Hi
I try to open AVC/AAC-encoded videos in VirtualDub but I don't get it to work. ffdshow is installed for decoding like I read it in other threads here, but it doesn't change I can't open those videos.
When I try to open it with an AviSynth-Script, I get this message:
(German part: "File can not played. The format is not supported")
When I try to open the file with the DirectShow Input Driver for VirtualDub:
System/Tools:
Windows 7 x64
VirtualDub 1.9.10 x32
ffdshow tryouts x32 rev. 3562 (08.09.2010)
AviSynth 2.5.8
ffdshow:
VFW:
H.264/AVC - ffmpeg-mt
Video Decoder:
H.264/AVC - ffmpeg-mt
Audio Decoder:
AAC - libfaad2
avs:
DirectShowSource("test.flv")
Media Info:
Format : Flash Video
Overall bit rate : 136 Kbps
Video
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L1.1
Format settings, CABAC : Yes
Format settings, ReFrames : 3 frames
Muxing mode : Container profile=Unknown@1.1
Duration : 3mn 36s
Bit rate : 58.1 Kbps
Width : 320 pixels
Height : 240 pixels
Display aspect ratio : 4:3
Frame rate mode : Constant
Frame rate : 25.000 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.030
Stream size : 1.50 MiB (43%)
Audio
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Format settings, SBR : Yes
Format settings, PS : No
Duration : 3mn 36s
Bit rate : 72.8 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 KHz
Bit depth : 16 bits
Stream size : 1.88 MiB (53%)
I hope you can help me, because I often have to edit AVC videos!
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Last edited by klischee; 10th Sep 2010 at 11:19.
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did you use preferred filter tweaker for win7 , to set ffdshow for decoding ? you have to override the default MS decoder
If you can't "fix" your directshow system , you can use another source filter e.g. ffmpegsource2
http://code.google.com/p/ffmpegsource/ -
I tried the Filter Tweaker and set H264 & AAC to ffdshow, but it still does not work.
ffmpegsource2 has no inf or installer or something, don't know how I can tell AviSynth and VirtualDub how to use it. -
You need an flv splitter that supports avc/aac if you are using directshow
You can find one here:
http://sourceforge.net/projects/guliverkli2/files/DirectShow%20Filters/
and register it with radlight filter manager, or through graphstudio
If you render the flv in graphstudio or graphedit , what does it show ? After you've registered the flv splitter , what does it show? do the pins connect ?
For ffmpegsource2, download the zip file, and unzip the plugins into avisynth\plugins directory (specifically the .avsi and .dll are required) . Full instructions are included as usual in the zip file
FFMpegSource2("video.flv", atrack=-1) -
Directshow Input driver is 32bit. You need to use Virtualdub 32bit with all 32bit plugins and install 32bit codecs in the SysWOW folder.
You could use FLVExtract to demux the h264/aac streams and use MP4Box or MKVmerge to mux and cut the files without recoding or better...
Use gdsmux.exe (part of Haaili splitter) to copy the streams to a TS container and then use H264TS Cutter to cut the files without recoding. Then use gdsmux.exe to mux the streams into an mkv or mp4 container. No recoding is done in any of this process.
EDIT: I was hoping you could use the ffmpeg copy command to batch convert the FLV files to TS files but it did not work for me.Last edited by DarrellS; 10th Sep 2010 at 14:06.
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Edit: I've got it to work! On the flv splitter page there was descriped how to register it:
"Extract the .ax file to a folder and then Start->Run and type regsvr32 "C:\Locationto\FLVSPlitter.ax" to install it."
Now I can open AVC encoded videos through AviSynth in VirtualDub! Thanks!
Original post (outdated ^^):
With RFM I get an error message when I click on "register"-button and choose the FLVSplitter.ax
When I try to render it says "cannot render file". How I can register the splitter I did not found out. I googled but did not found the solution.
Ok, this works fine, but Audio is missing of course now... . It would be better to handle AVC through direct show, because it would be much more comfortable to create avs and open them (with scripts). Otherwise I would have to check the codecs of each video.
Thanks for your help, I hope you can help me further to install the FLV Splitter.
@DarrellS
Thanks, but I think/hope there's a easier way to open up AVC-videos
I know I can convert them to other formats, but I would like to skip that step.Last edited by klischee; 10th Sep 2010 at 14:20.
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Edit: I've got it to work! On the flv splitter page there was descriped how to register it:
"Extract the .ax file to a folder and then Start->Run and type regsvr32 "C:\Locationto\FLVSPlitter.ax" to install it."
Ok, this works fine, but Audio is missing of course now... . It would be better to handle AVC through direct show, because it would be much more comfortable to create avs and open them (with scripts). Otherwise I would have to check the codecs of each video.
are you sure there's no audio? did you specify atrack=-1 ?
The problem with directshowsource, is that it's not frame accurate (sync errors when editing), and prone to whatever filters you have in your directshow subsystem.
On the contrary, FFmpegsource is independent of the codecs , splitters in the video , and directshow requires you to check every video or at least ensure you directshow system is working with filters (such as you are doing now) . FFmpegsource2 also indexes your file, so it has higher reliability when editing or using temporal filtering than directshow .
If you are only doing a straight encode, directshowsource() is fine . If you're adamant on using directshow, consider using DSS2() instead, which is frame accurate when used with video in a container. It only serves video, so you need audiodub() to load the audio together (with an audio source filter)Last edited by poisondeathray; 10th Sep 2010 at 14:34.
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Thanks for all those infos, I will keep them in mind!
Mainly, I just want to cut, convert and copy out the audio of videos, the only filter I use sometimes with VDub is null transform/crop or resize. The first tests look good, edit, oops: audio is still not working. It's AAC. Any idea?
VirtualDub can export the stream to wav, but it can't reencode it f.e. to lame MP3.Last edited by klischee; 10th Sep 2010 at 14:40.
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Ok now you have several options
Just keep that stuff in the back of your mind, because most of the time problems can be directly attributed to directshow subsystem (splitter, codec, filters) - like they were for you today -
Yeah, I will
Did you read my edit? I was too early with saying audio is okay, my test video was a FLV with mp3, but FLV with AAC does not work for audio reencoding -.- Is there something like AAC-Splitters?
Btw: Thanks for investing so much of your time for meLast edited by klischee; 10th Sep 2010 at 14:56.
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In vdub, if you use file=>file information is it passing through audio ?
For encoding to MP3 in vdub , you can install lame ACM . There are special instructions for installing in 32 bit in 64-bit OS
https://forum.videohelp.com/threads/271712-Lame-MP3-Encoder-on-Vista-x64Last edited by poisondeathray; 10th Sep 2010 at 14:48.
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I already have installed lame, it works fine with other source input audio formats.
This is what VirtualDub tells me in file information:
When I save the wav (File --> Save WAV) it's playable but PCM. You can see the media info of the input video in the 1st post. I can't compress the audio in VirtualDub to lameMP3 or other formats. Any idea? -
Ok. But when the input is fine, why does VirtualDub not let me compress the audio?
(Audio --> Full processing Mode --> Compression...).
Lame compression works fine for other input files like wma, ogg, mp3, ... only AAC is not working. -
Ahh, you're a genius! Thanks so much, I would never have get this to work without you.
VirtualDub is able to convert to stereo, then it works with lame, cool -
Another option is to convert 5.1 to stereo in the avisynth script as well
a = DirectShowSource("video.flv")
fl = GetChannel(a, 1)
fr = GetChannel(a, 2)
c = GetChannel(a, 3)
lfe = GetChannel(a, 4)
sl = GetChannel(a, 5)
sr = GetChannel(a, 6)
l_sl = MixAudio(fl, sl, 0.2929, 0.2929)
c_lfe = MixAudio(lfe, c, 0.2071, 0.2071)
r_sr = MixAudio(fr, sr, 0.2929, 0.2929)
l = MixAudio(l_sl, c_lfe, 1.0, 1.0)
r = MixAudio(r_sr, c_lfe, 1.0, 1.0)
MergeChannels(l, r)
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I try to keep avs' as simple as possible, because I generate them with a batch script, so I don't have to create them by hand.
At moment, I click f.e. a .flv-file, "open it..." with the batch-script, then it generates a videoname.avs with "DirectShowSource("filename.ext") ", opens it in VirtualDub, and deletes the avs. So I don't have to do anything :O I'm not professional enough to detect the video settings for each file and generate a complex avs
<--big noob -
@DarrellS
Thanks, but I think/hope there's a easier way to open up AVC-videos
I know I can convert them to other formats, but I would like to skip that step. -
Sorry I should have clarified. It depends on what style of flv you are using, and if they are cut or not. It also depends on how your ffmpeg binary was compiled
If you want it to work consistently for most things, you need a special ffmpeg build that is muxer/demuxer only, that can properly handle h264 streams . Note, this is just a special re-wrapping build (it won't encode to other things)
http://forum.doom9.org/showthread.php?t=152419
for %%a in ("*.flv") do ffmpeg -i %%a -vcodec copy -acodec copy -f mpegts -vbsf h264_mp4toannexb %%~na.m2ts
For whatever reason, "m2ts" works more consistently than "ts" (I suspect it's because m2ts uses 192byte packets, whereas ts uses 188byte packets , the extra bytes are for timecode information)Last edited by poisondeathray; 10th Sep 2010 at 18:38.
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Hi
I have again a problem with AVC/AAC
AviSynth or VirtualDub seems to change AAC tracks to 5.1, so I can't reencode them with VirtualDub.
The source videos are 2ch Stereo, but after opening them with DirectShowSource("video.ext"), VirtualDub tells me the streams are 5.1, and it is not able to reencode them (not with lame, not with AC-3, even not when I use the stereo-conversion). I really have no clue why the video editors think the audio stream is 5.1. It only happens with almost all AAC encoded audio streams.WMA, MP3 and others stay Stereo. Any idea?
Media Info about the original source video:
Audio #2
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Format settings, SBR : No
Codec ID : 40
Duration : 12mn 50s
Bit rate mode : Constant
Bit rate : 80.0 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 32.0 KHz
Stream size : 7.44 MiB (3%)
Language : English
Last edited by klischee; 11th Sep 2010 at 12:00.
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Omg, are you never asleep?
Here's the complete report of media info:
General
Complete name : test.mov
Format : MPEG-4
Format profile : QuickTime
Codec ID : qt
File size : 126 MiB
Duration : 17mn 10s
Overall bit rate : 1 027 Kbps
Encoded date : UTC 2009-12-24 20:18:35
Tagged date : UTC 2009-12-24 20:18:48
Writing library : Apple QuickTime
Video #1
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Main@L4.2
Format settings, CABAC : Yes
Format settings, ReFrames : 4 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 17mn 10s
Bit rate mode : Variable
Bit rate : 897 Kbps
Width : 640 pixels
Height : 360 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 29.970 fps
Standard : NTSC
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.130
Stream size : 110 MiB (87%)
Language : English
Encoded date : UTC 2009-12-24 20:03:18
Tagged date : UTC 2009-12-24 20:18:48
Color primaries : BT.601-6 525, BT.1358 525, BT.1700 NTSC, SMPTE 170M
Transfer characteristics : BT.601-6 525, BT.601-6 625, BT.1358 525, BT.1358 625, BT.1700 NTSC, SMPTE 170M
Matrix coefficients : BT.601-6 525, BT.1358 525, BT.1700 NTSC, SMPTE 170M
Audio #2
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Format settings, SBR : No
Codec ID : 40
Duration : 17mn 10s
Bit rate mode : Constant
Bit rate : 128 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 KHz
Stream size : 15.4 MiB (12%)
Language : English
Encoded date : UTC 2009-12-24 20:03:18
Tagged date : UTC 2009-12-24 20:18:48
Is remux still possible or needed when there's only one real audio stream? I never muxed, what tool do you prefer? I'm really not into all that video/audio stuff Never remuxed, demuxed and so on. -
assuming mediainfo is correct , you shouldn't need to remux
if you type info() in the script, how many channels does it report (when you preview the script in vdub or avsp, or a media player ?)
the problem is likely ....guess what.... your directshow decoder
make sure it's not setup to upsampling 2.0 => 5.1
I noticed this last video is .mov wrapped. They can be problematic depending how you have your directshow filters setup. You have to fix your directshow filters to use directshowsource()
You can use tools such as graphstudio to determine what directshow filter chain is currently being used
or you could use ffmpegsource2 as suggested earlier, to avoid directshow issues . Repeat the above experiment with ffmpegsource2("test.mov", atrack=-1) and use info() and see how many channels it reportsLast edited by poisondeathray; 11th Sep 2010 at 12:45.
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Players and Vdub show me on screen: Audio Channels: 6
Good idea! I had activated the matrix in ffdshow audio (2.0 to 5.1). But after deactivating it, it's still 6ch when I open videos with avisynth.
It seems to use AC-3 Filter, not ffdshow. AC-3 is set to output stereo.
You can have a look to graph edit when you open the attachement capturing. Possible you detect something?
With this script it works! It detects 2ch and is recompress the AAC to lame MP3 without any problems. Great! I will try to get into it what formats are opened the best with which method. It's so complex, uh!
BIG thanks again for your help. -
I will try to get into it what formats are opened the best with which method.
FFMS2 should work on everything (at least all the common video formats)
It does have a frame rate bug with some transport streams (.m2ts, .ts) so you might want to fix it in avisynth (e.g. with assumefps) , or use another source filter to load if your source is transport stream. -
BTW , when you use graphstudio, you open the native file, not the avs, to see the filter chain
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