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• Use ffmpeg to stream live content to azure media services

  9 juin 2016, par Dadicool

  I’ve been trying to stream content to azure media services using ffmpeg as it’s one of the options described here : http://azure.microsoft.com/blog/2014/09/18/azure-media-services-rtmp-support-and-live-encoders/

  My command is :

  ffmpeg -v verbose -i 300.mp4 -strict -2 -c:a aac -b:a 128k -ar 44100 -r 30 -g 60 -keyint_min 60 -b:v 400000 -c:v libx264 -preset medium -bufsize 400k -maxrate 400k -f flv rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7

  I have made sure that the streaming endpoint has one active streaming unit, I also made sure that the channel is actually Ready and I even get it to start streaming (which makes a PublishURL available).

  When I execute the ffmpeg command to start streaming, I keep getting the following error :

  ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers
 built on Dec 30 2014 11:31:18 with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
 configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 --arch=x86_64 --enable-runtime-cpudetect
 libavutil      54. 15.100 / 54. 15.100
 libavcodec     56. 13.100 / 56. 13.100
 libavformat    56. 15.102 / 56. 15.102
 libavdevice    56.  3.100 / 56.  3.100
 libavfilter     5.  2.103 /  5.  2.103
 libswscale      3.  1.101 /  3.  1.101
 libswresample   1.  1.100 /  1.  1.100
 libpostproc    53.  3.100 / 53.  3.100
Routing option strict to both codec and muxer layer
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] overread end of atom 'colr' by 1 bytes
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] stream 0, timescale not set
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f9a0a002c00] max_analyze_duration 5000000 reached at 5003637 microseconds
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '300.mp4':
 Metadata:
   major_brand     : mp42
   minor_version   : 0
   compatible_brands: mp42isomavc1
   creation_time   : 2014-01-11 05:39:32
   genre           : Trailer
   artist          : Warner Bros.
   title           : 300: Rise of an Empire - Trailer 2
   encoder         : HandBrake 0.9.9 2013051800
   date            : 2014
 Duration: 00:02:33.24, start: 0.000000, bitrate: 7377 kb/s
   Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, bt709), 1920x1080 (1920x1088), 7219 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc (default)
   Metadata:
     creation_time   : 2014-01-11 05:39:32
     encoder         : JVT/AVC Coding
   Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 157 kb/s (default)
   Metadata:
     creation_time   : 2014-01-11 05:39:32
   Stream #0:2: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 101x150 [SAR 72:72 DAR 101:150], 90k tbr, 90k tbn, 90k tbc
rtmp://nessma-****.channel.mediaservices.windows.net:1935/live/584c99f5c47f424d9e83ac95364331e7: Input/output error

  The Azure blog post clearly states that this should be possible but I can’t find a working example anywhere.

  Environment :

  • MacOS Maverick
  • FFMPEG installed from official build
  • 300.mp4 : 1080p trailer of the latest 300 movie

• NodeJS/Dokku/Docker : ffmpeg exited with code 127

  2 mars 2015, par XHR

  I’m trying to run ffmpeg inside of a Docker container via Dokku. I’ve heavily modified a Dokku plugin to accomplish this as I need h264 support. I currently get an error from fluent-ffmpeg which states

  Error : ffmpeg exited with code 127

  I’ve tried to look up what error code 127 is but haven’t found much. Some people seem to suggest that it’s some sort of issue with finding ffmpeg. However, I did have a libx264 error from ffmpeg earlier in the process so I’m pretty sure the ffmpeg binary is being found. Unfortunately fluent doesn’t send the used command line for ffmpeg to the error event so I’m not sure what the exact command being ran is.

• libavformat/avienc : Fix duration of audio segment in OpenDML master index

  3 mars 2015, par Tobias Rapp

  libavformat/avienc : Fix duration of audio segment in OpenDML master index
  Fixes the duration field of the OpenDML master index "indx" chunk to
  
  contain the number of samples instead of the number of packets for
  
  (linear/PCM) audio streams.
  This matches the OpenDML V1.02 standard text which states that the
  
  duration field shall contain "time span in stream ticks".
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavformat/avienc.c