i'm curious if there is a way to detect if audio is missing or broken from a MOV video file. Recently there was a MOV file which was playable, but the audio was "missing".
I tried ffmpeg to get specific errors but it did not give me any. I only noticed that the audio bitrate was very low, 2kb/s. Apart from checking the audio bitrate is there a way to detect such broken video clips with ffmpeg?
The mising audio could have been caused by an unsuccessful copying of the file.
ffmpeg -i movie.avi -af volumedetect -f null -
This will print out a histogram telling you the volume in the file
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I am recording AVI files with Camtasia. For some reason the video stream length is 2,3-5 seconds less than the audio stream.
When I convert the video with ffmpeg from AVI to MP4 it cuts the audio to the video length.
Would duplicating the last frame until the end of the audio be a solution? If yes how can this be done using ffmpeg?
The important thing is to convert the AVI to MP4 using ffmpeg and keep the audio stream of the video complete.
Thank you.
Edit 1: This issue is automatically solved by ffmpeg 2.x somehow but ffmpeg 4.x will cut audio. With the same settings the old version converts correctly.
Edit 2: tpad helped. Thank you very much #kesh. I used
-filter_complex 'tpad=stop=NUMBER_OF_FRAMES:stop_mode=clone'
I tried to get the duration using ffprobe and multiplied the number of seconds with number of frames per second but it was not enough. For each video I had to increase that number with 100,150 frames.
The issue is I cannot detect the exact number of frames to tell tpad. I also tried
-filter_complex 'tpad=stop=-1:stop_mode=clone'
but it freezez while processing.
Is there any other option?
I have a bunch of mkv files, with FLAC as the audio codec and FFV1 as the video one.
The files were created using an EasyCap aquisition dongle from a VCR analog source. Specifically, I used VLC's "open acquisition device" prompt and selected PAL. Then, I converted the files (audio PCM, video raw YUV) to (FLAC, FFV1) using
ffmpeg.exe -i input.avi -acodec flac -vcodec ffv1 -level 3 -threads 4 -coder 1 -context 1 -g 1 -slices 24 -slicecrc 1 output.mkv
Now, the files are progressively out of sync. It may be due to the fact that while (maybe) the video has a constant framerate, the FLAC track has variable framerate. So, is there a way to sync the track to audio, or something alike? Can FFmpeg do this? Thanks
EDIT
On Mulvya hint, I plotted the difference in sync at various times; the first column shows the seconds elapsed, the second shows the difference - in secs. The plot seems to behave linearly, with 0.0078 as a constant slope. NOTE: measurements taken by hands, by means of a chronometer
EDIT 2
Playing around with VirtualDub, I found that changing the framerate to 25 fps from the original 24.889 (Video->Frame rate...->Change frame rate to) and using the track converted to wav definitely does work. Two problems, though: VirtualDub crashes when importing the original FFV1-FLAC mkv file, so I had to convert the video to H264 to try it out; more, I find it difficult to use an external encoder to save VirtualDub output.
So, could I avoid using VirtualDub, and simply use ffmpeg for it? Here's the exported vdscript:
VirtualDub.audio.SetSource("E:\\4_track2.wav", "");
VirtualDub.audio.SetMode(0);
VirtualDub.audio.SetInterleave(1,500,1,0,0);
VirtualDub.audio.SetClipMode(1,1);
VirtualDub.audio.SetEditMode(1);
VirtualDub.audio.SetConversion(0,0,0,0,0);
VirtualDub.audio.SetVolume();
VirtualDub.audio.SetCompression();
VirtualDub.audio.EnableFilterGraph(0);
VirtualDub.video.SetInputFormat(0);
VirtualDub.video.SetOutputFormat(7);
VirtualDub.video.SetMode(3);
VirtualDub.video.SetSmartRendering(0);
VirtualDub.video.SetPreserveEmptyFrames(0);
VirtualDub.video.SetFrameRate2(25,1,1);
VirtualDub.video.SetIVTC(0, 0, 0, 0);
VirtualDub.video.SetCompression();
VirtualDub.video.filters.Clear();
VirtualDub.audio.filters.Clear();
The first line imports the wav-converted audio track.
Can I set an equivalent pipe in ffmpeg (possibly, using FLAC - not wav)? SetFrameRate2 is maybe the key, here.
I’ve got a bunch of stereo files recorded for a documentary with a Zoom in 4 channel mode. Basically it’s sets of pairs of stereo file s— file A would be a stereo file with a lav or boom mike recording, file B of identical length would be a proper stereo recorded by Zoom itself.
Now I’m trying to convert all this into something I can correctly ingest into editing suite. Files A are a mess but I came up with a ffmpeg script which downconvert them to mono then reconvert them back to stereo (to get rid of inconsistensies). Now how do I merge two stereo files into a single WAV or AIFF file containing two separate stereo channels? I browsed around for any workflows and/or standards on that but can’t really find anything useful.
Any ideas on how to do that with ffmpeg (or anything else, really) would be appreciated!
Don't know if FCP-X reads multi track WAVs but you can output to a multi-track MOV.
ffmpeg -i file1.wav -i file2.wav -c copy -map 0 -map 1 file.mov
I'm using ffmpeg to extract audio from MPEG Transport Stream file recorded by DVB-S card. The command:
ffmpeg -i video.ts -vn audio.wav
The source file seems to be corrupted. I noticed the corruption happens from time to time, especially for videos longer than 1 hour. I've got errors like these:
[mp2 # 0x1bb5500] Header missing
Error while decoding stream #0:1
[mpegts # 0x17eaf40] Continuity check failed for pid 5261 expected 2 got 6
The problem is that the resulting audio.wav is shorter than the source video (40m33s and 40m59s accordingly). I'm looking for the way to preserve the original length in the resulting audio file.
I tried the recent ffmpeg under Windows and avconv under Ubuntu, output format was MP3 and WAV. For every case I've got the same results.
I didn't find whether it's possible to do it with ffmpeg however I found ProjectX - a tool which tries to fix the broken TS stream. Website: http://project-x.sourceforge.net/
With:
java -jar ProjectX.jar -demux my_video.ts
the stream is demuxed into audio and video files which are guaranteed to have the same length. I simply mux them back using ffmpeg.
I got some audio pieces in flv format. Each of them is about 10 seconds long.
My question is how to detect whether the audio pieces has "sound", in other words, sometimes the audio pieces has no sound even the size of it is not 0 byte, so how to find those broken/silent audio files by some linux tool/command?
Maybe ffplay can do this? any available advice?
If you want to quickly check if the stream is video only, audio only or if it contains both audio and video, try hexdump -C filename | head. The fifth byte contains information about the contents of the file.
0x01 - video only
0x04 - audio only
0x05 - audio + video
You can also try to play the file using VLC media player. There is a menu option that enables informational messages from the media being played back.