a/v is gradually getting asynch

[snowbeach]

Audio and movie is gradually getting asynch. I encoded the audio MP3 file with GraphEdit. The movie I encoded is Matrix.

a/v is synch at the beginning, but the more we get to the end the more a/v is asynch. I read the Doom9 guides regarding this problem, but I've difficulties to understand them. And I searched the forum about a similar problem, but haven't found anything yet.

Do you have any suggestions to get along with this problem? Thanks! :confused:

[kilg0r3]

before doing anything to the movie, you must find out if the problem is due to the movie file or the player or the player configuration.

do you have the same problem with other movies?

and we will need some more information:

what machine and os do you have?

what players do you use?

which is the sound device the player uses (wave out, default direct sound etc.)?

what application(s) did use to encode the movie?

what codec did you use?

which application did you use to mux it?

is it an avi or an ogm file?

well that's about it. if you feel that answering these questions is too much work try rereading doom9's guides:sly:

[snowbeach]

do you have the same problem with other movies?
Hmmm, the only problem I have with other movies is - that after forwarding or rewinding the video (moving the slider of Windows Media Player to the front or to the back) is much faster while the audio has the normal speed. But only for a few seconds (no more than 2 or 3), then it is synch. I heard that this is normal.

what machine and os do you have?
AMD 900MHz TB, Win XP Pro

what players do you use?
XP's Windows Media Player

which is the sound device the player uses (wave out, default direct sound etc.)?
Where I can find this out???

what application(s) did use to encode the movie?
VirtualDub 1.4.10

what codec did you use?
video: DivX 502
audio: LAME DirectShow filter for GraphEdit

which application did you use to mux it?
Nandub 1.0 RC2 lumafix

is it an avi or an ogm file?
movie: avi
audio: mp3

well that's about it. if you feel that answering these questions is too much work try rereading doom9's guides :sly:
Hey, I'm gratefull that you want to help me. :sly:

[kilg0r3]

ok

[edit] i just tried out aviframerate changer which you can download here (http://www.doom9.org/Soft21/Editing/avifrate110.zip). just download it open your avi with it and try it out. if you don't get along with it, give us a ring :) [edit]

possibility one is to load the muxed movie into nandub, selcet the video menu and then click on framerate. in the appearing dialog check the radio button 'change to ... frames per second' and then enter a new value. i would try increments of 0.005, ie. depending on which of what lacks behind audio or video you must speedup or slow down the speed of the video stream.
AFAIK it is impossible to check if a/v is in synch inside nandub. so you must save the file and then play it inside your media player. Don't forget to set video and audio to directstream copy.

you can do the same with avi framerate changer or a tool called avi information which is quite comfortable but does only work under win9x

possibility two would be to rencode the audio with besweet and just try again muxing the whole thing. i don't know if there are issues with the lame directshow encoder. i can only say, i never had any problems with besweet.

BTW if you used the vfapi converter for your encoding, i would suggest changing over to gknot anisnyth. for i myself had asynch problems with vfapi.

i don't know if i have been clearer than the doom9 guide. if not try rereading step 11 of the virtualdub procedures guide and repeat the steps in nandub. let me know about your resultss:)


THIS POST IS A MESS !! sorry for that

[snowbeach]

because of my university studies, i have no time to test it. but, i will do it soon.

i have another "petit question" ;): experienced that lame gives sometimes the wrong duration of the audio - encoded e.g. with besweet! :confused: why?????

[kilg0r3]

i have never experienced such a thing. For problem solving with besweet it is helpful to have the relevant logfile.

[snowbeach]

the duration of the audio regarding the log files has to be 1:58:18. i get this duration with the LAME DirectShow filter using graph edit. this way it's ok. but what i get with LAME using besweet is 48:49.
btw. both have all audio data included, but just the wrong duration in case of LAME using besweet.

ps. it must be something about the lame engine, because i have the same mystery encoding audio files from a music cd. not using besweet.

pps. media player shows me the duration, wich is stored in the data properties of the mp3 file, while winamp shows the correct duration (also for the lame encoded file using besweet). so the time stamp is wrongly set to the data properties by the lame engine (lame.exe or lame_enc.dll & co.)?? :confused:
as i said the directshow filter (lame_dshow.ax) has not the problem with the time stamp/duration!

[snowbeach]

the 2 log files...

BeSweet v1.3RC by DSPguru.
--------------------------
Using azid.dll v1.8 (b825) by Midas (midas@egon.gyaloglo.hu).
Using lame_enc.dll v1.28 (18/4/2002), Engine 3.92 <http://www.mp3dev.org/>.

Logging start : 06/24/02 , 17:38:45.

D:\GORDIA~1\besweet.exe -core( -input E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3 -output E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.mp3 -logfile E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.log ) -ota( -G max ) -azid( -L -3db -c normal ) -lame( --alt-preset 128 --scale 1 ) -profile( Gordian Knot 0.26 )

[00:00:00:000] +------- BeSweet -----
[00:00:00:000] | Input : E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
[00:00:00:000] | Output: E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.mp3
[00:00:00:000] | Floating-Point Process: Yes
[00:00:00:000] +-------- AZID -------
[00:00:00:000] | Output Stereo mode: Dolby surround compatible
[00:00:00:000] | Total Gain: 0.0dB, Compression: Normal
[00:00:00:000] | LFE levels: To LR -3.0dB, To LFE 0.0dB
[00:00:00:000] | Center mix level: BSI
[00:00:00:000] | Surround mix level: BSI
[00:00:00:000] | Dialog normalization: No
[00:00:00:000] | Rear channels filtering: No
[00:00:00:000] | Source Sample-Rate: 48.0KHz
[00:00:00:000] +-------- LAME -------
[00:00:00:000] | 'abr 128' preset is used
[00:00:00:000] +---------------------
[01:58:19:008] Gain of 13.5dB had been asserted to file.
[01:58:19:008] Conversion Completed !
[01:58:19:008] Actual Avg. Bitrate : 125kbps
[00:45:30:000] <-- Transcoding Duration
Logging ends : 06/24/02 , 18:24:15.



Gordian Knot log


24.06.2002 17:38:44: Job "Movie 1" started.

o Transcode AC3 -> MP3: E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
End of Job 1 (Movie 1).
_______________________________________________________________________________________________________________________

Gordian Knot 0.26.0.6
Encoding Job Data:

Type: EncAudio
Number: 1
Name: Movie 1
Platform: WinXP (5.1.2600).2

Audio:
mode: ac3 -> mp3
sourceFile: E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
wavFile: E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.wav
mp3File: E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.mp3
azid: "-L -3db -c normal"
lame: "--alt-preset 128"
delay: 0
interval: 2
preload: 500
mux: 1
recalc: 1
deleteWav: 1

none.


17:38:44: Started Transcoding Audio.

17:38:44: Using BeSweet to transcode E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3
Commandline: D:\GORDIA~1\besweet.exe -core( -input "E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.ac3" -output "E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.mp3" -logfile "E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.log" ) -ota( -G max ) -azid( -L -3db -c normal ) -lame( --alt-preset 128 --scale 1 ) -profile( Gordian Knot 0.26 )
18:24:15: Finished. Duration: 45 minutes, 31 seconds.

18:24:16: Audio Done.
18:24:16: MP3-File-Size is 108724 kb.
18:24:16: MP3 Audio File = E:\Movies\DivX\2\blow AC3 T01 3_2ch 448Kbps DELAY 0ms.mp3
Total Encoding Time: 45 minutes, 31 seconds.
24.06.2002 18:24:16: Job "Movie 1" finished.
_______________________________________________________________________________________________________________________

[kilg0r3]

sorry no idea what is going on here. perhaps some ...guru might be able to help.

Just to be sure that i got it right:

According to the logfile the besweet-encoded file is 1:58:19:008.
However, the besweet-encoded file only plays 1:48:49:xxx
The graphedit file is 1:58:18 long, i.e., it actually plays 1:58:18.
BTW how long is the ac3 file? One way to find this out is to trun the besplit commandline below and then examine the logfile in c:\

One thing, you said that you used LAME DirectShow filter and GraphEdit for your first attempt, and that the result was AV asynchronity. Right? Now, I don't know if, in your first attempt the audio was too long and hence lagged behind the movie or vice versa. However, if the graphedit-file was too long for the movie, the 'error' in the besweet-file might fix the asynchronity. So, have you tried to mux the 'buggy' besweet file with your movie? It might actually be the source file that has the wrong timestamps.

If this is so, you could also try to fix the ac3 file with besplit. The command line should be sth like this.


BeSplit -core( -input movie.ac3 -prefix e:\movie-fixed -type ac3 -logfilea "C:\BESPLIT.LOG" -a -ending -fix )


And still, there is the option to simply change the framerate of the movie, i.e., make it faster if the audio is too short or make it slower if the audio is too long. this, definitely, might be the quickest way. With Aviframeratechanger you don't even need to resave the whole movie. It simply changes the frame-rate value of an existing file.

Waiting for your results

[snowbeach]

in fact both files play 1:58:xx. that is what WINAMP shows, too.
the only difference is that media player shows different timestamps. media player shows the timestamps that are stored in the data properties (graphedit-encoded 1:58:xx and besweet-encoded 48min and 49sec).

the ac3 file plays in media player the correct 1:58:19!!!

sorry that i might coonfused you, but the now new mentioned "problem" has nothing to do with my other problem regarding AV asynchronity. maybe i should open a new thread and post this thread then to DSPguru?

as i said, i think this is a LAME based problem, because i think lame stores a wrong timestamp of the audio duration in the data properties of the mp3 file!!

[kilg0r3]

Yeah, why not. Yet, i don't think you need to adress anyone personally.

BTW - You may quote me :) - I can confirm the problem. Encoded one piece of music (Razorlame, Lame 3.92, -b 56 -m s -V 0 -B 320 -q 2). WMP 6.4 shows AND plays (!!!) 4:12 (BTW pitch is correct), Winamp shows AND plays 3:42 same with Cooledit. However the file properties tell me that it lasts 1:47 and, additonally, tells a far too high bitrate.

[pacohaas]

it's a known issue with microsoft that they don't understand VBR mp3's. you can have problems with their times in WiMP and explorer. It has to do with reading the beginning of the file and seeing the first bitrate used, then assuming the file is CBR, and calculating the time from the filesize(way to go M$).

Now, the original question had to do with async audio and video with the matrix, but your logfile appears to be for the movie Blow. Can we assume that you used BeSweet and the final muxed file was ok? If it is, great, if not, lets try to get to the bottom of it.

[kilg0r3]

I was allready suspecting that we were talking about old shoes. The problem seemed too obvious ...

[snowbeach]

Sorry for the confusion regarding the original question!
The movie, I have gradually getting asynch a/v problems with, is Matrix.
Video is encoded with an AVIsynth 1.06 script in VirtualDub 1.4.10 and DivX 5.0.2. Audio is encoded with GraphEdit 011008 and the LAME DirectShow filter. Muxed with Nandub 1.0 RC2 lumafix. The way of encoding is based on the guides of Doom9!
BTW. I do not think that there is any difference, if I use for muxing the audio file encoded by BeSweet or the audio file encoded by GraphEdit!

changing the framerate does not help (used AVI [FR] Changer 1.10 by ). Playing around with source displacement does not help, too. At the beginning of the movie it is synch, but later on it is asynch again (audio is faster then video) and vice versa! [B]As I said, audio and movie are gradually getting asynch.

There is an alternative method described in the "VirtualDub Procedures" (Doom9) guide using the Synch tool. But due to the fact that I do not have the WAV file anymore (which I need in this case) and that I do not have a wave editor for now, I will not be able to test it - not for the moment.
Or do you know other methods to solve my asynch problem using the MP3 file? Which software do I need then?

[pacohaas]

Originally posted by snowbeach
As I said, audio and movie are gradually getting asynch.As someone said before, this is because the rate of either the audio or the video is incorrect. So...it can be solved by either varying the video or audio rate. I remember having this problem a long time ago with an old versoin of FU, before it used to demux the ac3 track for you. I would make sure the beginning synced up, as yours does, then following this guide here (http://nickyguides.digital-digest.com/audio-synch.htm). The procedure was painstakingly long and tedius, so I was happy when new methods arose that made this process obsolete. Apparently, you have found a process where this is not obsolete, I suggest you seek out an alternative method that does not introduce this sync problem.

[kilg0r3]

changing the framerate does not help

what does this mean? what exactly does happen when you change the framerate? as far as i remember, you have to close avi-fr-changer before it actually applies the fr change, and of course, the movie must not be opened by any other application.

in order to test if the programm works at all, make a backup copy of your file and then apply a dramatic change to the fr, e.g., change it to 15 fps. you should be able to notice immediately if the movie plass slower then or not and if it also changes the speed of audio (which it shouldn't) or not.

in your case, since the audio is too fast the fr must be increased. try 25.002 fps or 25.005 max.

[snowbeach]

changed fps to 25.010 with avi-fr-changer.the program works fine.

the results:

beginning of the movie: a/v synch
middle of the movie: a/v asynch :angry:
end of the movie: a/v synch

this is a big, big §$%&/!%. I'm doing something wrong!

[kilg0r3]

This is so strange! Try to load the movie into nandub then press 'ctrl+r' in the opening dialog check the radio button labeled 'change to ... frammes per second' which is positioned under the segment 'source rate adjustment'. there, enter the 25.010 fps and then save your movie. video AND audio processing must be set to 'directstream copy' before saving.

[snowbeach]

Just to repeat the problematic situation. The movie Matrix (a/v), I encoded, is synch at the beginning and the more it comes to the end, the more it gets asynch (audio faster than video). A really great disaster!

Changing the frame rate does not help. The result is that at the beginning of the movie a/v are synch, at the middle a/v are asynch (video faster than audio) and at the end of the movie a/v are synch again. Very mysterious!
Playing around with audio interleaving a/v are synch at the end, but at the beginning a/v asynch (video faster than audio)!

Maybe some useful information about the VOB files I had:
Not all VOB files were 1.048.544 KB - some files were smaller (3rd and 4th - not quite sure).
e.g.
vts_01_1.vob 1.048.544 KB
vts_01_2.vob 1.048.544 KB
vts_01_3.vob 256.378 KB
vts_01_4.vob 918.489 KB
vts_01_5.vob 1.048.544 KB
vts_01_6.vob 432.233 KB

PS: Do you know a "wave editor" that allows to work with MP3 files and to save the file again as a MP3 with the LAME encoder, which means that the editor has to allow the integration of the LAME encoder (lame.exe)?

[kilg0r3]

i thought there might be a difference between changing the frame rate with aviFRchanger and virtualdub.

cooledit is a sound editor that can handle mp3 files but as any other editor only by de-recompressing.

sorry, i am at a complete loss of how to help you any more. if you had the audio source i would recommend to use an editor to lengthen it or to compress it to ogg vorbis and to then mux it into an ogg media-stream which is very easy to do but ...

just another guess: perhaps the problem lies in the avi file (broken index or something).

again sorry but i have to give up here. perhaps somebody else can take over?

[Swede]

What if you encode the audio CBR? Sometimes the VBR seems to create these 'voodoo'-problems.

[snowbeach]

the prob of a/v getting gradually asynch is that the rip of the dvd to hd has produced some errors and the vob files were "damaged".

after ripping the dvd again, everything was fine. a/v was synch! :)