I have 30 mkv files which have multiple audio streams and multiple subtitles.
For each file I am trying to: extract the dutch audio and subtitles from that file (25fps)
And merge it with another mkv file (23.976216fps)
With this command it seems like I extract the dutch audio and subtitles into a mkv:
ffmpeg -y -r 23.976216 -i "S01E01 - Example.mkv" -c copy -map 0:m:language:dut S01E01.mkv
But it does not adjust the fps from 25 to 23.976216.
I think I am going to use mkvmerge to merge the two mkv's, but they need to be the same framerate
Anyone knows how I could make this work? Thanks! :)
The frame rate of the video has nothing to do with the frame rate of audio. They are totally independent. In fact there is really no such thing as audio frame rate (well, there is, but that’s a byproduct of the codecs). If you are changing the video frame rate by dropping frames, you are not changing the videos duration, hence you should not change the audios duration. If you are slowing down the video, you must decode the audio, slow it down (likely with pitch correction) and re-encode it.
Something like this would change the audio pitch from standard PAL to NTSC framerate (example valid if your audio track is the 2nd in list, -check with ffmpeg -i video.mkv and see-)
ffmpeg -i video.mkv -vn -map 0:1 -filter:a atempo=0.95904 -y slowed-down-audio-to-23.976-fps.ac3
(23976/25000 = 0.95904 so this is the converted frame rate needed for NTSC films)
Conversely, you can figure out how to speed up NTSC standard frame rate audio to the PAL system (1.0427094).
This trick works, for example, should you want to add a better quality audio track obtained from a different source.
Related
I have an old black and white film with audio that was recorded on a phonograph and the playback speed of the audio is a semi-tone higher than it should be. How can I process this video with ffmpeg and slow down BOTH the video and audio together, bringing the audio down a half step (semi-tone) while keeping the video in sync?
I have tried filters like asetrate, setpts, atempo, etc and although I could change the pitch of the audio, the video became badly out of sync with the audio.
ffmpeg -i input.mp4 -filter:v "setpts=0.5*PTS" -an output.mp4
0.5 is the ratio
When extracting segments from a media file with video and audio streams without re-encoding (-c copy), it is likely that the requested seek & end time specified will not land precisely on a keyframe in the source.
In this case, ffmpeg will grab the nearest keyframe of each track and position them with differing starting PTS values so that they remain in sync.
Video keyframes tend to be a lot more spaced apart, so you can often end up with something like this:
Viewing the clip in VLC, the audio will start at 5 seconds in.
However, in other video players or video editors I've noticed this can lead to some playback issues or a/v desync.
A solution would be to re-encode both streams when extracting the clip, allowing ffmpeg to precisely seek to the specified seek time and generating equal length & synced audio and video tracks.
However, in my case I do not want to re-encode the video, it is costly and produces lower quality video and/or greater file sizes. I would prefer to only re-encode the audio, filling the initial gap with generated silence.
This should be simple, but everything I've tried has failed to generate silence before the audio stream begins.
I've tried apad, aresample=sync=1, and using amerge to combine the audio with anullsrc. None of it works.
All I can think to possibly get around this is to use ffprobe on the misaligned source to retrieve the first audio PTS, and in a second ffmpeg process apply this value as a negative -itoffset, then concatting the audio track with generated silence lasting the duration of silence... But surely there's a better way, with just one instance of ffmpeg?
Any ideas?
I just stumbled across the solution by trying some more things.
I take the misaligned source media and process it with another ffmpeg instance with some audio filters:
ffmpeg -fflags +genpts -i input.mkv -c copy -c:a aac -af apad,aresample=async=1:first_pts=0 -ac 2 -shortest -y output.mkv
And it does exactly what I want, pads the beginning (and end) of the audio stream with silence making the audio stream equal length to the video.
The only drawback is that I can't combine this with my original ffmpeg command that extracts the clip, the only way this works is as a 2-step process.
I am using ffmpeg to rotate videos 90 or 180 degrees in a Python script. It works great. But, I am curious as to why the output file would be a smaller amount of bytes than the input file.
Here are the commands I use:
180 degrees:
ffmpeg -i ./input.mp4 -preset veryslow -vf "transpose=2,transpose=2,format=yuv420p" -metadata:s:v rotate=0 -codec:v libx264 -codec:a copy ./output.mp4
90 degrees:
ffmpeg -i ./input.mp4 -vf "transpose=2" ./output.mp4
For example, a GoPro Hero 3 MP4 file was originally 2.0 GB. The resulting output file was 480.9 MB. Another GoPro file was 2.0 and its resulting file was 671.5 MB. Is this maybe because the GoPro files were 2.0 but contains empty space, sort of like how some NTFS filesystems make a minimal 4k file, even when there is less bytes in it?
If this isn't the GoPro Hero 3, how do I rotate the files 90 or 180 degrees but ensure the output file size is the same? Or, is data loss expected? Does the data loss have to do with the format?
Note that the quality of the video doesn't appear to be damaged, which is good. So, I am interested in learning more about why this is happening, then I can read the section of ffmpeg documentation that is relevant to this.
Thank you!
Bitrate is ignored from the start
ffmpeg fully decodes the input into uncompressed raw video and audio (except when stream copying – more about that below). The input format or bitrate does not matter: it does this for all formats. The encoder then works from these raw, decoded frames. See diagram.
H.264 vs H.264
Your input and output are both H.264. A format, such as H.264, is created by an encoder. Anyone can make an encoder. However, not all encoders are equal. Given the same input, the output from one H.264 encoder may have the same quality as an output from another H.264 encoder, but the bitrate may be several times smaller.
The GoPro H.264 encoder was made to work on a platform with limited hardware. That means bitrate (file size) is sacrificed for speed and quality. x264 is the ultimate H.264 encoder: nothing can beat its quality-to-bitrate ratio.
Rotate without re-encoding
You can stream copy (re-mux) and rotate at the same time. The rotation is handled by the metadata/sidedata:
ffmpeg -i input.mp4 -metadata:s:v rotate=90 -c copy output.mp4
Downside is your player/device may ignore the rotation, so you may have to physically rotate with filters which requires re-encoding, and therefore stream copy can't be used.
I had the same rotation issue once...
I fixed it by "resetting" the rotation instead...
ffmpeg ...... -metadata:s:v rotate="0" ......
I'm creating a fragmented MP4 for the use of playing in Media Source Extensions.
The command line is: ffmpeg.exe -probesize 10000000 -r 10 -i - -vcodec copy -an -f mp4 -reset_timestamps 0 -blocksize 30000 -movflags empty_moov+default_base_moof+frag_keyframe -loglevel debug -
The source of the video is an IP camera streaming H.264.
The configured and expected frame rate is 10FPS but there is no guarantee for 10FPS, for example a frame may get dropped occasionally, or the camera may just not play nice with what it declares.
I have simulate a 10% p-frames drops to emphasize the following issue:
With the above command, the output video plays faster than real-time and this is a problem because the whole pipe is a live stream.
With the 10% frame drop simulation, the effective playback rate it 1.1x.
I don't want to obligate to a fixed frame rate because there is no guarantee for a fixed-rate.
If I remove the -r 10 flag entirely, the MP4 seems to the playing at 2x-3x speed.
Is there a way building the MP4 timestamps in a more dynamic way? for example, giving it the RTP timestamp or somehow telling ffmpeg to build the MP4 with the timestamp of the "feed" time?
I generate a bunch of videos by looping .PNG inputs along with .AMR audio.
ffmpeg -loop 1 -i slide001.png -i slide001.amr -vcodec rawvideo -acodec pcm_s16le -t 5 slide001.avi
Eventually I join the .AVIs using Mencoder. I haven't had much success trying to join them with FFmpeg.
It seems that because the .AMR's actual duration is shorter than the 5 seconds I specified for example, when the time comes to join the .AVIs, the audio stream will be out of sync (ahead of the video).
I suspect I need to find a way of padding the audio with silence or specifying its duration somehow so that the audio and video streams in my .AVIs are of equal length before joining them.
Any ideas?
Thanks.
I ended up solving this by using the SoX audio processor in addition to ffmpeg: http://sox.sourceforge.net/Main/HomePage
The command I use is
sox slide001.wav slide001padded.wav pad 0 3
The above adds 3s of silence to the end of the audio. When ffmpeg's "-t 5" duration command is applied in the video creation, it trims the extra silence so that the video and audio streams are the same length.
This works only when the audio is shorter than the video. You need to add enough silence so that the audio duration equals or exceeds the video duration.