I have a mkv file with audio, video and subtitles. It also has a lot (~50) fonts (idk why).
The size and all is fine, but I want to make the one subtitle stream, default.
The problem is, when I use "-c copy", it loses all fonts. I know that I can keep everything with "-map 0", but this encodes all streams and goes from 70k fps (-c copy) down to only 20 fps -> takes a very long time.
This is the command I used:
$ ffmpeg -i 1.mkv -c copy -disposition:s:0 default 2.mkv
From the documentation regarding automatic stream selection:
Data or attachment streams are not automatically selected and can only be included using -map.
Example to include all streams:
ffmpeg -i 1.mkv -map 0 -c copy -disposition:s:0 default 2.mkv
See FFmpeg Wiki: Map.
Related
I've been looking all over the web & StackOverflow, and can't get this to work. I have an audio file that I'd like to split into mp3 files and generate a corresponding m3u8 file.
I've tried this, which was the closest:
ffmpeg -i sometrack.wav -c:a libmp3lame -b:a 256k -map 0:0 -f segment -segment_time 10 -segment_list outputlist.m3u8 -segment_format mpegts 'output%03d.mp3'
But all the mp3 files are garbled when I play them.
There are two issues here. FFmpeg normally looks at the extension of the output files to determine output container. However, when the output format is forced -segment_format for segment muxer or just -f format for most others, ffmpeg will pay heed to that and no longer look at the extension. In this case, segment_format is set to mpegts so that's what the output files will be. To ensure valid mp3 files, set segment_format to mp3.
The second issue is that since the extension is mp3, my guess is that hls.js is not able to correctly determine the format of the segments, or it assumes a wrong format and tries to parse them that way. Either way, there should be some messages in the browser console to that effect. See https://github.com/video-dev/hls.js/pull/1190 for issues that hls.js has had with format probing.
I need to convert an SEC file into any video format that I can share and/or upload to Youtube. MP4, etc.
I'm a complete newbie at all things terminal. I've tried:
ffmpeg -i video.sec video.mp4
ffmpeg -i video.sec -bsf:v h264_mp4toannexb -c:v copy video.avi
ffmpeg -i video.sec -b 256k -vcodec h264 -acodec aac video.mp4
I don't understand what any of these mean, they're just examples I found online. However, whatever I try returns this error:
Invalid data found when processing input
Any thoughts? Thanks!
I had to add the following option so it would skip the SEC's custom header.
-skip_initial_bytes 48
i know this is old, but i was trying to figure this out as well, what ended up finally working for me was this command.
./ffmpeg -f h264 -i INPUT.sec -filter:v "setpts=4*PTS" OUTPUT.avi
the -f h264 was the part i was missing. and the -filter:v "setpts=4*PTS" part is to slow it back down to the original speed. you can also change the .avi at the end to whichever format works best for you.
i hope this helps someone out :)
OK, just to clear up some recent threads…
The Samsung DVR used here was an SRD-440. RB kindly sent me a file to test and he sent me a .BU file with an associated .db2 file. This was a bit of a surprise as in all older Samsung DVR’s, the .bu files can only be played back in the DVR. I mentioned this here, https://spreadys.wordpress.com/2014/07/21/ifsec-samsung-exports/
It appears that Samsung have caught on, and the BU file is now playable due to it being a H264/AVC Stream conforming to a standard profile. I have updated the IFSEC Post mentioned above to highlight this change.
Back to RB’s stream and the challenge was to get these files viewable in WMV format. They were all field based, at 704×288.
The speed of playback is controlled by the Samsung software, using the .db2 file. As such, the metadata and timing information in the video stream was wrong. This caused speed issues and then quality issues when attempting to correct this.
As a result, I found it necessary to force an input rate and generate a new Presentation Time Stamp BEFORE the input file.
The following FFmpeg string did the job…
ffmpeg -r 12 -fflags genpts -i FILE.bu -vf scale=704:528 -sws_flags lanczos -q:v 2 FILE.wmv
Remember, this is for preview – analysis would be completed differently due to the scaling, the interpolation method, and the WMV compression!
As its likely that RB may have quite a few .bu files in a folder, I placed this into a batch file to transcode the whole lot within a few minutes… more on batch files coming in a new post soon!
https://spreadys.wordpress.com/2014/07/21/ifsec-samsung-exports/
or
ffmpeg -i (name of file).sec (name of final file).mp4
ffmpeg -i (name of file).sec -filter:v "setpts=3.3*PTS" (name of final_file).mp4
I have an application used in government and subject to regulation that prevents transcoding or altering the video quality in any way.
I’m attempting to utilize FFmpeg to change a video into an MP4 by copying the raw streams to a new container.
This is the command being used:
ffmpeg.exe -y -i INPUT.ASF -c:av copy OUTPUT.MP4
Notice the -c:av copy. The FFmpeg documentation says, “a special value copy (output only) to indicate that the stream is not to be re-encoded.“
Visually the videos before and after appear to be identical quality with no pixelation on the ships.
Is this altering the video quality or could this be considered transcoding?
There's a syntax error but other than that, yes, copy will avoid transcoding of the stream. The hitch is that the output container may not support all codecs that the input container does.
ffmpeg.exe -y -i INPUT.ASF -c copy OUTPUT.MP4
Your current command was transcoding the video since ffmpeg's parser only consumes the first character in stream type i.e. -c:av is treated as -c:a. -c copy will copy all stream types. Use -c:v copy -c:a copy to separately set codec mode for video and audio.
ffmpeg handles RTMP streaming as input or output, and it's working well.
I want to stream some videos (a dynamic playlist managed by a python script) to a RTMP server, and i'm currently doing something quite simple: streaming my videos one by one with FFMPEG to the RTMP server, however this causes a connection break every time a video end, and the stream is ready to go when the next video begins.
I would like to stream those videos without any connection breaks continuously, then the stream could be correctly viewed.
I use this command to stream my videos one by one to the server
ffmpeg -re -y -i myvideo.mp4 -vcodec libx264 -b:v 600k -r 25 -s 640x360 \
-filter:v yadif -ab 64k -ac 1 -ar 44100 -f flv \
"rtmp://mystreamingserver/app/streamName"
I looked for some workarounds over the internet for many days, and i found some people talking about using a named pipe as input in ffmpeg, I've tried it and it didn't work well since ffmpeg does not only close the RTMP stream when a new video comes but also closes itself.
Is there any way to do this ? (stream a dynamic playlist of videos with ffmpeg to RTMP server without connection breaks
Update (as I can't delete the accepted answer): the proper solution is to implement a custom demuxer, similar to the concat one. There's currently no other clean way. You have to get your hands dirty and code!
Below is an ugly hack. This is a very bad way to do it, just don't!
The solution uses the concat demuxer and assumes all your source media files use the same codec. The example is based on MPEG-TS but the same can be done for RTMP.
Make a playlist file holding a huge list of entry points for you dynamic playlist with the following format:
file 'item_1.ts'
file 'item_2.ts'
file 'item_3.ts'
[...]
file 'item_[ENOUGH_FOR_A_LIFETIME].ts'
These files are just placeholders.
Make a script that keeps track of you current playlist index and creates symbolic links on-the-fly for current_index + 1
ln -s /path/to/what/to/play/next.ts item_1.ts
ln -s /path/to/what/to/play/next.ts item_2.ts
ln -s /path/to/what/to/play/next.ts item_3.ts
[...]
Start playing
ffmpeg -f concat -i playlist.txt -c copy output -f mpegts udp://<ip>:<port>
Get chased and called names by an angry system administrator
Need to create two playlist files and at the end of each file specify a link to another file.
list_1.txt
ffconcat version 1.0
file 'item_1.mp4'
file 'list_2.txt'
list_2.txt
ffconcat version 1.0
file 'item_2.mp4'
file 'list_1.txt'
Now all you need is to dynamically change the contents of the next playlist file.
You can pipe your loop to a buffer, and from this buffer you pipe to your streaming instance.
In shell it would look like:
#!/bin/bash
for i in *.mp4; do
ffmpeg -hide_banner -nostats -i "$i" -c:v mpeg2video \
[proper settings] -f mpegts -
done | mbuffer -q -c -m 20000k | ffmpeg -hide_banner \
-nostats -re -fflags +igndts \
-thread_queue_size 512 -i pipe:0 -fflags +genpts \
[proper codec setting] -f flv rtmp://127.0.0.1/live/stream
Of course you can use any kind of loop, also looping through a playlist.
I figure out that mpeg is a bit more stabile, then x264 for the input stream.
I don't know why, but minimum 2 threads for the mpeg compression works better.
the input compression need to be faster then the output frame rate, so we get fast enough new input.
Because of the non-continuing timestamp we have to skip them and generate a new one in the output.
The buffer size needs to be big enough for the loop to have enough time to get the new clip.
Here is a Rust based solution, which uses this technique: ffplayout
This uses a JSON playlist format. The Playlist is dynamic, in that way that you can edit always the current playlist and change tracks or add new ones.
Very Late Answer, but I recently ran into the exact same issue as the poster above.
I solved this problem by using OBS and the OBS websockets plugin.
First, set your RTMP streaming app as you have it now. but stream to a LOCAL RTMP stream.
Then have OBS load this RTMP stream as a VLC source layer with the local RTMP as the source.
then (in your app), using the OBS websockets plugin, have your VLC source switch to a static black video or PNG file when the video ends. Then switch back to the RTMP stream once the next video starts. This will prevent the RTMP stream from stopping when the video ends. OBS will go black durring the short transition, but the final OBS RTMP output will never stop.
There is surely a way to do this with manually setting up a intermediate RTMP server that pushes to a final RTMP server, but I find using OBS to be easier, with little overhead.
I hope this helps others, this solutions has been working incredible for me.
I am trying to use ffmpeg (under linux) to add a small title to a video. So, I use:
ffmpeg -i hk.avi -r 30000/1001 -metadata title="SOF" hk_titled.avi
The addition of title seems to work, but, the problem is the output file is about a 1/3rd of the file size of the input file and I was wondering why this is? Is this at the expense of quality of the video? I am unsure.. How do I preserve the same quality/size as the input file?
The main point I am unable to figure out is the use of -r option. Going through the ffmpeg docs, it seems to suggest that -r is frames per second (The input video is 23.9fps). At the moment, (30000/1001) works out to 29 fps, but I was unsure if I should be using this value.
Thanks for your time.
The default settings for ffmpeg do not always provide a good quality output when you encode, but this depends on your output format and the available encoders. With your output ffmpeg will use the default of -b 200k or -b:v 200k.
However, you can tell ffmpeg to simply copy the input streams without re-encoding and this is recommended if you just want to add or edit metadata. These examples do the same thing but use different syntax depending on your ffmpeg version:
ffmpeg -i hk.avi -vcodec copy -acodec copy -metadata title="SOF" hk_titled.avi
ffmpeg -i hk.avi -c copy -metadata title="SOF" hk_titled.avi