I have some audio (wave file) that is sampled at a rate of 48000 samples per second.
This audio was created to match a 30 FPS video. However, the video actually plays back on the target at the NTSC framerate of 29.97 (30 X 1000/1001).
This means that I need to time-dilate the audio so that there are 48048 samples where there were previously 48000 samples (it plays back 1.001 times slower) but still maintains that the final audio file's rate is 48000 samples per second.
Ideally, also, I'd like to do this resample using the sox library option for FFMPEG since I hear it has much higher quality.
Can anyone help me with the command line necessary to process a file in this manner?
Basic command is
ffmpeg -i in.wav -af asetrate=47952,aresample=48000:resampler=soxr out.wav
This assumes that libsoxr is linked.
Related
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 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.
Our security system records and archives our IP cameras streams with ffmpeg -use_wallclock_as_timestamps 1 -i rtsp://192.168.x.x:554/mpeg4 -c copy -t 60 my_input_video.avi
I run it with crontab every minute so it creates videos of 60 seconds (~15Mb) for each camera every minute. When an intrusion occurs, the camera sends a picture through FTP and a script called by incrontab:
1- forwards immediately the picture by email
2- selects the video covering the minute the intrusion occured, compress it with h264 (to ~2,6Mb) and sends it by email
It is working really well but if a thief crosses the path of various cameras, the connection to the SMTP server is not fast enough so video emails are delayed. I'd like to compress the videos even more to avoid that. I could lower the resolution (640x480 to 320x240 for example) but sometimes 640x480 is handy to zoom on something which looks to be moving...
So my idea is to drop frames in the video in order to lower the filesize. I don't care if the thief is walking like a "stop motion Lego" on the video, the most important is I know there is someone so I can act.
mediainfo my_input_video.avi says Frame rate = 600.000 fps but it is of course wrong. FPS sent by IP cameras are always false because it varies with the network quality; this is why i use "-use_wallclock_as_timestamps 1" in my command to record the streams.
with ffmpeg -i my_input_video.avi -vcodec h264 -preset ultrafast -crf 28 -acodec mp3 -q:a 5 -r 8 output.avi the video is OK but filesize is higher (3Mb)
with ffmpeg -i my_input_video.avi -vcodec h264 -preset ultrafast -crf 28 -acodec mp3 -q:a 5 -r 2 output.avi the filesize is lower (2,2Mb) but the video doesn't work (it is blocked at the first frame).
Creating a mjpeg video (mjpeg = not interlaced frames) in the middle of the process (first exporting to mjpeg with less frames and then exporting to h264) creates same results.
Do you know how I can get my thief to walk like a "stop motion Lego" to lower the filesize to a minimum?
Thanks for any help
What are your constraints file size wise? 2.6MB for 60 seconds of video seems pretty reasonable to me, thats about 350kbps, which is pretty low for video quality.
You need to specify the video bitrate -b:v 125000 (125kbps, should drop you to about 900kb) to control the bitrate/s you want the video encoded at. Your not giving FFMpeg enough hints as to how you want the video handled, so its picking arbitrary values you don't like. As you drop the frame rate, its just using up the buffers allocating more bits to each frame. (one big thing you need to keep in mind with this is, as you stretch the video out over a longer time period the more likely the scene will change significantly require an I frame (full encoded frame vs frame based on previous frame) so reducing the frame rate will help some, but may not help as much as you'd think).
Your "(it is blocked at the first frame)." is most likely an issue with you trying to start decoding a stream when it is not at an I frame and not an issue with your settings.
Say I have something like this
ffmpeg -i video.avi -ar 22050 -ab 32 -f flv -s 320x240 video.flv
-ar (Audio sampling rate in Hz)
-ab (Audio bit rate in kbit/s)
regarding the -ar and the -ab how do I know what rate to use? I got this ffmpeg command from a site somewhere and I was wondering how the person knew what values to put for the rates? Do I need to understand audio in order to figure that out?
Probably 44100 for audio sampling rate and 128 for bit rate should be sufficient.
Check Wikipedia's sampling rate and audio bit rate articles for examples to see if those values are too high or too low for what you're trying to do.
You have to use "ffmpeg -i video.avi" to know the sampling rate and the bitrate of the audio stream in the source video.avi.
The audio stream can be extracted with the same sampling rate and bitrate without lose quality.
You can decide to reduce one of them for size reasons, but don't increment one of them to increase quality because you never can't upgrade the original quality.
I'm using -ar 22050 and -ab 48 for Avi and Mpeg video files. It works normally.