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• Use ffmpeg to stream live content to azure media services

  19 mars 2018, 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

• 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

• FFmpeg, how to skip late input ?

  14 novembre 2017, par user3343357

  I’m running ffmpeg to display incoming stream on a Decklink BlackMagic card with the following command line :

  ffmpeg -y -f ourFmt -probesize 32 -i - -f decklink -preset ultrafast 
  
     -pix_fmt uyvy422 -s 1920x1080 -r 30 -af volume=0.1 -max_delay 10000 
  
       DeckLink Mini Monitor

  Basically I get the video over the internet by UDP and stream it to ffmpeg stdin. Both audio and video streams have pts and dts and are fully in sync, if the connection is good there is no problems.

  However if there are issues with the connection i start getting errors, sometimes the video delay grows significantly, and audio stops working.
  The errors i get are :

  ffmpeg : [decklink @ 0x26cc600] There are not enough buffered video
  frames. Video may misbehave ! ffmpeg : [decklink @ 0x26cc600] There’s no
  buffered audio. Audio will misbehave ! ffmpeg : Last message
  repeated 4 times ffmpeg : [decklink @ 0x26cc600] There are not enough
  buffered video frames. Video may misbehave ! ffmpeg : [decklink @
  0x26cc600] There’s no buffered audio. Audio will misbehave ! ffmpeg :
  Last message repeated 3 times ffmpeg : frame= 5204 fps= 30 q=-0.0
  size=N/A time=00:02:53.76 bitrate=N/A dup=385 drop=5 speed=0.993x
  ffmpeg : [decklink @ 0x26cc600] There’s no buffered audio. Audio will
  misbehave ! ffmpeg : Last message repeated 18 times ffmpeg :
  [decklink @ 0x26cc600] There are not enough buffered video frames.
  Video may misbehave ! ffmpeg : [decklink @ 0x26cc600] There’s no
  buffered audio. Audio will misbehave !

  The problem is when the connection is back to normal, the video keeps misbehaving until I restart the stream. What I want to do is for FFmpeg to skip to the content of the last second and play synchronized video from there, drop all the late data in between, is it possible ?