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?
Related
I have a recording as a collection of files in mpegts format, like
audio: a-1.ts, a-2.ts, a-3.ts, a-4.ts
video: v-1.ts, v-2.ts, v-3.ts
I need to make a single video clip in mp4 or mkv format.
However, there are two problems:
audio and video segments have different duration each, number of audio segments is different from number of video segments. Total duration of audio and video matches. Hence I can not concat pairwise audio video segments using mpeg and merge them afterwards, I get sync issues increasing progressively
few segments are corrupt or missing. So if I concat audio and video streams separately using ffmpeg I get streams of different lengths. When I merge these streams using ffmpeg I have correct a/v synchronization until time when first missing packet is encountered.
It's OK if video freezes for a while or there is silence for a while as long as most of the video is in sync with audio.
I've checked with tsduck and PCR seems to be present in all audio and video segments yet I could not find a way to merge streams using mpegTS PCR as sync reference. Please advise how can I achieve this.
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'd like to sync an audio and video file together so that they end at the same time. Is there an easy way to do this with ffmpeg?
The only ideas I have are 1) finding the lengths of the two (which is annoying) and delaying the shorter one by the difference between the two lengths, and 2) reversing the two, syncing them, then reversing again. There's got to be a better way...
After finding the lengths of the two and calculating the required delay, you can run
ffmpeg -i video -i audio -af "adelay=6500|6500" -c:v copy output
Here, both the channels of a stereo audio stream are delayed by 6500 milliseconds i.e. 6.5 seconds. The video stream is copied over without recompression.
I have a .mov file (codec = motion jpeg) that has an audio stream that includes small pulses at every second.
When I convert this file to mp4 using ffmpeg I notice that all my pulses are now off by one frame.
I simply used "ffmpeg -i source_file.mov target_file.mp4"
Here is an image of the comparison between the audio signals:
A1 is the original audio (.mov) and A2 is the mp4 output audio of ffmpeg.
As you can see the pulses are one frame late compared to the original.
I know that the h264 codec is lossy but one frame offset seems like a big loss if you ask me.
Is there any option I could use with ffmpeg to have a better audio stream ?
Here is the input file: https://www.dropbox.com/s/6y5g7lo5dvu0ub1/BBB_09_tree_trunk_009_ANIM_001.mov?dl=0
Here is the output file:
https://www.dropbox.com/s/10zuzwn0qs8l853/BBB_09_tree_trunk_009_ANIM_001.mp4?dl=0
If you copy the audio over, you shouldn't get the shift.
ffmpeg -i source_file.mov -c:a copy target_file.mp4
I've been working on this issue for my own needs and my file format has to be mp4. I'm working from mxf files. I've tried several options and found this to give the most accurate result (I've removed specifics for simplicity):
ffmpeg -ss 00:00:00.021 -i "input.mxf" -itsoffset -0.044 -i "input.mxf" -c:v libx264 -c:a aac -map 0:a -map 1:v "output.mp4"
Starting the first file at 21ms and mapping it as the audio, then shifting the video back 44ms gave gave me the most accurate sync (within several samples). I don't know why 22ms wasn't as accurate (when that's what the primer sample issue seems to equate to) and I found nothing that allowed me to work more granular, in samples. A filter with a PTS offset had no affect. Perhaps it works differently with different file formats. It's also worth noting that the same command without the -itsoffest gave the same sync result with one difference; the video stream duration was 1 frame and 1ms off the audio and container durations. With the -itsoffest, the durations were only 1ms different. You can use 22ms to achieve an accurate duration, but check your sync, it might be out that slightest bit more.
Also worth noting that I stumbled across some developer commentary on the -itsoffset tag which clarified that it doesn't work on audio, it works on video. It seems like the answer above is suggesting to map the offest against the audio, which apparently is not how the function is built to work. https://trac.ffmpeg.org/ticket/1349
try mpeg2 audio: -acodec mp2 it worked for me
I know that there are a million ways to download a video from youtube and then convert it to audio or do further processing on it. But recently I was surprised to see an app called YoutubeToMp3 on mac actually showing "Skipping X mb of video" and supposedly only downloading the audio from the video, without the need to use bandwith to download the entire video and then convert it. I was wondering if this is actually correct and possible at all because I cant find any way to do that. Do you have any ideas ?
EDIT:
After some tests here is some additional information on the topic. The video which I tried to get the audio from is just a sample mp4 file from the internet:
http://download.wavetlan.com/SVV/Media/HTTP/MP4/ConvertedFiles/MediaCoder/MediaCoder_test6_1m9s_XVID_VBR_306kbps_320x240_25fps_MPEG1Layer3_CBR_320kbps_Stereo_44100Hz.mp4
I tried
ffmpeg -i "input" out.mp3
ffmpeg -i "input" -vn out.mp3
ffmpeg -i “input” -vn -ac 2 -ar 44100 -ab 320k -f mp3 output.mp3
ffmpeg -i “input” -vn -acodec copy output.mp3
Unfortunately non of these commands seems to be using less bandwith. They all download the entire video. Now that you have the video can you confirm if there is actually a command that downloads only the audio stream from it and lowers the bandwith usage? Thanks!
After a lot of research I found out that this is not possible and developed an alternative approach:
Download the mp4 header
Parse the header and get the locations of the audio bytes
Download the audio bytes with http range requests and offsets
Assemble the audio bytes and wrap them in a simple ADTS container to produce a playing m4a file
That way only bandwidth for the audio bytes is used. If you find a better approach of doing it please let me know.
For a sample Android APP and implementation check out:
https://github.com/feribg/audiogetter/blob/master/audiogetter/src/main/java/com/github/feribg/audiogetter/tasks/download/VideoTask.java
FFmpeg is capable of accepting an URL as input. If the URL is seekable, then FFmpeg could theoretically skip all the video frames, and thus it would need to download only the data for the audio stream.
Try using
ffmpeg -i http://myvideo.avi out.mp3
and see if it takes less bandwidth.