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| Antonjo |
| Posted: Jan 2 2013, 10:45 PM |
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I am trying to edit .flv files in VirtualDub x64.
Files are typical youtube files with AVC/AAC encoding; this according to MediaInfo.
Using FFInputDriver_64.vdplugin, when I click play the first time, I get an error "while seeking file". The second time I can play the file, which eventually crashes on the very last frame.
With MP4 containers and AVC/AAC encoding everything is smooth.
Using FLV64.vdplugin by fccHandler I have no crashes but the AAC audio codec is reported as missing. The error is "No audio compressor could be found to decompress the source audio format (source format tag: 0x00FF)". So I can work on FLV files only with no-sound option. Note that MP4 AVC/AAC files can be played in Windows Media Player, so this AAC codec is present in my system. And in fact the AAC decoder should be part of the Microsoft Media Foundation.
With FLVs based on MP3 codecs, using the FLV64 plugin, everything is smooth.
This behaviour happens for all the files with the said codecs/containers, not for a specific one. I suppose that I could somehow reconstruct some index file and use the FFInputDriver plugin without crashing.
Note:
I am on Win7 x64.
I set FFDShow the libavcodec parameter for FLV1, VP6F in VfW and for the AAC decoder as well.
Any chances of working with FLV AVC/AAC files in VD?
Antonio |
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| dloneranger |
| Posted: Jan 2 2013, 11:15 PM |
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Virtualdub uses vfw video codecs and acm audio codecs
FFDShow's audio codec isn't an acm one, and neither is Microsoft Media Foundation
You'll need an aac acm codec for the FLV plugin to use - fccHandlers site has one (mirror http://gral.y0.pl/~fcchandler/)
It's also in my sig, with an installer for 32/64 bit
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| TCmullet |
| Posted: Jan 3 2013, 09:28 PM |
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Lone,
I'm trying to use the ACCACM. Have Windows 7-64. Using Vdub 1.10.2-32. Have installed ffdshow. And the ACCACM codec from the new "fccHandler's Stuff", http://gral.y0.pl/~fcchandler/.
It plays fine, whether playing or Vdubbing, except when I get to a certain spot in my 90 min. program, Vdub aborts with the message: "The audio codec reported an error while decompressing audio data. Error code: 1 (MMSYSERR_ERROR)" The certain spot was about 10 minutes in. Vdub didn't crash. I simply "okay"d the error message and it was stopped at that spot. I could move back in time and it plays fine. I can skip forward a few seconds and it will play fine. And the "bad spot" totally plays fine in some FLV players.
Can you please tell me what's going on?? (And what we should do next?) |
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| dloneranger |
| Posted: Jan 3 2013, 09:51 PM |
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The codec got to something it didn't like and died
Usually that's something like a corrupted file (but can just be a bug)
Your options are
1)
leave it alone and just direct stream copy the audio
hopefully a player will have better luck with it and you might just get a 'click' in the audio
players are a lot more lenient in what they do with bad data
(also, no good if you don't want aac though)
2)
export the audio to a file and then try some other programs that may handle converting it better
eg a cmdline program like faad may decode it http://www.rarewares.org/aac-decoders.php#faad2-win (FAAD2v20100614 CVS snapshot for Win32)
Then replace the audio in the video with the newly converted audio - compress it how you will
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| TCmullet |
| Posted: Jan 7 2013, 04:57 PM |
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Thanks, Ranger. But could you please help a bit more?
I did save of .wav to "has-acc.wav", direct stream copy. I run "faad -i has-acc.wav", and it comes back saying it's "RAW".
I rename file from "has-aac.wav" to "has-aac.aac", in anticipation it will create a .wav file.
I run "faad has-aac.aac" and it comes back with "Error: Channel coupling not yet implemented" "0% decoding..."
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| dloneranger |
| Posted: Jan 7 2013, 05:10 PM |
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Did you 'save as wav' or 'export raw audio'
It's the export one you want, save it as something.aac and faad should decode it fine
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| TCmullet |
| Posted: Jan 9 2013, 07:56 PM |
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Tried this. It DID start to work. Gave the word "RAW"; printed a little chart showing config as "2 ch"; 00 is Left front, 01 is Right front". then a line that dynamically counted up the percentage done. 1%, 2%.... then paused at 33% (apparently where there is the seeming glitch in the file) It then says "Error: Gain control not yet implemented", and stops.
I can play the video through the "glitch point" fine in several players. But Vdub aborts as I previously reported. I captured this video file through the fine program Replay Media Capture 4.4. It's own conversion-to-AVI routine bombed part way through, with the complex log seeming to indicate a problem with audio. I downloaded the file again and the 2nd copy was identical to the first, and with same errors. Therefore I assume that I do have my copy identical to what is on their server. And as their copy plays fine, I don't think they would consider the file to be in error. Yet noone but a piece of player software can decode the audio at the glitch point.
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| dloneranger |
| Posted: Jan 9 2013, 08:14 PM |
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One other you could try is ffmpeg
It's a cmdline on and the command would be
ffmpeg.exe -i "c:\myfile.aac" -acodec pcm_s16le "c:\mynewfile.wav"
If that fails, then I'd go with the opinion of every program you've tried "it's a bad file"
"Plays" doesn't really count, most players will skip over any errors with errors, but may skip sections or make pops/crackles
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| Abrazo |
| Posted: Jan 9 2013, 08:41 PM |
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Sorry to intervene dloneranger, but what do you think about having a try with the DirectShow inputdriver plugin for VirtualDub ?
In combination with LAVSplitter or FLVSplitter it should be possible to use ( ffdshow ) Directshow codecs for Audio and Video ?
One never know if that would be a solution ?
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| dloneranger |
| Posted: Jan 9 2013, 08:44 PM |
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Could well be, depends on what the codec does when it hits an error
On the other hand it's a different encoder, and one of the many might get past the error where the others can't
Certainly doesn't hurt to try
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| TCmullet |
| Posted: Jan 9 2013, 09:00 PM |
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Ranger, I have more info for you. Replay Media Catcher is using ffmpeg to do my conversion. The log gets huge, but I can paste here the last few good lines right before it goes bad, then I skip to the end (it's mostly greek to me):
| CODE |
Converting - frame=165485 fps=120 q=0.0 size= 7344203kB time=5521.92 bitrate=10895.4kbits/s
Converting - frame=165585 fps=120 q=0.0 size= 7345545kB time=5525.21 bitrate=10890.9kbits/s
Converting - frame=165686 fps=120 q=0.0 size= 7347295kB time=5528.59 bitrate=10886.9kbits/s
Converting - frame=165786 fps=120 q=0.0 size= 7348615kB time=5531.93 bitrate=10882.3kbits/s
Converting - Last message repeated 2 times
Converting - [aac @ 0x2ab8070]Number of bands (44) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - frame=165891 fps=120 q=0.0 size= 7349990kB time=5535.38 bitrate=10877.5kbits/s
Converting - [aac @ 0x2ab8070]Prediction is not allowed in AAC-LC.
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (45) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (39) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (46) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (72) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (81) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (64) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (45) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]invalid band type
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (48) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (81) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (68) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (49) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (64) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Noise gain (-249) out of range.
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (42) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (40) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (38) exceeds limit (36).
Converting - Error while decoding stream #0.1
.
. (skipping way down to where the RMC job aborts...)
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Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (72) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (52) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - [aac @ 0x2ab8070]Number of bands (81) exceeds limit (36).
Converting - Error while decoding stream #0.1
Converting - Resampling with input channels greater than 2 unsupported.
Converting - Can not resample 7 channels @ 48000 Hz to 2 channels @ 48000 Hz
Complete with errors. Double click to view. (7.01 GB)
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| dloneranger |
| Posted: Jan 9 2013, 09:04 PM |
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That certainly doesn't look good
Guess your last option is the directshow plugin 
--------------------
MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| dloneranger |
| Posted: Jan 9 2013, 09:07 PM |
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Though you might want to try with the latest ffmpeg
It's very much a work in progress, in that it's being modified a lot and the latest version came out just the other day
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MultiAdjust JoinWav WavNormalize FFMPeg Input Plugin v1827 UnSharpMask
Windows7/8 Codec Chooser
All FccHandlers Stuff inc. Installers for acm codecs AAC, AC3, LameMp3 |
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| TCmullet |
| Posted: Jan 9 2013, 09:16 PM |
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Will look at all options given.
Wanted to add that I just played the offending region of the timeline in the flv player that RMC provides. I could discern no glitch. Only possibility is that maybe a syllable got omitted. But the back ground noise never wavered. (It's in a noisey gymnasium.)
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| TCmullet |
| Posted: Jan 9 2013, 09:22 PM |
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Ranger, sorry to be so dumb, but it's an ffmpeg zoo out there, a zillion versions seemingly. Can you please give a direct link to what you think I should get?? (Again, sorry.) Am running Win 7 64, but am trying to keep most of software 32-bit so everything works. |
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