How to get ffmpeg to fail if stdin stream has errors? - ffmpeg

I've got some code that outputs rawvideo to ffmpeg via pipe, which then overlays a png over it. If the code exits with an error, ffmpeg keeps going forever, I don't know how to stop it :( Any ideas? Here's an example:
set -o pipefail
(echo "error!!"; exit 1) | \
ffmpeg -f rawvideo -c:v rawvideo -s 1280x720 -pix_fmt bgra -i - \
-i my_icon.png \
-filter_complex "[0][1]overlay=50:50" \
out.mp4
Note ffmpeg DOES show an error message, but still does not stop, stuck in 'frame 0' forever, with a strange time attribute e.g. time:-577014:32:22.77:
[rawvideo # 0x563fc94427c0] Packet too small (8)
Error while decoding stream #0:0: Invalid data found when processing input
Finishing stream 0:0 without any data written to it.:32:22.77 bitrate= -0.0kbits/s speed=N/A
frame= 0 fps=0.0 q=0.0 size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A
ffmpeg version information as requested in comments:
ffmpeg version 3.4.2-2 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 7 (Ubuntu 7.3.0-16ubuntu2)
configuration: --prefix=/usr --extra-version=2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100

Related

Error while concatenating audio files with ffmpeg

While concatenating audio files with
ffmpeg -safe 0 -f concat
-segment_time_metadata 1
-i {m3u_file_name}
-vf select=concatdec_select
-af aselect=concatdec_select,aresample=async=1 {ogg_file_name}
I get the following output
ffmpeg version 3.4.8-0ubuntu0.2 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, concat, from 'fm3u.m3u':
Duration: N/A, start: -0.006500, bitrate: N/A
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (opus (native) -> vorbis (libvorbis))
Press [q] to stop, [?] for help
Output #0, ogg, to 'fout.ogg':
Metadata:
encoder : Lavf57.83.100
Stream #0:0: Audio: vorbis (libvorbis), 48000 Hz, mono, fltp
Metadata:
encoder : Lavc57.107.100 libvorbis
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
...
...
Error while decoding stream #0:0: Invalid data found when processing input
[opus # 0x558b2c245400] Error parsing the packet header.
Error while decoding stream #0:0: Invalid data found when processing input
size= 47kB time=00:00:10.84 bitrate= 35.2kbits/s speed=4.38x
video:0kB audio:42kB subtitle:0kB other streams:0kB global headers:3kB muxing overhead: 10.251080%
/fout.ogg
The final audio sounds good.
ffprobe on the 2 participating files looks like
# ffprobe <FILE>
ffprobe version 3.4.8-0ubuntu0.2 Copyright (c) 2007-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, ogg, from 'f1.ogg':
Duration: 00:00:01.08, start: 0.000000, bitrate: 34 kb/s
Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
# ffprobe <FILE>
ffprobe version 3.4.8-0ubuntu0.2 Copyright (c) 2007-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.2 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, ogg, from 'f2.ogg':
Duration: 00:00:05.00, start: 0.000000, bitrate: 6 kb/s
Stream #0:0: Audio: vorbis, 48000 Hz, mono, fltp, 80 kb/s
Metadata:
ENCODER : Lavc57.107.100 libvorbis
What is the reason of the error?
As explained in this answer
"file-level concatenation (-f concat) requires all files in the listing to have the exact same codec parameters. "
In this case, the input files have codecs opus and vorbis.
To resolve the issue, I recreated some files (silence files) so they all have the same CODEC as the files I am mixing them with.
In particular, instead of creating them using:
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a libvorbis silence-5.8.ogg
I use now (note libopus)
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a libopus silence-5.8.ogg
And the "concat with ffmpeg" works without any problem.
I also tried
ffmpeg -f lavfi -i anullsrc=r=48000:cl=mono -t 5.8 -c:a opus silence-5.8.ogg
But that returns with
[opus # 0x5607f1765900] The encoder 'opus' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
[opus # 0x5607f1765900] Alternatively use the non experimental encoder 'libopus'.
So, libopus works just fine (no need for opus)

How to convert .srt file into ttml based xml subtitle file using ffmpeg?

I have an .srt subtitle file. I want to convert it into .ttml format(or xml) using ffmpeg command. I tried using traditional way to convert this. But ffmpeg throwing an error. Is there any way to convert this in Linux platform? Or any other command line applications do this conversion?
Below is the command I tried.
ffmpeg -i my_srt.srt -y srt-to-xml.xml -v verbose
ffmpeg version 4.2.4-1ubuntu0.1 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 31.100 / 56. 31.100
libavcodec 58. 54.100 / 58. 54.100
libavformat 58. 29.100 / 58. 29.100
libavdevice 58. 8.100 / 58. 8.100
libavfilter 7. 57.100 / 7. 57.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Input #0, srt, from 'my_srt.srt':
Duration: N/A, bitrate: N/A
Stream #0:0: Subtitle: subrip
Output #0, webm_dash_manifest, to 'srt-to-xml.xml':
Output file #0 does not contain any stream
[AVIOContext # 0x5614924ef500] Statistics: 22267 bytes read, 0 seeks
I even tried like below.
ffmpeg -i my_srt.srt -y srt_to_ttml.ttml
It didn't work too.
It looks like ffmpeg 4.4 has minimal support for converting srt into ttml: font styles and positions are not preserved.
ttconv is an alternative.
The FFmpeg TTML muxer uses the .ttml extension:
ffmpeg -i input.srt output.ttml
The TTML encoder and muxer do not have any additional options as of writing this answer (see ffmpeg -h encoder=ttml & ffmpeg -h muxer=ttml).

No Output when transcoding RTP to HLS with ffmpeg

I am running ubuntu 18.04(bionic) and i have generated a live RTP stream from kurento-media-server and i am converting it to HLS with this command of ffmpeg:
ffmpeg -protocol_whitelist file,udp,rtp -i rtp://127.0.0.1:55000 -vcodec libx264 -acodec libfdk_aac -f hls /live-stream/kurento-rtmp/hls/playlist.m3u8
However. it shows this output and doesn't do anything and stays there. Any ideas why this is happening are really appreciated.
ffmpeg version 4.3.1-0york0~18.04 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version='0york0~18.04' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libzimg --enable-pocketsphinx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil 56. 51.100 / 56. 51.100
libavcodec 58. 91.100 / 58. 91.100
libavformat 58. 45.100 / 58. 45.100
libavdevice 58. 10.100 / 58. 10.100
libavfilter 7. 85.100 / 7. 85.100
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 7.100 / 5. 7.100
libswresample 3. 7.100 / 3. 7.100
libpostproc 55. 7.100 / 55. 7.100
I believe that at this stage ffmpeg is trying to determine the duration of input stream but since it is live it will never finish. If so, how would i flag ffmpeg that it is a live stream and not a local video.
I solved the issue by using -re flag which slows down ffmpeg to not read the whole stream at once and get frame by frame however, i should have passed sdp file instead of rtp uri because it gives this error:
[rtp # 0x55bee6277180] Unable to receive RTP payload type 96 without an SDP file describing it

Requested output 'm4a' is not a suitable output format

I am attempting to combine youtube-dl and ffmpeg to clip audio from YouTube videos. Everything was fine on my laptop, but when I moved to my server (Debian 9) I started coming across this error. Why is this? Am I missing a dependency ... if so which one?
Input
ffmpeg -f m4a $(youtube-dl -f bestaudio[ext=m4a] -g \'https://www.youtube.com/watch?v=qnjYyfkcaNI\' | sed "s/.*/-ss 0 -i &/") -to 60 -c copy -y /public/audio/test.m4a
Error
[NULL # 0x55cf161f1e60] Requested output format 'm4a' is not a
suitable output format /public/audio/test.m4a: Invalid argument
ffmpeg info
ffmpeg version 3.2.12-1~deb9u1 Copyright (c) 2000-2018 the FFmpeg developers
built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
configuration: --prefix=/usr --extra-version='1~deb9u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
The error was not with ffmpeg. The youtube-dl command was failing and resulting in the error.
Updated command
ffmpeg -f m4a $(youtube-dl -f bestaudio[ext=m4a] -g https://www.youtube.com/watch?v=qnjYyfkcaNI | sed "s/.*/-ss 0 -i &/") -to 60 -c copy -y /public/audio/test.m4a
Basically just had to remove the \'.
I also had to go one step further and give permission to the www-data apache group for the output folder since the command was being executed from a PHP script.

Adding audio to mp4 using ffmpeg at certain offset gives output file #0 does not contain any stream

I am attempting to add audio to a MP4 file using ffmpeg at a specific time, eventually I also want the audio to only play until a certain point, so if you can help with that also that would be great, otherwise if you can just assist me in figuring out what is causing this error.
The arguments I use
-y "E:\Documents\Content Bar/Photo Booth\PhotosDrop\2018-05-15 14-01-16OUT.mp4" -itsoffset -i "E:\Downloads\wetransfer-250b26\05 Call of Love.mp3" -map 0:0 -map 1:0 -c:v copy -shortest "E:\Documents\Content Bar\Photo Booth\PhotosDrop\2018-05-15 14-01-16OUTOUT.mp4"
This results in the following error:
ffmpeg version N-81646-gc19da0c Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-libebur128 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
libavutil 55. 29.100 / 55. 29.100
libavcodec 57. 55.101 / 57. 55.101
libavformat 57. 48.103 / 57. 48.103
libavdevice 57. 0.102 / 57. 0.102
libavfilter 6. 62.100 / 6. 62.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Output #0, mp4, to 'E:\Documents\Content Bar/Photo Booth\PhotosDrop\2018-05-15 14-01-16OUT.mp4':
Output file #0 does not contain any stream
Okay so I figured it out using the following command
-i "E:\Documents\Content Bar/Photo Booth\PhotosDrop\2018-05-15 14-01-16OUT.mp4" -i "E:\Downloads\wetransfer-250b26\05 Call of Love.mp3" -filter_complex "[1]atrim=end=20,asetpts=PTS-STARTPTS[aud1];[aud1]adelay=6000|6000[aud];[0][aud]amix" -c:v copy out.mp4

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