Hi. Loving audacity as my favorite audio editor but I’m having a bug that I’m unable to fix so I’m coming here for help, please.
I’m getting a gradual desync introduced from the audio track I’m editing of longer video files (gameplay recordings). I use ffmpeg to extract the audio track from the mp4 video file, then when I process it in Audacity 2.0.3 and then save the resulting m4a file, the length of the track is shorter by about 2 seconds of a 53 minute recording. The audio/video sync is perfect before processing with Audacity.
To try to determine if it’s Audacity or ffmpeg introducing the desync I used ffmpeg to extract the audio track and then re-combine it back (with no Audacity processing) and there was no desync at all. So it’s not an ffmpeg issue. I’ve also tried saving the audio file in Audacity in several different formats with no success. This wasn’t usually a problem for me until recently when I reloaded Windows 7 onto my machine so maybe it’s something that happened with that. I don’t know. (okay it used to happen occasionally before but now it happens with every recording)
The audio tracks in question are 16bit, 48khz and I’m saving in the exact same format. It’s a roughly 2 seconds lag over 53 minutes. I’ve made sure that all sound devices are 16bit, 48khz in Windows 7 sound devices panel, especially the Xonar DG. Edit: I just checked and I see a ‘wave’ xonar dg audio device in the recording tab set to 44khz but I am not using that.
I need to use Audacity to process and combine my commentary into the video file (amplify, compress and remove noise of commentary audio).
Please help!!
GoG
System specs:
Win7 64 bit
Intel i7-3770
ASUS Xonar DG PCI sound card (drivers show as dated 4/11/2013 version 7.0.8.1821)
All newest drivers and all devices show as properly installed in the device manager.
What processing are you doing?
You can check the length of the Audacity track before and after processing by using the “Length” option in the Selection Toolbar and selecting the entire track.
That’s about a frame off per minute isn’t it? That’s the offset between American video 29.97 and 30. It could be complete coincidence, but I have enough bloody knuckles with those two numbers to recognize the symptoms. I have no idea where you would get that error since we haven’t actively used analog broadcast video for several years, now, but I do know sometimes the digital transmissions are at the older television framerates.
Back to reality for a second, I expect that if you did everything at 48 KHz, 16-bit that there should be little or no error.
The 44.1/48 thing is much too big an error for what you have. You notice that right away.
Games are not noted for their surgically accurate video standards. As an experiment, rip a long piece and do something really simple like Effect > Amplify > OK. still off? Are the two shows, before and after the same length?
You can tell Audacity to count minutes, seconds and television frames. It’s in the pulldowns at the timers at the bottom of the screen. That might be handy until we resolve this.
Thanks for replying so quickly! I’m not doing any processing besides loading into audacity and saving it. Now once I get the sync issue resolved I can start processing my commentary into it, but this is just the game audio track.
As soon as I open the file, Audacity reports the file length as 52m 43.392 seconds. When I open the same file in Goldwave, it reports it as 52m 44.971. I believe Goldwave is correct here.
In Audacity (2.0.3), after I save the track as an mp4 audio and recombine it into the MP4 game video file, the audio is ahead of the video by about 2 seconds at the end of the video. If I scroll through the video, I see that the audio desync increases over time.
Again, to reiterate… The game audio is in perfect sync with the video -before- I edit it with Audacity. So I’m rather sure that Audacity is the cause.
Exactly how are you extracting the audio with FFmpeg? Do you mean you are using a standalone version of FFmpeg to extract an M4A audio file from the video, then you load that M4A into Audacity?
Or are you importing the video file into Audacity and the standalone FFmpeg is just for a comparison test?
Which version of FFmpeg is the standalone, and where did you get it?
Which version of FFmpeg are you using in Audacity and where did you get it? The official version we recommend is 0.6.2 from Audacity Manual .
Which file, the M4A you exported from Audacity? Or an M4A extracted using standalone FFmpeg?
Using FFmpeg 0.6.2 in Audacity, if I export 53 minutes exactly of stereo audio to M4A, Audacity sees it when reimported as 52 minutes 59.950 seconds. From memory, AAC export in FFmpeg used by Audacity has had different bugs at different times according to whether libfaac or FFmpeg’s own AAC encoder was used for the particular build of FFmpeg.
In a nutshell however (aside from the two seconds being short), AAC is lossy and has silence padding like MP3 and many other lossy formats. So it is no use for video sync unless you are planning to adjust the delay for the edited audio in the video editor.
I would assume you have iTunes, or Goldwave would not be able to open the M4A. So for an immediate workaround I suggest you export as WAV (which won’t have length changes) then convert to AAC (if you need to) in iTunes. To convert in iTunes, follow the steps here except choose the AAC encoder in iTunes Preferences instead of the MP3 encoder.
Yes maybe that 29.97 framerate issue relates but I don’t know, yet. The video is in 1080p @30fps.VLC shows framerate at 30fps.
Once I start processing the audio, the length remains consistently off-sync but length doesn’t change at all.
Okay good idea. When I select ‘NTSC drop frames’ in the timers selector and have ‘length’ selected, Audacity says the length is 52m 43s+11 frames. However when I select ‘NTSC non-drop frames’ it says the length is 52m 40s+07 frames. That 3 seconds is about what I’m experiencing of desync by the end of the video where the audio runs ahead of the video after re-merging it back into the video file.
Note also that yes, this capture card (Live Gamer HD by Avermedia) does drop some frames when capturing. However, the audio and video sync is perfect in the captured video file, before processing with Audacity. I’ve tried VLC and Media Player Classic as well as YouTube to test and they are all consistently showing the desync. I’d be happy to upload the files to a file sharing site so that you can open try to reproduce the issue. The video file is big, 7.4GB. The extracted audio from the video file is 95MB. I’ve put the unprocessed audio file here - http://www.4shared.com/video/urfIkyjm/20130609235524audioonly.html (note that this is extracted from the video file with ffmpeg using the ‘audio copy’ codec. )
This is the command line used “ffmpeg.exe -i inputfilename.mp4 -acodec copy -map 0:0 outputfilenameaudioonly.mp4”.
ffmpeg version N-49757-g969039e Copyright (c) 2000-2013 the FFmpeg developers built on Feb 9 2013 20:33:22 with gcc 4.7.2 (GCC)
Again, I’ve done tests to determine that ffmpeg is probably not the cause of the desync. What I did was extract the audio and then re-insert it (with no processing from any program) and the sync remained perfect.
Drop-Frame actually drops time elements through the course of the show to make everything come out even. You’re not actually missing television frames in the show. If you number every frame in NTSC, 60:00:00 is not an hour-long show. That’s non-drop frame. If you use drop-frame, it is. Drop-Frame causes editing errors like this:
:14
:15
:17
:18
One of the frame numbers has been dropped. The errors are absolutely predictable and Drop Frame editors “know” where they are and automatically compensate for them. The obvious one is once every minute, but then there’s an extra one every ten minutes, etc.
The actual error is 1000/1001. It’s calculated intentionally to never come out even. The 10,000 foot view of Drop-Frame is a broadcast television signal whose visible errors never “gang up” and distort the picture.
I would guess that some parts of the system are running in 29.97 and others are running in 30:00. You would get that error if you presented an NTSC game and the system insisted it was in 30.
Gale let I’m responding to you but I’ll respond to kozikowski first please since this should be quicker.
kozikowski so since my video is truly 30fps then this NTSC frames issue that you brought up should not apply in this case? So all I’ve been able to resolve so far is that Goldwave seems to show the proper length while Audacity desyncs by two seconds.
I opened the mp4 audio file in Goldwave, then re-saved it, re-joined the m4a Goldwave saved back into the video and it seems that Goldwave does NOT introduce the desync. So I’m confident that I’ve narrowed the cause down to something in Audacity. (Unfortunately, I find Goldwave to be unintuitive so much that I can’t use it.)
Hi Gale. There’s too much to quote from your post so I’ll just answer you straight in line here. I posted my exact extracting method above. Yes, the standalone, command line version, of ffmpeg. Yes then I load that mp4 audio file into audacity. No I’m not importing the video file into audacity. Using the command line ffmpeg is my standard workflow method. It used to work fine, until recently.
The version of ffmpeg I use is in my previous post. I got it from the official site, one of the download links. Yes I’m using ffmpeg 0.6.2 within Audacity.
Which file, the M4A you exported from Audacity? Or an M4A extracted using standalone FFmpeg?
The mp4 that standalone ffmpeg command line extracts from the recorded video.
Using FFmpeg 0.6.2 in Audacity, if I export 53 minutes exactly of stereo audio to M4A, Audacity sees it when reimported as 52 minutes 59.950 seconds. From memory, AAC export in FFmpeg used by Audacity has had different bugs at different times according to whether libfaac or FFmpeg’s own AAC encoder was used for the particular build of FFmpeg.
Please try 53 minutes of stereo audio produced by another program, or use the download link I provided above of the exact file in question. Maybe it’s a compatibility issue between the audio track my video capture card creates and audacity, I don’t know. But this issue never used to happen, but there’s new software for the capture card that I’m using and Audacity has also been updated. Again, Goldwave shows the proper length of the track so looks like a new bug in Audacity…?
I’m willing to upload the 7.4GB video file, if needed.
In a nutshell however (aside from the two seconds being short), AAC is lossy and has silence padding like MP3 and many other lossy formats. So it is no use for video sync unless you are planning to adjust the delay for the edited audio in the video editor.
Well as I said the audio is in perfect sync in the original unedited video. Also Goldwave does not desync the track.
It almost has to be a framerate conversion issue (although I can’t figure out how). That error and the fact that the sound gets there first is normal. I’ve had tools report the wrong framerate to me.
I’m sticking with the “walks like a duck” theory until somebody proves me wrong. Koz
We’ve had file management programs that posted the wrong framerate in the file header. If the program paid attention to it, it gave one number and if it didn’t, you got another. One of those errors took us days to figure out.
You don’t need iTunes to get Goldwave to work, but you may need QuickTime Services – or equivalent. QT for Windows. You might want that anyway because Windows will not handle QuickTime Videos and several other video formats without it.
I’m trying to put together the pieces that are distributed across the previous 10 posts.
Have I got this correct and can you fill in some of the blanks:
You have a M4A video file of length <…blank…>
With standalone FFMpeg you convert the M4A video to M4A Audio format
You import the M4A audio file in Audacity - Audacity says that it is 52m 43.392 seconds.
You import the M4A audio file in Goldwave - Goldwave says that it is 52m 44.971 seconds
You export from Audacity in M4A format and import the file into your video editor application (which is <…blank…>)
The audio that is imported into the video editor is 52m 43.392 seconds. but the video is 52m 44.971 seconds.
Questions:
What is the length of the video in (1) Is it 52m 44.971 seconds?
Why do you use standalone FFMpeg to extract the audio from the video file and then FFMpeg in Audacity to import the M4A audio file?
Why not just import the audio from the video file with FFMpeg in Audacity or
extract the audio from the video file with standalone FFMpeg into an uncompressed format (such as WAV) and work with uncompressed audio until you import the track back into your video editor?
Okay I tried both of your ideas and neither worked. First was to drag & drop the mp4 audio/video file into audacity and sure enough it opens it, showing the incorrect length of 52m 43.393 seconds.
I then used command line ffmpeg to convert the audio track to WAV format and ffmpeg reported the following length:
Then when I open the WAV file in Audacity it shows the incorrect length of 52m 43.392 seconds. So while it was worth trying, it didn’t fix the issue.
Did any of you try opening or examining the mp4 audio file I linked above? It’s the same one. Also open it in Goldwave and you’ll see the length doesn’t match. If you use command line ffmpeg to convert the above mp4 audio file into WAV format it will also report time=00:52:44.97 with the following convert command line: c:ffmpeg.exe -i 20130609235524audioonly.mp4 test.wav (it will generate a lot of error messages I don’t know why but it will succeed).
Maybe the errors ffmpeg reports while converting the mp4 into WAV are a clue to the cause here:
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
[aac @ 000000000215a980] decode_band_types: Input buffer exhausted before END el
ement found
Error while decoding stream #0:0: Error number -1 occurred
decode_band_types: Input buffer exhausted before END element found
Error while decoding stream #0:0: Error number -1 occurred
Does that help diagnose the issue?
This file is just an example file. It happens with all of my videos now and I don’t know how to fix it except to generate some silence every ten minutes into the track. That’s not a good solution in the medium or long term because it’s time consuming and inaccurate.
Here are the answers to your questions…
You have a M4A video file of length 52m 52m 44.971 seconds.
With standalone FFMpeg you convert the M4A video to M4A Audio format = YES
You import the M4A audio file in Audacity - Audacity says that it is 52m 43.392 seconds. YES
You import the M4A audio file in Goldwave - Goldwave says that it is 52m 44.971 seconds CORRECT
You export from Audacity in M4A format and import the file into your video editor application NONE (video is already compressed from the capture card solution. I use ffmpeg to reinsert the audio track back into the video file)
The audio that is imported into the video editor is 52m 43.392 seconds. but the video is 52m 44.971 seconds. NO I don’t use a video editor. Audacity is my editor because I only need to edit the audio tracks. Re-compressing 2 hour video segments takes too much computer time.
Ideas?
Thanks again for all the help in trying to narrow this down and hopefully get me working again. Much appreciated!
Okay so how would I fix that? The issue still happens when importing the same audio as a WAV file. Goldwave doesn’t desync the audio. External FFmpeg shows the correct length. Only Audacity is having this issue.
Downloading from 4shared requires logging in, but I don’t have an account and I do not find their TOC acceptable. If you can upload the file to somewhere that it can be downloaded without the requirement to log in then I’ll be happy to take a look.
I have part of an answer. The file is faulty, possibly due to a problem in FFMpeg, or possibly due to dropped frames (not intentional ones) in the original video recording.
The file information states that the duration is 52 minutes 44 seconds, but SoX, avconv, FFMpeg, VLC, mplayer and Audacity all produce similar errors when trying to decode the file. Examples below:
avconv
[aac @ 0x9779d00] Input buffer exhausted before END element found
Error while decoding stream #0:0
SoX
[aac @ 0x88bea40] Input buffer exhausted before END element found
Audacity
Debug: Log: [aac @ 0xb20a2720] Input buffer exhausted before END element foun
VLC:
faad warning: Array index out of range
mplayer
[aac @ 0xb644c720]Input buffer exhausted before END element found
A:3149.5 (52:29.4) of 3164.9 (52:44.9) 0.2%
On attempting to decode the file, SoX gave up after the first 1 min 37.152 seconds. The other applications all completed the conversion, but with the above errors.
Foobar2000 reported the file to be 52:44.950 (151 917 590 samples) but on decoding showed similar errors:
Decoding error: Unsupported format or corrupted file, frame: 147639 of 148358
and again the decoded ouput was 52 minutes 43.392 seconds.
Question:
When you opened the file in Goldwave, did you do anything in Goldwave other than open the file? I ask because I suspect that Goldwave was simply reporting the file information and not actually decoding the file.
Maybe the errors ffmpeg reports while converting the mp4 into WAV are a clue to the cause here:
So FFMpeg is complaining there is something seriously wrong. This is not a graceful conversion.
Since this seems to be one unresolved spaghetti pathway after another, I want to back up a step. Where, exactly where, did the original file come from? Do you have software that grabs the game stream or possibly run screen capture software? Some combination? Koz
Yes it’s very likely that the audio track in the original file has dropped frames, because the capture card itself does drop frames. It’s an Audacity Live Gamer HD c985 card. The newest beta drivers also seem to have made the issue worse. RECentral V1.3.0.40 (Build 13052104), Beta Release with drivers C985_Drv_V3.3.x.38_Bet. I doubt the issue is caused by ffmpeg but can’t be sure, yet. I’ll work to determine that tomorrow.
In Goldwave I was able to import the file, and then export it to m4a format it, then using ffmpeg.exe I merged the saved m4a file back into the video file where it was in perfect sync. However I find Goldwave unintuitive enough that it would take me quite a while to learn to use it. Also I’ve come to like Audacity for many reasons. I’ve done so much audio editing in it that it’s almost like driving a car or riding a bike, second nature… So to speak.
Alright well I will post this bug on the Avermedia beta drivers forum and see what happens. I can’t expect you guys to spend time trying to see if it could be imported correctly with some updated code, but it would be nice if possible. xD I doubt they can fix the dropped frames issue but the Avermedia developers might be able to save a more edit-friendly audio track to compensate for it.
