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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

• webM files shows green and purple effects on mobile

  11 octobre 2015, par Naveen Gamage

  I have converted several GIFs to webM files using ffmpeg on my Ubuntu 14.04 server.

  Heres the code I used for conversation.

  ffmpeg -i your_gif.gif -c:v libvpx -crf 12 -b:v 500K output.webm

  source https://gist.github.com/ndarville/10010916

  The problem is converted webM files shows perfectly fine on PCs but on my mobile it shows with green and purple shadows.

  PC

  

  Mobile

  

  I tried changing -crf and -b:v values to their max but nothing happens.

  webM file : http://d1pnsuxwa0it39.cloudfront.net/uploads/comments/webm/4673555.webm

  edit :

  also I can see webM files on some other sites fine. I think this has to do something with the way I convert files.

  edit :

  I have tried another code I found on stackoverflow but still the same.

  ffmpeg -f gif -i infile.gif outfile.mp4

  EDIT :

  If anyone think this has something to do with the way I installed FFMPEG, I followed the steps on FFMPEG official docs.

  https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu

  EDIT :

  Input file :

  http://d1pnsuxwa0it39.cloudfront.net/test/1.gif

  Output file :

  http://d1pnsuxwa0it39.cloudfront.net/test/output.webm

  FFMPEG CLI output

   /home/naveencg/bin/ffmpeg -i 1.gif -c:v libvpx -crf 12 -b:v 500K output.webm
  
  ffmpeg version 2.5.git Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Dec 31 2014 14:37:15 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --prefix=/home/naveencg/ffmpeg_build --extra-cflags=-I/home/naveencg/ffmpeg_build/include --extra-ldflags=-L/home/naveencg/ffmpeg_build/lib --bindir=/home/naveencg/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 19.100 / 56. 19.100
  
    libavformat    56. 16.102 / 56. 16.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  6.100 /  5.  6.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, gif, from '1.gif':
  
    Duration: N/A, bitrate: N/A
  
      Stream #0:0: Video: gif, bgra, 350x169, 25 fps, 25 tbr, 100 tbn, 100 tbc
  
  [libvpx @ 0x1e2bf60] v1.3.0
  
  Output #0, webm, to 'output.webm':
  
    Metadata:
  
      encoder         : Lavf56.16.102
  
      Stream #0:0: Video: vp8 (libvpx), yuva420p, 350x169, q=-1--1, 500 kb/s, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc56.19.100 libvpx
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (gif (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  frame=   21 fps=0.0 q=0.0 size=      58kB time=00:00:00.84 bitrate= 569.7kbits/sframe=   44 fps= 41 q=0.0 size=     110kB time=00:00:01.76 bitrate= 512.4kbits/sframe=   62 fps= 39 q=0.0 size=     153kB time=00:00:02.48 bitrate= 505.9kbits/sframe=   84 fps= 40 q=0.0 size=     210kB time=00:00:03.36 bitrate= 510.8kbits/sframe=   88 fps= 41 q=0.0 Lsize=     218kB time=00:00:03.52 bitrate= 508.3kbits/s
  
  video:216kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.971527%

• Is FFmpegAudioDecoder supposed to reinitialize upon append of new init segment

  14 novembre 2023, par martin

  I am attempting to switch audio tracks but when switching the FFmpegAudioDecoder never reinitializes like it does with video tracks of differing resolutions. I am not certain if this is the intended behavior of FFmpegAudioDecoder and would love to learn more about the expected behavior.

  


  When switching audio tracks I end up calling the following operations :

  


  if sourceBuffer.getIsUpdate() {sourceBuffer.abort()}
sourceBuffer.remove(0-videoDuration)
initSegmentDataStream = fetch init segment of new audio representation
sourceBuffer.appendBuffer(initSegmentDataStream)


  


  These are the Media tab messages from initial video load

  


  ChunkDemuxer
Selected FFmpegAudioDecoder for audio decoding, config: codec: aac, profile: unknown, bytes_per_channel: 2, channel_layout: STEREO, channels: 2, samples_per_second: 48000, sample_format: Signed 16-bit, bytes_per_frame: 4, seek_preroll: 0us, codec_delay: 0, has extra data: false, encryption scheme: Unencrypted, discard decoder delay: false, target_output_channel_layout: STEREO, target_output_sample_format: Unknown sample format, has aac extra data: true
Cannot select DecryptingVideoDecoder for video decoding
Cannot select VDAVideoDecoder for video decoding
Cannot select VpxVideoDecoder for video decoding
Selected Dav1dVideoDecoder for video decoding, config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1280,720], visible rect: [0,0,1280,720], natural size: [1280,720], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}
Dropping audio frame (DTS 0us PTS -105375us,-62709us) that is outside append window [0us,9223372036854775807us).
Dropping audio frame (DTS 42666us PTS -62708us,-20042us) that is outside append window [0us,9223372036854775807us).
Truncating audio buffer which overlaps append window start. PTS -20041us frame_end_timestamp 22625us append_window_start 0us
Effective playback rate changed from 0 to 1


  


  For comparison this is what I get when appending the init segment of a different video resolution / track

  


  video decoder config changed midstream, new config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1920,1080], visible rect: [0,0,1920,1080], natural size: [1920,1080], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}



  


  Chrome version : Version 119.0.6045.123 (Official Build)

  


  When appending the new init segment of an audio track I was expecting the FFmpegAudioDecoder to be reinitialized like the Dav1dVideoDecoder does for video tracks