appear to have.



If necessary, the output from the ffmpeg it was linking against:
ffmpeg version r46401 git-3af7919 Copyright (c) 2000-2012 the FFmpeg developers
built on Nov 3 2012 19:00:58 with gcc 4.7.2 (GCC)
configuration: --prefix=/home/qyot27/win32_build --cross-prefix=i686-w64-mingw32- --enable-gpl
--enable-version3 --disable-w32threads --enable-memalign-hack --enable-avresample
--disable-decoder=utvideo --enable-libutvideo --enable-libopus --disable-encoders --disable-muxers
--disable-debug --disable-network --disable-hwaccels --disable-indevs --disable-outdevs
--cpu=pentium3 --extra-cflags='-march=pentium3 -mtune=pentium3 -DPTW32_STATIC_LIB'
--target-os=mingw32 --arch=x86
libavutil 52. 3.100 / 52. 3.100
libavcodec 54. 71.100 / 54. 71.100
libavformat 54. 35.100 / 54. 35.100
libavdevice 54. 3.100 / 54. 3.100
libavfilter 3. 21.106 / 3. 21.106
libswscale 2. 1.102 / 2. 1.102
libswresample 0. 16.100 / 0. 16.100
libpostproc 52. 1.100 / 52. 1.100

[18fps]

Thank you for all the work you put in this great plugin!
I'd like to know something. It's possible to load a sequence of numbered images (for ex. 00000000.dpx, 00000001.dpx, etc.) with FFVideoSource (like ffmpeg with the option -f image2)?

[Myrsloik]

Thank you for all the work you put in this great plugin!
I'd like to know something. It's possible to load a sequence of numbered images (for ex. 00000000.dpx, 00000001.dpx, etc.) with FFVideoSource (like ffmpeg with the option -f image2)?

No. THIS IS FILLER MATERIAL

[18fps]

No. THIS IS FILLER MATERIAL

Well, I can always create a .mov (prores 10 bit) with ffmbc and then open it with ffmpeg in avisynth. Thank you!

[LigH]

ImageSource() in AviSynth scripts can open image sequences. But probably only usual image formats. I doubt that it knows *.dpx - it may have restrictions to support only RGB32 compatible formats.

[18fps]

ImageSource() in AviSynth scripts can open image sequences. But probably only usual image formats. I doubt that it knows *.dpx - it may have restrictions to support only RGB32 compatible formats.

There's also Immaavs, which uses ImageMagick to open the dpx, but it dithers them to 8 bits. Creating the mov I keep the 10 bits and with the ffms2 hack (http://forum.doom9.org/showthread.php?p=1580141#post1580141) with "enable10bithack" I can even process it in high bit depth. I'm still trying to understand what kind of avisynth processing I can do. At the end I want to output either 10 bit dpx or x264 10 bit.

[mariner]

Problem with hard telecined video

Greetings.

I have a number of hard telecined BD which I'd like to convert to 23.976fps progressive with the following script. For the attached sample, x264 returns 960 frames, twice the frame count and duration using DirectShowSource.

Appreciate advice on correct usage.

System : W7x86, Avisynth 2.58, ffms2-r725-icl-4, latest Haali and FFdshow, Intel G45 graphics card.

FFvideoSource("zoroark.m2ts")
AssumeTFF()
TFM()
TDecimate()


Zoroark.m2ts (http://www2.zshare.ma/o9h4mnlaioed)

[sneaker_ger]

ffms can't find a suitable output format in VapourSynth for these:
http://media.xiph.org/sintel/sintel-4k-png16/
http://www11.zippyshare.com/v/74717882/file.html

ffms2 with 10 bit hack for AviSynth seems to work. (Does it downconvert/dither/clamp to 10 bit or are the 16 bits kept?)

[LigH]

@ mariner:

Never load transport streams with FFMS2 directly until their authors report that their transport stream splitter is finally correct. Instead, remux it to MKV with mkvmerge (mkvtoolnix) before indexing.

[mariner]

@ mariner:

Never load transport streams with FFMS2 directly until their authors report that their transport stream splitter is finally correct. Instead, remux it to MKV with mkvmerge (mkvtoolnix) before indexing.

Greetings LigH, thanks for the kind reply.

1. Does FFVidoeSource have problem dealing with interlaced video?

2. Is m2ts to mkv remuxing required for progressive video?

3. I've encountered a strange problem with FFVidoeSource outputting incorrect number of frames. Using the attached 136 frame progressive clip as input, a simple x264 encode with FFM2_r725_icl_4 results in 134 frames, while the version complied by Weirdo results in 136 frames. Is this a different problem?

Many thanks and best regards.

cut.m2ts
(http://www2.zshare.ma/fbbp7noio4sf)

[LigH]

1. Interlaced H.264 in (M2)TS: Most probably yes. If not anymore, FFMS2 authors will certainly announce that, but until today, instead, people kept pointing at the same explanation why it is so difficult to fix.

2. Still recommendable; FFMS2 splitters officially do not support any transport streams well; and interlaced content seems to be just another dimension of issues.

[mariner]

1. Interlaced H.264 in (M2)TS: Most probably yes. If not anymore, FFMS2 authors will certainly announce that, but until today, instead, people kept pointing at the same explanation why it is so difficult to fix.

2. Still recommendable; FFMS2 splitters officially do not support any transport streams well; and interlaced content seems to be just another dimension of issues.

Thanks for the reply, LigH.

1. Did you get a chance to look at issue #3? Weirdo's version is found here:

http://forum.doom9.org/showthread.php?p=1594210#post1594210

2. Is it necessary to remove audio when remux to mkv?

3. Does FFvideoSource work with raw h264 format?

Many thanks and best regards.

[LigH]

Sorry, I do not belong to its developers. I do not even convert videos very frequently.

Usually it is recommendable to keep audio tracks while remultiplexing to MKV, because that keeps them in sync with the video if done correctly. Just note the documentation regarding indexing files with audio and video streams, don't miss the function FFmpegSource2() { import("FFMS2.avsi") }.

Yes, FFMS2 comes with a documentation. It does contain useful information. Another answer to your questions is:

Because of LAVF's demuxer, most raw streams (such as elementary h264 and other mpeg video streams) will fail to work properly.

Thumb rule: Try to use ffmsindex.exe (a command line tool) with the desired media file before loading an AviSynth script the first time. It may already show error messages if there are problems. Or it will create a comprehensive index file for best use.

[Abs62]

FFMS2 compiled with last ffmpeg can't decode aac audio ("FFAudioSource: Bad audio format"). It seems after commit "aacdec: use float planar sample format for output" (26.11).

[qyot27]

FFMS2 compiled with last ffmpeg can't decode aac audio ("FFAudioSource: Bad audio format"). It seems after commit "aacdec: use float planar sample format for output" (26.11).
This is also the case with MP3 and now AC3 (it's really picked up in the last couple months with many of the other audio decoders as well). I've had to resort to using externally-decoded PCM files for audio input when I've tested with this.

This is being pulled upstream from libav, so it definitely affects both of them.

http://git.libav.org/?p=libav.git&a=search&h=HEAD&st=commit&s=planar
http://git.videolan.org/?p=ffmpeg.git&a=search&h=HEAD&st=commit&s=planar

I'm hoping that this isn't actually a limitation of AviSynth that prevents it from working. In such a case, I wonder if the solution would be to finally do something with SWResample and/or AVResample.

[JEEB]

There basically needs to be some work on switching the audio decoding to the current audio decoding API (avcodec_decode_audio4), as well as then possibly auto-packing it with avresample into the buffer that the calling application provides.

I did hack (https://github.com/jeeb/ffms2/commit/cfe4627a698ea987718b0a8f076df9b50dc1dde9) on the problem some time ago, and it doesn't seem to be too hard to fix, but it was a hack I wrote while very tired so it doesn't work (incorrect dumping of audio at least in the indexer as I left that out completely), as well as I never tell ffms2 that it won't be getting the type of audio it was thinking it'd get. It does seem to actually stay relatively stable memory consumption wise when decoding, though (tested by having the command line indexer go over and decode an audio track of a sample AAC file I had lying around -- which did produce a nice 7GB+ w64 file from 1min30sec of stereo, but seemed stable otherwise).

Also, looking at how I tried to use avresample there, it doesn't seem like a too shabby of a library usability wise.

[easyfab]

I have a problem with ffms2 > r722 and mpc-hc .
With r722, it works but r725 doesn't work with mpc-hc but ok with graphedit ?
As someone the same problem and is this a mpc-hc bug or FFMS2 ?

[LigH]

I don't understand your problem.

FFMS2 is a source plugin for AviSynth, not for MPC-HC.

AviSynth scripts are not meant to be viewed in MPC-HC, because AviSynth is a VfW frameserver, but MPC-HC is a media player for DirectShow (or own internal decoders). To view an AviSynth skript, use a program with VfW support (VirtualDub) or native AviSynth environment support (AvsPmod).

The DirectShow compatibility bridge for VfW may work in DirectShow media players, but doesn't have to.

[jmac698]

I'm trying to return select frame types with ffms2. I haven't been able to get it to work.


X=dir+"20121105002302.MTS"
A = FFAudioSource(X)
V = FFVideoSource(X,varprefix = "src")
ScriptClip("""
subtitle(chr(srcFFPICT_TYPE))
""")

It's showing FFPICT_TYPE as undefined, also srcFFPICT_TYPE. How can I return only select frames, when it does work?

Edit, need frame after for first frame:

varprefix = "src"
A = FFAudioSource(X)
V = FFVideoSource(X,varprefix = varprefix)
ScriptClip("""
chr(eval(varprefix + "FFPICT_TYPE"))=="I"?last:nop
""", after_frame=true)

Now, how to delete frames that aren't I frames? Does it have to be two pass? Can't I use a compile time trick somehow?

[Gavino]

Now, how to delete frames that aren't I frames? Does it have to be two pass? Can't I use a compile time trick somehow?
Deleting frames via ScriptClip is a bit tricky, but you can use the technique from here.

[Myrsloik]

I'm trying to return select frame types with ffms2. I haven't been able to get it to work.


X=dir+"20121105002302.MTS"
A = FFAudioSource(X)
V = FFVideoSource(X,varprefix = "src")
ScriptClip("""
subtitle(chr(srcFFPICT_TYPE))
""")

It's showing FFPICT_TYPE as undefined, also srcFFPICT_TYPE. How can I return only select frames, when it does work?

Edit, need frame after for first frame:

varprefix = "src"
A = FFAudioSource(X)
V = FFVideoSource(X,varprefix = varprefix)
ScriptClip("""
chr(eval(varprefix + "FFPICT_TYPE"))=="I"?last:nop
""", after_frame=true)

Now, how to delete frames that aren't I frames? Does it have to be two pass? Can't I use a compile time trick somehow?

Compile time as in c++ compile time. Using the ffms2 api directly it's very easy to build the list. It would probably only take 15 lines if you modify the ffms2 filter for your special needs.

[StainlessS]

@Jmac, If I understand correctly what you want to do, see here:
http://forum.doom9.org/showthread.php?p=1539598#post1539598

You might want to convert to using FrameSelect() rather than Prune() by removing the '0' clip index in WriteFileIf.

[StainlessS]

This extracts I/KEY frames only


X = "D:\avs\avi\1.avi"
PREFIX = "SRC_"
FFIndex(X)
V=FFVideoSource(X,varprefix = PREFIX)
NewClip=0
GSCript("""
for(i=0,V.FrameCount-1) {
V.RT_AverageLuma(n=i,w=1,h=1) # Force update of SRC_FFPICT_TYPE
if(SRC_FFPICT_TYPE==73) { # 'I'
NewClip = (IsClip(NewClip)) ? NewClip ++ V.Trim(i,-1) : V.Trim(i,-1)
}
}
""")
return NewClip

[StainlessS]

I modded the Prune select script linked 2 posts earler for this purpose, creates list for FrameSelect() instead of Prune().


X = "D:\avs\avi\1.avi"
PREFIX = "SRC_"
PROCESS = True # False will create list of I/KEYFrames, Need to play all way through to create list.
# True will auto create list and additionally select and return I/KEYFrames ONLY
FFIndex(X)
V=FFVideoSource(X,varprefix = PREFIX)
Clip = V.SelectFrames("IFrames.txt","SRC_FFPICT_TYPE==73") # Create list of I/KeyFrames

GSCript("""
if(PROCESS) {
For(i=0,Clip.framecount-1) {
Clip.RT_AverageLuma(n=i,w=1,h=1) # Force list creation by sampling single pixel from each frame, result discarded
}
Clip = 0 # Destroy Clip, Not necessary here. Can be used to call clip destructor to close/flush eg file writing
Clip = V.FrameSelect(CMD="IFrames.txt") # Select all I/KeyFrames using newly created list.
}
""")

return Clip


#SelectFrames("In_cmd.txt","YDifferenceFromPrevious>=0.1 || current_frame==0") # Always keeps first frame 0 (else delete)
#DeselectFrames("In_cmd.txt","YDifferenceFromPrevious>=0.1 || current_frame==0") # Always deletes first frame 0 (else keep)
#SelectFrames("In_cmd.txt","YDifferenceToNext>=0.1 || current_frame==framecount-1") # Always keeps last frame (else delete)
#DeselectFrames("In_cmd.txt","YDifferenceToNext>=0.1 || current_frame==framecount-1") # Always deletes last frame (else keep)

Function SelectFrames(clip c, string fileName, string condition, bool "Fast") {
# Conditional KEEP frames Command File generator for FrameSelect()
# MUST call with correct colorspace for condition eg YDifferenceFromPrevious requires a Planar colorspace.
# May require additional condition for end frames for eg YDifferenceFromPrevious and YDifferenceToNext.
# Any-DifferenceFromPrevious,
# Gives 0 for frame zero and so you may want to add to condition " || current_frame==0" to include frame 0.
# Any-DifferenceToNext,
# Gives 0 for last frame and so you may want to add to condition " || current_frame==framecount-1 " to include last frame.

c=(Default(Fast,true))?c.AssumeFPS(250.0):c # Fast as we can if Fast = true (Default true)
WriteFileIf(c, fileName, condition, "current_frame", append=false)
}

Function DeselectFrames(clip c, string fileName, string condition, bool "Fast") {
# Conditional DELETE frames Command File generator for FrameSelect()
# MUST call with correct colorspace for condition eg YDifferenceFromPrevious requires a Planar colorspace.
# Is EXACT opposite of SelectFrames() for same conditional.
# NOTE, any additional end frame condition as used in SelectFrames, will also be inverted
# eg " || current_frame==0", frame 0 would be deleted.

c=(Default(Fast,true))?c.AssumeFPS(250.0):c # Fast as we can if Fast = true (Default true)
condition = "!(" + condition + ")"
WriteFileIf(c, fileName, condition, "current_frame", append=false)
}


EDIT: After list created, can just use this:

X = "D:\avs\avi\1.avi"
Return FFVideoSource(X).FrameSelect(CMD="IFrames.txt")


FrameSelect() Here: http://forum.doom9.org/showthread.php?t=164497

EDIT: At the risk of stating the obvious, you would not want to try above on eg HuffYUV compressed AVI as they are ALL
keyframes. For AVI, try with DivX or XVid or some other lossy codec.

EDIT: You can add "Show=True" to the FrameSelect() line to show original frame numbers.

[qyot27]

Some new builds:
FFMS2 r739 (http://www.mediafire.com/?9j091vw73zu9kf4)
VapourSynth-only plugin, FFmpeg N-48238-g10a3fa8

FFMS2 r739 C-plugin (http://www.mediafire.com/?8u1c14yqymvxmnc)
AviSynth C-plugin & VapourSynth plugin, FFmpeg N-47083-g66c3bac


The reason for the disconnect is the current stuff going on in libavcodec concerning planar audio. The build with AviSynth support uses an FFmpeg from just before the shift occurred for aacdec and mpegaudiodec (that is, it's from November 25th). I didn't want to mess with doing targeted reverts, so it's just rolled back to the last good commit (although Vorbis is still affected; it got switched over on Nov 20th). The VapourSynth-only build uses a current version of FFmpeg, since audio doesn't matter for it anyway.

[Selur]

can someone compile a normal FFMS2 version for Avisynth (no C-plugin?)

[LigH]

The "planar audio" changes in libavcodec may be the reason why it doesn't compile easily anymore as usual AviSynth plugin, probably due to the calling conventions of specific function parameters supported by C.

[StainlessS]

can someone compile a normal FFMS2 version for Avisynth (no C-plugin?)

If you just want an autoload plug, then create an avsi in plugins that autoloads the C plug.

[Selur]

thanks, for the advice but I don't want an autoload plug,... :)

[StainlessS]

OK, I see, its for use by your 'Hybrid'.

[Selur]

Mainly, since I create most of my small script using it (or a slightly modified version of it). :)

[qyot27]

If you just want an autoload plug, then create an avsi in plugins that autoloads the C plug.
My builds of the C plugin (and the corresponding git repo) have the load function inside of FFMS2.avsi. Not that it matters for the reason why Selur was asking for a trunk/MSVC build.

The "planar audio" changes in libavcodec may be the reason why it doesn't compile easily anymore as usual AviSynth plugin, probably due to the calling conventions of specific function parameters supported by C.
The planar audio stuff in libavcodec only means that audio doesn't work (unless it's PCM); it doesn't stop it from compiling. At least, not yet - and it's not the planar audio that would stop it anyway.

[JEEB]

Plorky has done some more work (https://github.com/tgoyne/ffms2/commits/master) on libavresample'ing the audio part so interleaved audio can be gotten where needed.

It's not finished and currently adds a new API for setting the needed audio format as far as I can see. Should end up in the ffms2's official repository at some point.

[burfadel]

Some new builds:
FFMS2 r739 C-plugin (http://www.mediafire.com/?8u1c14yqymvxmnc)
AviSynth C-plugin & VapourSynth plugin, FFmpeg N-47083-g66c3bac

I tried this avisynth build. It worked most of the time, but when doing batch encoding occasionally it crashed, but worked the next time I ran it.

I looked around for alternative recent builds to see if they were affected too. I came across this one:
ffms2-r742+8-ffmpeg-gb454c64
https://skydrive.live.com/redir?resid=5FEFAEC438258885%21982&authkey=%21ADzTzz_MWxo3Dks

and it works beautifully! I have been continually batch encoding (on list) for a day or so with that build now. With Qyot27's build it crashed every couple of hours or so.

Qyot27, I don't know what it is about our build, but something isn't quite right? It isn't a criticism, I do appreciate the new builds! I'm just pointing it out in case something needs rectifying.

[qyot27]

Without more information, the only thing I can guess is that it might be related to the memleak-related fixes in r740 and r741. r739 certainly wouldn't have these, while r742 would. Unless it was actually something in FFmpeg that's gotten fixed between November and now - that's also a possibility.

[burfadel]

That is a possibility! I just thought I'd let you know and point it out to others if they have the same issue. If just encoding a single file you probably won't come across it.

EDIT: Progressing through a large batch of files (encoding at HD, custom very slow settings in x264), still going good with the r742 I linked earlier (unlike the r739 which crashed a couple of times a day).

[Selur]

FFVideoSource("H:\TEST\yuv422_16bitBitEndianStereo.mov",cachefile="H:\Temp\mov.ffindex",threads=1)
crashes
mov contains: progessive YUV2 and pcm audio
- plays fine with mplayer/ffplay
- ffindex doesn't report any problems on index creation
sample: https://docs.google.com/file/d/0B_WxUS1XGCPAblRwN2ozazd0Y28/edit?usp=sharing

anyone got an idea where the problem is?

[kolak]

Try disabling audio.

[Selur]

How? afaik FFVideoSource is video only.

[burfadel]

I believe that is the case too, hence the difference between FFVideoSource and FFAudioSource

[kolak]

How? afaik FFVideoSource is video only.

Delete it in MOV with QT player Pro or create ref mov file without audio.

update- still bad

It does crash for me also (with different version of ffvideo)- including avisynth. It also crashes after re-wrapping to AVI.

FFmpeg converts it fine thought.

[Selur]

FFmpeg converts it fine thought.
as does mencoder, that's why I wondered if I missed some restriction of FFMS.

[LigH]

As far as I remember, I had issues with some raw YUV AVIs in FFMS2 too (Direct Stream Copy from AviSynth to AVI in VirtualDub). Since then I always recommended to prefer AviSource over FFMS2 for AVIs with no or lossless encoding.

[Selur]

-> add a new issue to the bug tracker regarding this problem: https://code.google.com/p/ffmpegsource/issues/detail?id=108

[Farfie]

Hi there, I was directed to come here by the kind people in the "new" Avisynth thread. And mistakenly accusing them for my problem (I'm not exactly the biggest video guru,) they said I might have better luck here.
Anyway, encodes (8 or 10bit, x264) with open-gop are having a hard time opening with virtual dub, or megui's preview thing since the new avisynth, when trying to use FFVideoSource. What happens varies greatly: hanging, crashing, freezing, but in the end I cannot find a work around. One thing I've noticed is that if it's a smaller file it might work for a bit, but eventually start to exhibit problems, and hang indefinitely.
I just hope I'm talking to the right people this time, but if not, feel free to ask questions or point me somewhere else :)
My only goal is to better the software that I enjoy using, so here I am.

[Selur]

I'm no ffms developer, but here are some basic questions that you might want to answer:

0. What version of ffms are you using? (Have you tried other versions?)
1. You wrote you open x264 8/10bit with open-gop content, is it RAW content or does is reside in a container? If your streams are not raw streams in what container are they?
2. Do you use other filters than ffms.dll in your script? If you do, try to throw these out first to make sure this really is a ffms problem.
3. Are you using 32bit or 64bit Avisynth? What version of Avisynth are you using? (No clue what you call the ' "new" Avisynth thread'.)
4. What is the memory consumption of the Virtual Dub while opening and scrolling through your clip on your system?
5. Can you can reproduce the problem with a small clip ? If you can, it might help if you could share that clip with others so they too can try to reproduce the problem.
6. Is your system is overclocked in any way? (If so reset it to the default system speed, to make sure the problem is not caused due to a problem with your system.)
....
other users or a developer might have additional questions, but these should help to get a better grip on the problem.

Cu Selur