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• ffmpeg pipe output and script [duplicate]

  27 juillet 2018, par James Magnus

  This question already has an answer here :

  • Execute command in a variable don’t execute the latter part of a pipe
    
    1 answer
    

  I need to extract raw video and pipe it to another program. This runs perfectly in a terminal :

  ffmpeg -i SampleVideo_1280x720_5mb.mp4 -s 1920x1080 -c:v rawvideo -pix_fmt yuv420p -f rawvideo - | ffplay -f rawvideo -pix_fmt yuv420p -s 1920x1080 -

  But when I try to run this command from a script it fails. Here is a very simplified version of the script :

  #!/bin/bash
  
  
  
  c='ffmpeg -i SampleVideo_1280x720_5mb.mp4 -s 1920x1080 -c:v rawvideo -pix_fmt yuv420p -f rawvideo - | ffplay -f rawvideo -pix_fmt yuv420p -s 1920x1080 -'
  
  $c

  The output

  ~/ets/videos$ ./test.sh 
  
  ffmpeg version N-91403-gd24c9e5 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.10) 20160609
  
    configuration: --extra-cflags='-fPIC -I/home/jlbeaussart/ffmpeg_build/include' --extra-ldflags=-L/home/jlbeaussart/ffmpeg_build/lib --extra-libs='-lpthread -lm' --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-shared --enable-libcaca --enable-libxvid --enable-libxml2 --enable-opencl --enable-opengl --enable-cuvid --enable-ffnvcodec --enable-libdrm --enable-nvenc --enable-nvdec --enable-vaapi --enable-vdpau --enable-nonfree
  
    libavutil      56. 18.102 / 56. 18.102
  
    libavcodec     58. 20.104 / 58. 20.104
  
    libavformat    58. 17.101 / 58. 17.101
  
    libavdevice    58.  4.101 / 58.  4.101
  
    libavfilter     7. 25.100 /  7. 25.100
  
    libswscale      5.  2.100 /  5.  2.100
  
    libswresample   3.  2.100 /  3.  2.100
  
    libpostproc    55.  2.100 / 55.  2.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SampleVideo_1280x720_5mb.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      creation_time   : 1970-01-01T00:00:00.000000Z
  
      encoder         : Lavf53.24.2
  
    Duration: 00:00:29.57, start: 0.000000, bitrate: 1421 kb/s
  
      Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 1032 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
  
      Metadata:
  
        creation_time   : 1970-01-01T00:00:00.000000Z
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 383 kb/s (default)
  
      Metadata:
  
        creation_time   : 1970-01-01T00:00:00.000000Z
  
        handler_name    : SoundHandler
  
  [NULL @ 0x18d5d40] Unable to find a suitable output format for '|'
  
  |: Invalid argument

• convert G.723.1 to normal wav and split channels ?

  10 août 2019, par user9316498

  I have a wav file who’s info is

  ion@aurora:~/Inbound$ mediainfo 48401-3405-48403--18042018170000.wav 
  
  General
  
  Complete name                            : 48401-3405-48403--18042018170000.wav
  
  Format                                   : Wave
  
  File size                                : 327 KiB
  
  Duration                                 : 4mn 11s
  
  Overall bit rate                         : 10.7 Kbps
  
  
  
  Audio
  
  Format                                   : G.723.1
  
  Codec ID                                 : A100
  
  Duration                                 : 4mn 11s
  
  Bit rate                                 : 10.7 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 8 000 Hz
  
  Stream size                              : 327 KiB (100%)

  and the audacity shows like this
  

  To tackle it I tried

  ffmpeg -i 48401-3405-48403--18042018170000.wav  -f wav test.wav

  But it just appends the second channel at the back of first channel , So I get only single channel.
  and its media info is :
  ion@aurora : /Inbound$ mediainfo test.wav

  General
  
  Complete name                            : test.wav
  
  Format                                   : Wave
  
  File size                                : 7.67 MiB
  
  Duration                                 : 8mn 22s
  
  Overall bit rate mode                    : Constant
  
  Overall bit rate                         : 128 Kbps
  
  Writing application                      : Lavf56.40.101
  
  
  
  Audio
  
  Format                                   : PCM
  
  Format settings, Endianness              : Little
  
  Format settings, Sign                    : Signed
  
  Codec ID                                 : 1
  
  Duration                                 : 8mn 22s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 1 channel
  
  Sampling rate                            : 8 000 Hz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 7.67 MiB (100%)

  Any idea how to solve this ?

  PS : When I hear the original wav file, It feels like the talking is too fast.

  This is how the media info of processed file should look like.

  ion@aurora:~/Downloads/Call Recordings$ mediainfo 9417648939\ \(G\ Call\)_Done.wav 
  
  General
  
  Complete name                            : 9417648939 (G Call)_Done.wav
  
  Format                                   : Wave
  
  File size                                : 8.01 MiB
  
  Duration                                 : 8mn 44s
  
  Overall bit rate mode                    : Constant
  
  Overall bit rate                         : 128 Kbps
  
  
  
  Audio
  
  Format                                   : ADPCM
  
  Format profile                           : U-Law
  
  Codec ID                                 : 7
  
  Codec ID/Hint                            : CCITT
  
  Duration                                 : 8mn 44s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 8 000 Hz
  
  Bit depth                                : 8 bits
  
  Stream size                              : 8.01 MiB (100%)

• FFmpeg -> JSMpeg Websocket Closes Repeatedly

  13 mars 2018, par Kyle Martin

  I’m trying to create a fairly simple streaming server/site. Here’s the current flow :

  • OBS streams to an RTMP URL
  • Nginx accepts the RTMP stream and uses exec-push to have FFmpeg pick up the stream and transcode it
  • FFmpeg transcodes the stream and outputs it to a JSMpeg application, which displays the stream on a webpage.

  When I have my exec_push statement as follows, everything seems to work perfectly, except the browser says Possible garbage data. Skipping. on every frame it receives :

  exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -f mpeg1video  http://localhost:8080/supersecret;

  This behavior is understandable, because JSMpeg must receive MPEG-TS data, not MPEG1 data. It sees the MPEG1 frames and thinks they’re garbage.

  So through some online research, I found this :

  exec_push /usr/bin/ffmpeg -re -i rtmp://127.0.0.1:1935/$app/$name -c:v copy -c:a copy -f mpegts http://localhost:8080/supersecret;

  Supposedly, this is supposed to transcode my RTMP stream into an MPEG-TS format, which should be compatible with JSMpeg.

  However, with the second version of the command, my FFmpeg -> JSMpeg stream keeps connecting and disconnecting, connecting and disconnecting, and so on. This behavior is observed in terminal :

  Stream Connected: ::1:40208
  
  close
  
  Stream Connected: ::1:40212
  
  close
  
  Stream Connected: ::1:40216
  
  close
  
  Stream Connected: ::1:40220
  
  close
  
  Stream Connected: ::1:40224
  
  close
  
  ...

  What would cause this ? I am pretty certain the issue is in my exec_push command. OBS is perfectly content, which tells me that the stream is making it to the server, and if I do a push, I can do a test push to Ustream just fine, which tells me that Nginx is at least processing the stream with some reasonable degree of success.

  ---

  Disclaimer : I have no idea what I’m talking about. Everything I know about FFmpeg and JSMpeg/Node is from snippets of code that I found online.