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• Avconv : Select german stream not highest quality one

  6 novembre 2017, par mblaettermann

  I am converting some input stream from my DVB S2 Card to RTMP.

  Everything works fine after switching to recent avconv and x264 :)

  The only thing I couldn’t find out is, how do I select the right audio stream ?

  The source sometimes has up to 6 audio tracks. Avconv automatically chooses the one with the highest bitrate. However I want to select the "ger" one :

  Here are the streams of ARTE german/french TV Channel for example :

  Input #0, mpegts, from 'http://192.168.1.50:9981/stream/channelid/1035':
  
    Duration: N/A, start: 19083.694722, bitrate: 15576 kb/s
  
    Program 1
  
      Stream #0.0[0xa8], 127, 1/90000: Video: mpeg2video (Main), yuv420p, 544x576 [PAR 32:17 DAR 16:9], 1/50, 15000 kb/s, 25 fps, 90k tb50 tbc
  
      Stream #0.1[0x70](fre), 204, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 192 kb/s
  
      Stream #0.2[0x71](ger), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
  
      Stream #0.3[0x72](eng), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
  
      Stream #0.4[0x73](qaa), 207, 1/90000: Audio: mp2, 48000 Hz, stereo, s16p, 128 kb/s
  
    No Program
  
      Stream #0.5[0x3b], 126, 1/90000: Audio: mp1, 0 channels, s16p

  libav Docs are really not that helpful. Who does now the right syntax ?

  EDIT : I found the -map option : http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option But it is not possible to map by name ? Only by index ?

  Maybe I need to use avprobe then, to find the corrent stream index for "ger".

• Concat Demuxer in FFMPEG Autogen

  21 juin 2021, par JunaidAmjad

  Is there option available for Concat Demuxer in FFMPEG.Autogen library. I tested it in FFMPEG and it works well through the here

  


• Streaming live video from Windows to Android with FFmpeg

  29 avril 2018, par Alan Daniels

  I am capturing live video on a Windows PC and encoding it with FFmpeg. I can stream the content live to another PC using rtsp://[dest_ip:port]/live.sdp as FFmpeg’s output on the sender and using ffplay -rtsp_flags listen rtsp://[dest_ip:port]/live.sdp on the receiver. However, I have to run FFplay before starting the sender. Also, VLC cannot play the rtsp path :

  main debug: net: connecting to 127.0.0.1 port 5555
  
  main error: connection failed: Connection refused by peer
  
  access_realrtsp error: cannot connect to 127.0.0.1:5555
  
  access_realrtsp debug: could not connect to: 127.0.0.1:5555/live.sdp

  However, FFplay and VLC can play something like ffplay rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov.

  On the Android side, I am using the Media API which can play content from a URI. It works with HTTP and RTSP (as far as I know).

  I’ve looked at FFmpeg’s streaming guide : https://trac.ffmpeg.org/wiki/StreamingGuide, but I am still confused about the difference between my path rtsp://[dest_ip:port]/live.sdp and rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov.

  Do I need a streaming server in order to have my content accessible by URI ? Any recommendations since FFserver is depreciated ?