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• How to use ffmpeg to change the h264 properties, yuv, fps, tbr, tbn, and tbc of an mp4 file ?

  23 août 2020, par bguiz

  Using ffmpeg, I would like to convert a video file such that its video stream changes like so :

  


  

  • Current video stream : Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 2560x1080 [SAR 4:3 DAR 256:81], 60 kb/s, 15 fps, 15 tbr, 15360 tbn, 30 tbc (default)

  


  • Target video stream : Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 2560x1080, 272 kb/s, 20 fps, 20 tbr, 10240 tbn, 40 tbc (default)

  


  


  (values that differ : h264 properties, yuv, fps, tbr, tbn, and tbc)

  


  ... and its audio stream changes like so :

  


  

  • Current audio stream : Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 191 kb/s (default)

  


  • Target audio stream : Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 69 kb/s (default)

  


  


  (values that differ : aac properties)

  


  How can I do this ?

  


  ---

  


  Detailed version

  


  I would like to be able to concatenate, without re-encoding, main.mp4, followed by outro.mp4, using the following commands :

  


  echo "file 'main.mp4'" > concat.txt
echo "file 'outro.mp4'" >> concat.txt
ffmpeg \
  -f concat \
  -safe 0 \
  -i concat.txt \
  -c copy \
  concat.mp4


  


  What results is a file which plays till the end of main.mp4, and then the video freezes, and I hear the audio of outro.mp4. The same video frame then stays on, with no sound, for an extra 30 minutes (end time shown in VLC).

  


  My assumption is that these files are incompatible with each other in some way that prevents them from being concatenated using -codec copy (without re-encoding).

  


  Here is the ffprobe output for main.mp4 :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'main.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.45.100
  Duration: 01:13:00.65, start: 0.000000, bitrate: 348 kb/s
    Stream #0:0(und): Video: h264 (High 4:4:4 Predictive) (avc1 / 0x31637661), yuv444p, 2560x1080, 272 kb/s, 20 fps, 20 tbr, 10240 tbn, 40 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 69 kb/s (default)
    Metadata:
      handler_name    : SoundHandler


  


  Here is the ffprobe output for outro.mp4 :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'outro.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.45.100
  Duration: 00:00:04.12, start: 0.000000, bitrate: 254 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 2560x1080 [SAR 4:3 DAR 256:81], 60 kb/s, 15 fps, 15 tbr, 15360 tbn, 30 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 191 kb/s (default)
    Metadata:
      handler_name    : Stereo


  


  How can I convert outro.mp4, such that I may concatenate the files without re-encoding ?

  


  Note that I am OK with re-encoding the outro.mp4 on its own,
I simply want to avoid re-encoding during the concatenation step,
and avoid re-encoding main.mp4.

  


  ---

  


  ffmpeg build

  


  $ ffmpeg -version
ffmpeg version 4.3-2~18.04.york0 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version='2~18.04.york0' --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-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


  


• av_hwdevice_iterate_types returns an empty list

  9 juillet 2020, par Ruslan Ablyazov

  I used an example https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/hw_decode.c

  


  The av_hwdevice_iterate_types function returns an empty list. What could be the reason ?

  


  And avcodec_find_decoder_by_name("h264_cuvid") returns NULL.

  


  FFmpeg version :

  


  ffmpeg version 4.3 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-ladspa --enable-libpulse --enable-libsoxr --enable-libspeex --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --enable-libass --enable-libfreetype --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-nonfree --disable-ffplay --enable-libxvid --enable-cuda --enable-cuda-nvcc --enable-cuvid --enable-nvenc --enable-nonfree --enable-libnpp --extra-cflags=-I/usr/local/cuda/include --extra-ldflags=-L/usr/local/cuda/lib64
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


  


  The command ffmpeg -c:v h264_cuvid -i 7.mp4 71.mp4 outputs :

  


  ...
Stream mapping:
Stream #0:0 -> #0:0 (h264 (h264_cuvid) -> h264 (libx264))
....


  


  And it works.

  


  The command ffmpeg-hwaccel cuda-i 7.mp4 71.mp4 outputs :

  


  ...
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
....


  


  The command ffmpeg -codecs outputs :

  


   ...
 DEV.LS h264                 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 (decoders: h264 h264_v4l2m2m h264_cuvid ) (encoders: libx264 libx264rgb h264_nvenc h264_v4l2m2m nvenc nvenc_h264 )
 D.VIL. hap                  Vidvox Hap
 DEV.L. hevc                 H.265 / HEVC (High Efficiency Video Coding) (decoders: hevc hevc_v4l2m2m hevc_cuvid ) (encoders: libx265 nvenc_hevc hevc_nvenc hevc_v4l2m2m )
 ...


  


• ffmpeg stops capturing whole hour of HTTP stream after some time

  7 juillet 2020, par CompuChip

  First of all, sorry if I'm using the wrong terminology. I've been playing around with nginx and I'm still a bit confused about RTMP and HLS and other acronyms.

  


  I've managed to setup OBS to stream to an nginx server, which takes the RTMP stream and chops it into pieces for HLS. Here's the relevant part of the nginx configuration file.

  


  rtmp {
    server {
        listen 1935;
        chunk_size 4000;
        ping 30s;
        deny play all;

        application live {
            live on;
            hls on;
            hls_nested on; # Create a new folder for each stream
            hls_path /mnt/hls/live;
            hls_fragment 3s;
            hls_fragment_naming timestamp;
            hls_playlist_length 60s;
        }
    }
}

http {
    server {
        listen 81 ssl;

        #creates the http-location for our full-resolution (desktop) HLS stream - "http://localhost:8080/live/test/index.m3u8"
        location /live {
            # Elided caching and CORS for brevity

            alias /mnt/hls/live;
            add_header Cache-Control no-cache;
            index index.m3u8;
        }
    }
}


  


  This works well, I can view the stream in VLC or on a website and it looks smooth. Now I wanted to add some logging : I'd like to write full hours (starting at xx:00:00 and ending at xx:59:59) to a file named log_yyyymmdd_hh.mp4, e.g. log_20200707_18.mp4 for the files of 7 July 2020, 18:00 - 19:00 hrs. So I've set up an hourly cron job with the following ffmpeg command :

  


  ffmpeg -i https://stream.example.com:81/live/<streamkey> -preset veryfast -maxrate 2000k \&#xA;    -bufsize 2000k -g 60 -t 3600 -y /var/video/log/$(date &#x2B;\%Y\%m\%d_\%H00).mp4 >/dev/null 2>&amp;1&#xA;</streamkey>

  


  At first this seemed to work well, so I left it running happily for about 24 hours. When I checked, most of my hourly files were small ( 100MB) files of about 10 to 15 minutes long. It seems like any small delay in the stream will cause ffmpeg to stop writing to the file. I suspect such hiccups may for example be caused by an OBS plugin and I'll need to look into that, but I would prefer that ffmpeg will retry for some time before giving up. What arguments should I be passing to ffmpeg to make it not break when the stream is down for, say, up to a second every now and then ?.

  


  When I view back the HLS files there don't seem to be any noticeable gaps, so eventually all the data arrives. I went for the crontab solution with ffmpeg because when recording from nginx I could not figure out how to start recording at the start of the whole hour.