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• ffmpeg issue after adding background audio

  9 April 2021, by spicy

  So I have an issue to which I cannot find an answer. I have a video which is 7min 38sec long after concat. When I want to add background audio to this video it cuts off the last 7sec of the original video. I cannot find the reason.

  


  If anyone could help me in any way, it would be amazing because I have been troubleshooting this past 3 days.

  


  My FFmpeg version is ffmpeg version 4.2.3 Copyright (c) 2000-2020 the FFmpeg developers.

  


  For concat I use this: FFmpeg -f concat -safe 0 -i txtfile.txt -c copy -ac 2 output.mp4

  


  For adding background audio I use this: ffmpeg -i output.mp4 -i background.mp3 -c:v copy -filter_complex "[0:a]aformat=fltp:44100:stereo,apad[0a];[1]aformat=fltp:44100:stereo,volume=0.2[1a];[0a][1a]amerge[a]" -map 0:v -map "[a]" -ac 2 -shortest output_with_background.mp4

  


  And the console output while doing its stuff is:

  


  ffmpeg version 4.2.3 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  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
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baee72440] Auto-inserting h264_mp4toannexb bitstream filter
Input #0, concat, from 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\videos.txt':
  Duration: N/A, start: -0.023220, bitrate: 18916 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 18861 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 54 kb/s
    Metadata:
      handler_name    : SoundHandler
File 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\output.mp4' already exists. Overwrite ? [y/N] y
Output #0, mp4, to 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\output.mp4':
  Metadata:
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 18861 kb/s, 30 fps, 30 tbr, 15360 tbn, 15360 tbc
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 54 kb/s
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baee72440] Auto-inserting h264_mp4toannexb bitstream filter
    Last message repeated 5 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baeeec180] Auto-inserting h264_mp4toannexb bitstream filter
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baee72440] Auto-inserting h264_mp4toannexb bitstream filter
    Last message repeated 14 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baee72440] Auto-inserting h264_mp4toannexb bitstream filter
    Last message repeated 20 times
[mov,mp4,m4a,3gp,3g2,mj2 @ 0000024baee72440] Auto-inserting h264_mp4toannexb bitstream filter
    Last message repeated 7 times
frame=   52 fps= 42 q=-1.0 Lsize=    7838kB time=00:07:38.82 bitrate= 139.9kbits/s speed= 374x
video:4802kB audio:2956kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 1.044608%
ffmpeg version 4.2.3 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  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
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\output.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.29.100
  Duration: 00:07:38.85, start: 0.000000, bitrate: 139 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 87 kb/s, 0.12 fps, 30 tbr, 15360 tbn, 60 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 52 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Input #1, mp3, from 'C:\Users\shooey\Desktop\junky\Content\assets\background.mp3':
  Metadata:
    major_brand     : dash
    Software        : Lavf57.56.101
    minor_version   : 0
    compatible_brands: iso6mp41
  Duration: 00:20:03.02, start: 0.025057, bitrate: 167 kb/s
    Stream #1:0: Audio: mp3, 44100 Hz, stereo, fltp, 167 kb/s
    Metadata:
      encoder         : LAME3.100
File 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\output_bckg.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
  Stream #0:1 (aac) -> aformat
  Stream #1:0 (mp3float) -> aformat
  Stream #0:0 -> #0:0 (copy)
  amerge -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[Parsed_amerge_4 @ 0000017093965c80] No channel layout for input 1
[Parsed_amerge_4 @ 0000017093965c80] Input channel layouts overlap: output layout will be determined by the number of distinct input channels
Output #0, mp4, to 'C:\Users\shooey\Desktop\junky\Content\07-04-2021\output_bckg.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 87 kb/s, 0.12 fps, 30 tbr, 15360 tbn, 15360 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      encoder         : Lavc58.54.100 aac
frame=   52 fps=5.5 q=-1.0 Lsize=   11993kB time=00:07:31.67 bitrate= 217.5kbits/s speed=  48x
video:4802kB audio:7111kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.670061%
[aac @ 00000170935b4980] Qavg: 319.394


  


• ffmpeg -ss then apply filter then concat producing timestamp errors

  6 August 2020, by Bob Ramsey

  Using ffmpeg, I have split a file into multiple parts using -ss. Then I apply a filter to some of the files, then concat the files back together. When I do that, I get: Non-monotonous DTS in output stream 0:0; previous: 341334, current: 340526; changing to 341335. This may result in incorrect timestamps in the output file. The output file plays, but there are noticeable skips where the files are joined.

  


  Here's how I am splitting the file:

  


  ffmpeg -i full_source.mp4 -ss 0 -to 14.264250 -c copy  01-plain.mp4
ffmpeg -i full_source.mp4 -ss 14.264250 -to 18.435083 -c copy  01-filtered.mp4

ffmpeg -i full_source.mp4 -ss 18.435083 -to 29.988292 -c copy  02-plain.mp4
ffmpeg -i full_source.mp4 -ss 29.988292 -to 31.865167 -c copy  02-filtered.mp4
...
ffmpeg -i full_source.mp4 -ss 0 -to 14.264250 -c copy  10-plain.mp4
ffmpeg -i full_source.mp4 -ss 234.484203 -to 300.000 -c copy  10-filtered.mp4


  


  Then I apply a different drawtext filter on each of the 10 filtered files and save them with a new name, like:

  


  ffmpeg -hide_banner -loglevel warning -y -i 01-filtered.mp4 -filter_complex "drawtext=fontfile=calibri.ttf:fontsize=24:fontcolor=white:x=300:y=500:text='hello world'" -crf 15 01-filtered-complete.mp4


  


  Finally, I join all of the plain and complete files back together like this:

  


  ffmpeg  -f concat -safe 0 -i mylist.txt -c:a copy -c:v copy  outfile.mp4


  


  And that's where the timing error comes in. I've tried adding -vsync drop in the concat command, but that didn't really work either. Same version of ffmpeg does the split, the filter, and the concat. I've tried different versions, everything from 20170519 to one from May 2020 with the same result. Always making sure that all three steps are done by the same version of ffmpeg.

  


  The only thing I can see is that ffprobe shows a duration of 14.27 for 01-plain.mp4 when it should be 14.264250. All of the other files show a similar rounding difference. The files are 23.98 fps. If I do all of my filters in really long command without splitting the file, I can use the more precise numbers with no problem. It just takes 10 times as long. This is all scripted, it happens a couple of hundred times a day and time is money, so I can't take 10 times as long to do each file.

  


  Any ideas? Thanks in advance!

  


• How to initialize the AV format context from NetworkStream rather than url?

  23 July 2020, by Ahmad

  I'm using FFmpeg.AutoGen to encode catpured frames:
https://github.com/Ruslan-B/FFmpeg.AutoGen

  


  I send the encoded packets through NetworkStream to another program using TCP Client/Server application, using this code:

  


   //Send size and encoded packet
 _stream.Write(BitConverter.GetBytes(pPacket->size), 0, 4);
 using (var packetStream = new UnmanagedMemoryStream(pPacket->data, pPacket->size)) 
        packetStream.CopyTo(_stream);


  


  At the server-side, after UTF8 encoding, I receive content like this:

  


  


  gd ?r@?/?a 2&#xF;1?   h?C?,?  ????E???H??,? ?#??x264 - core 160 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=11 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 ?  e??   ??S?T????M?.O ??Ea???Q?OI??X?????[?P???42=h??M?K? ??{?@c;?t???Ke?>?????2??&#xB;?\qXEvg???U A?X??q/&#xB;(?? ?L?_c?g?X?c(To?k?Y?-F???R?]??U?&#xA;;|=r????\U???2?g??>??½?7?A&#xE;?1W9???A??&#xF;?_L??P?)?&#xF;M????q/?L??):??<??8d/*?&#xF;?^??}?B?H?_8C[y???CY?QM(J???1<3%?K?F;Q3Q????&#xA;??jy&#xF;J"???y]k?m?^}???S????h?&#xF;3????EP???pw???q?jKp???h?????

  &#xA;

  &#xA;

  My problem is, how to initialize the AV format context with this data, and decode the received packet to recover the original frame?

  &#xA;

  _pFormatContext = ffmpeg.avformat_alloc_context();
ffmpeg.avformat_open_input(&pFormatContext, url, null, null).ThrowExceptionIfError();


  &#xA;

  As problem exactly is, ffmpeg.avformat_open_input takes URL as an input, not byte[]. If _pFormatContext can be initialized using the received data, things are resolved, but HOW?

  &#xA;