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• adding delay to an audio stream of a live feed in ffmpeg

  19 avril 2020, par blackirishman

  I am currently capturing video via a Blackmagic decklink card on macOS. My audio and video are out of sync. The audio is ahead about a second. I suspect the video is slower on account of encoding latency. My solution is to retard the audio using the ffmpeg adelay filter. I originally added a -af "adelay=1000|1000" to my command to delay the audio by 1000ms but I found that this audio filter did nothing. Consequently, I tried to build a complex_filter, but this failed. My command is producing too many streams that ffmpeg can't route them to the proper rtp endpoint. So what is the best way to delay the audio and can I select which streams map to rtp endpoints ?

  



  ffmpeg \
-format_code 23ps \
-f decklink \
-i "DeckLink HD Extreme 3" \
-filter_complex "[0:a] adelay=2s|2s [delayed]" \
-map [delayed] -map 0:v \
-r 24 \
-g 1 \
-s 1920x1080 \
-quality realtime \
-speed 8 \
-threads 8 \
-row-mt 1 \
-tile-columns 2 \
-frame-parallel 1 \
-qmin 30 \
-qmax 35 \
-b:v 2000k \
-pix_fmt yuv420p \
-c:v libvpx-vp9 \
-strict experimental \
-an -f rtp rtp://myurl.com:5004?pkt_size=1300 \
-c:a libopus \
-b:a 128k \
-vn -f rtp rtp://myurl.com:5002?pkt_size=1300


  



  adding a full log when running the command with out any delay :

  



  -filter_complex "[0:a] adelay=2s|2s [delayed]" \
-map [delayed] -map 0:v \

ffmpeg version N-97362-g889ad93c88 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
  configuration: --prefix=/usr/local --pkg-config-flags=--static --extra-cflags='-fno-stack-check -I/Users/admin/Documents/ffmpeg_build/include -I/Users/admin/Documents/BDS/Mac/include' --extra-ldflags=-L/Users/admin/Documents/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/Users/admin/Documents/ffmpeg_build/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree --enable-decklink
  libavutil      56. 42.102 / 56. 42.102
  libavcodec     58. 80.100 / 58. 80.100
  libavformat    58. 42.100 / 58. 42.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 77.101 /  7. 77.101
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[decklink @ 0x7fcfb2000000] Found Decklink mode 1920 x 1080 with rate 23.98
[decklink @ 0x7fcfb2000000] Frame received (#2) - No input signal detected - Frames dropped 1
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, decklink, from 'DeckLink HD Extreme 3':
  Duration: N/A, start: 0.000000, bitrate: 797002 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
    Stream #0:1: Video: rawvideo (UYVY / 0x59565955), uyvy422(progressive), 1920x1080, 795466 kb/s, 23.98 tbr, 1000k tbn, 1000k tbc
[decklink @ 0x7fcfb2000000] Frame received (#3) - Input returned - Frames dropped 2
Stream mapping:
  Stream #0:1 -> #0:0 (rawvideo (native) -> vp9 (libvpx-vp9))
  Stream #0:0 -> #1:0 (pcm_s16le (native) -> opus (libopus))
Press [q] to stop, [?] for help
[libvpx-vp9 @ 0x7fcfb180d200] v1.8.2
Output #0, rtp, to 'rtp://myurl.com.com:5004?pkt_size=1300':
  Metadata:
    encoder         : Lavf58.42.100
    Stream #0:0: Video: vp9 (libvpx-vp9), yuv420p, 1920x1080, q=30-35, 2000 kb/s, 24 fps, 90k tbn, 24 tbc
    Metadata:
      encoder         : Lavc58.80.100 libvpx-vp9
    Side data:
      cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
Output #1, rtp, to 'rtp://myrul.com:5002?pkt_size=1300':
  Metadata:
    encoder         : Lavf58.42.100
    Stream #1:0: Audio: opus (libopus), 48000 Hz, stereo, s16, 128 kb/s
    Metadata:
      encoder         : Lavc58.80.100 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
t=0 0
a=tool:libavformat 58.42.100
m=video 5004 RTP/AVP 96
c=IN IP4 54.183.58.143
b=AS:2000
a=rtpmap:96 VP9/90000
m=audio 5002 RTP/AVP 97
c=IN IP4 54.183.58.143
b=AS:128
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1

frame=  434 fps= 24 q=0.0 size=   37063kB time=00:00:18.09 bitrate=16780.7kbits/s speed=1.01x   


  


• why ffmpeg php conversion 0 bytes empty

  9 avril 2020, par user3080392

  I'm trying to convert a .wav file to .ogg with php and ffmpeg. The ogg file keeps being created as 0 bytes :

  



  <?php
$ffmpeg = "/usr/local/bin/ffmpeg";
shell_exec("$ffmpeg -y -i clip.wav clip.ogg");
?>


  



  I've tried various parameters for the ogg file, but none work. This simple conversion should work, but it doesn't.

  



  Here is the log :

  



  ffmpeg started on 2020-04-08 at 22:11:58
Report written to "ffmpeg-20200408-221158.log"
Command line:
/usr/local/bin/ffmpeg -y -i clip.wav clip.ogg -report
ffmpeg version N-71954-gbc6f84f Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
configuration: --prefix=/usr --enable-version3 --enable-gpl --enable-shared - 
-enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libvpx -- 
enable-libx264 --enable-libxvid --enable-libopencore-amrwb --enable- 
libopencore-amrnb --enable-postproc --enable-nonfree --enable-pthreads -- 
enable-x11grab --enable-libfaac --enable-libopenjpeg --enable-zlib --disable- 
doc
libavutil      54. 23.101 / 54. 23.101
libavcodec     56. 37.101 / 56. 37.101
libavformat    56. 31.102 / 56. 31.102
libavdevice    56.  4.100 / 56.  4.100
libavfilter     5. 16.101 /  5. 16.101
libswscale      3.  1.101 /  3.  1.101
libswresample   1.  1.100 /  1.  1.100
libpostproc    53.  3.100 / 53.  3.100
Splitting the commandline.
Reading option '-y' ... matched as option 'y' (overwrite output files) with 
argument '1'.
Reading option '-i' ... matched as input file with argument 'clip.wav'.
Reading option 'clip.ogg' ... matched as output file.
Reading option '-report' ... matched as option 'report' (generate a report) 
with argument '1'.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option y (overwrite output files) with argument 1.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file clip.wav.
Successfully parsed a group of options.
Opening an input file: clip.wav.
[wav @ 0xe2fee0] Format wav probed with size=2048 and score=99
[wav @ 0xe2fee0] Before avformat_find_stream_info() pos: 44 bytes read:32768 
seeks:0
[wav @ 0xe2fee0] parser not found for codec pcm_s16le, packets or times may 
be invalid.
[wav @ 0xe2fee0] probing stream 0 pp:14
[wav @ 0xe2fee0] probing stream 0 pp:13
[wav @ 0xe2fee0] probing stream 0 pp:12
[wav @ 0xe2fee0] probing stream 0 pp:11
[wav @ 0xe2fee0] probing stream 0 pp:10
[wav @ 0xe2fee0] probing stream 0 pp:9
[wav @ 0xe2fee0] probing stream 0 pp:8
[wav @ 0xe2fee0] probing stream 0 pp:7
[wav @ 0xe2fee0] probing stream 0 pp:6
[wav @ 0xe2fee0] probing stream 0 pp:5
[wav @ 0xe2fee0] probing stream 0 pp:4
[wav @ 0xe2fee0] probing stream 0 pp:3
[wav @ 0xe2fee0] probing stream 0 pp:2
[wav @ 0xe2fee0] probing stream 0 pp:1
[wav @ 0xe2fee0] probed stream 0
[wav @ 0xe2fee0] parser not found for codec pcm_s16le, packets or times may 
be invalid.
[wav @ 0xe2fee0] All info found
[wav @ 0xe2fee0] After avformat_find_stream_info() pos: 204844 bytes 
read:213916 seeks:0 frames:50
Guessed Channel Layout for  Input Stream #0.0 : stereo
Input #0, wav, from 'clip.wav':
Duration: 00:00:01.11, bitrate: 1536 kb/s
Stream #0:0, 50, 1/48000: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 
Hz, 2 channels, s16, 1536 kb/s
Successfully opened the file.
Parsing a group of options: output file clip.ogg.
Successfully parsed a group of options.
Opening an output file: clip.ogg.
Successfully opened the file.
detected 32 logical cores
[AVFilterGraph @ 0xe25560] Error initializing threading.
[AVFilterGraph @ 0xe25560] Error creating filter 'anull'
Error opening filters!
[AVIOContext @ 0xe72180] Statistics: 0 seeks, 0 writeouts
[AVIOContext @ 0xe2f560] Statistics: 213916 bytes read, 0 seeks


  


• ffmpeg mp4 video cannot be played in old tv

  24 mai 2020, par Rick Brian

  I've convert many video to mp4 using ffmpeg and playing well on my TV.
  
But ever since I change my laptop to a new one, my conversion failed to load on the TV.

  



  I have tried to download old-stable ffmpeg Windows build, I tried also download a win-32 build, no good.
(I'm using x64 laptop with Windows 10 64-bit, just the same like previous laptop)
  
I also tried to add -pix_fmt yuv420p, still no good.

  



  This is code that I used to convert using ffmpeg :
  
ffmpeg -f concat -i "D:\Convert\LISTCAM.TXT" -c:v libx264 -c:a aac -pix_fmt yuv420p "Apr 2nd.mp4"

  



  I also tried to compare using ffmpeg -i between my playable mp4 file with new not-playable mp4,
  
both are just similar except the encoder header.

  



  Playable mp4 :

  



  ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 8.3.1 (GCC) 20190414
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:\2020-DEC\Dec 13-PSTM-001.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf57.83.100
  Duration: 01:01:47.46, start: 0.000000, bitrate: 1586 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x714 [SAR 1071:1072 DAR 120:67], 1450 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc (default)
    Metadata:
      handler_name    : VideoHandler
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 129 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
At least one output file must be specified


  



  Non-playable mp4 :

  



  ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 8.3.1 (GCC) 20190414
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --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
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'D:\Convert\TEST.mp4':
  Metadata:
    major_brand     : isom
    minor_version   : 512
    compatible_brands: isomiso2avc1mp41
    encoder         : Lavf58.20.100
  Duration: 00:00:18.54, start: 0.000000, bitrate: 4483 kb/s
    Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x808, 4517 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc (default)
    Metadata:
      handler_name    : video.264#trackID=1:fps=23.976 - Imported with GPAC 0.5.0-rev
    Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
    Metadata:
      handler_name    : GPAC ISO Audio Handler
At least one output file must be specified


  



  Update :

  



  

  • I did a single-file input instead of a concatenation list.

  


  • I did a codec copy from a playable file, it works, but when I try to re-encode, it does not.
    
ffmpeg -i "Playable-video-file.mp4" -c:v copy -c:a aac "Output.mp4" this works

  


  



  ffmpeg -i "Playable-video-file.mp4" -c:v libx264 -crf 23 -profile:v main -level:v 3.0 -preset:v medium -c:a aac "Output.mp4" and this don't

  



  

  • I did the very same syntax and the same ffmpeg build on a friend's laptop with the same Windows 10 x64 architecture... the output file works fine and playable on my TV...

  


  • Friend's is Intel i5, mine is Intel i7