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• FFMPEG - Stream discovered after head already parsed [on hold]

  7 février 2014, par John Doe

  I am trying to live transcode an RTMP stream to another RTMP HLS stream using the following command :

  ffmpeg -re -i rtmp://localhost/videochat/testing -c:v libx264 -c:a:0 libfaac -b:a:0 480k -f flv rtmp://localhost:12345/hls/mystream;

  However I receive the following error and the transcoding never begins :

  ffmpeg version git-2014-02-06-474db7a Copyright (c) 2000-2014 the FFmpeg developers
  
  built on Feb  6 2014 22:20:14 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  
  configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --               enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
  
  libavutil      52. 63.100 / 52. 63.100
  
  libavcodec     55. 49.101 / 55. 49.101
  
  libavformat    55. 30.100 / 55. 30.100
  
  libavdevice    55.  7.100 / 55.  7.100
  
  libavfilter     4.  1.102 /  4.  1.102
  
  libswscale      2.  5.101 /  2.  5.101
  
  libswresample   0. 17.104 /  0. 17.104
  
  libpostproc    52.  3.100 / 52.  3.100
  
  Metadata:
  
  description           Chat using VideoChat example.
  
  [flv @ 0x1ec89e0] Stream discovered after head already parsed
  
  ^C[flv @ 0x1ec89e0] Could not find codec parameters for stream 0 (Video: none):    unspecified size
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  Input #0, flv, from 'rtmp://localhost/videochat/testing':
  
  Metadata:
  
  description     : Chat using VideoChat ex   ?5P
  
  Duration: N/A, start: 0.000000, bitrate: N/A
  
  Stream #0:0: Video: none, 1k tbr, 1k tbn, 1k tbc
  
  Stream #0:1: Data: none
  
  Codec AVOption b (set bitrate (in bits/s)) specified for output file #0 (rtmp://localhost:12345/hls/mystream) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream.
  
  Output #0, flv, to 'rtmp://localhost:12345/hls/mystream':

  If anyone has ever dealt/solved this problem before, can you please share as I have been trying to solve this for 2 days but to no avail !

• ffmepg always get broken video streaming from rtsp protocol [on hold]

  6 janvier 2014, par poc

  I'm trying to capture rtsp streaming from Ip Camera by the command

  ffmpeg -t 10 -i  rtsp://172.19.1.42/live.sdp  -ss 00:00:02.500  -c:v copy download.mp4

  However I always get the broken streaming, but if I used the VLC or Qucicktime to watch the RTSP streaming, it worked fine , there was no broken streaming.

  Is there any setting or options for ffmpeg can improve the streaming quality ? Thanks

  [rtsp @ 0x7f9a84017c00] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, rtsp, from 'rtsp://172.19.1.42/live.sdp':
  
    Metadata:
  
      title           : RTSP server
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Main), yuv420p, 2048x1536, 13.33 tbr, 90k tbn, 180k tbc
  
  -t is not an input option, keeping it for the next output; consider fixing your command line.
  
  Output #0, mp4, to 'c0_s1_h264_1920x1536_10_cbr_500_6000000_imagequality.mp4':
  
    Metadata:
  
      title           : RTSP server
  
      encoder         : Lavf54.63.104
  
      Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 2048x1536, q=2-31, 90k tbn, 90k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
      Last message repeated 1 times
  
  RTP: missed 3 packets=-1.0 size=     521kB time=00:00:00.80 bitrate=5303.8kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
  RTP: missed 3 packets=-1.0 size=     895kB time=00:00:01.40 bitrate=5224.0kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
  RTP: missed 3 packets=-1.0 size=    1271kB time=00:00:02.00 bitrate=5192.3kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
  RTP: missed 3 packets=-1.0 size=    1646kB time=00:00:02.60 bitrate=5175.1kbits/s
  
  RTP: missed 2 packets=-1.0 size=    2029kB time=00:00:03.20 bitrate=5183.5kbits/s
  
  RTP: missed 3 packets=-1.0 size=    2420kB time=00:00:03.80 bitrate=5207.3kbits/s
  
  RTP: missed 2 packets=-1.0 size=    2801kB time=00:00:04.40 bitrate=5206.0kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
  RTP: missed 2 packets=-1.0 size=    3180kB time=00:00:05.00 bitrate=5201.6kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 3 packets
  
  RTP: missed 3 packets=-1.0 size=    4708kB time=00:00:07.41 bitrate=5204.6kbits/s
  
  RTP: missed 2 packets=-1.0 size=    5088kB time=00:00:08.01 bitrate=5202.7kbits/s
  
  RTP: missed 2 packets=-1.0 size=    5857kB time=00:00:09.21 bitrate=5207.3kbits/s
  
  [NULL @ 0x7f9a83805200] RTP: missed 2 packets
  
  frame=  100 fps= 10 q=-1.0 Lsize=    6382kB time=00:00:09.96 bitrate=5246.4kbits/s

• Raspberry Pi HLS streaming [on hold]

  5 mars 2015, par Razvan-Catalin Olaru

  I have a rev. B Raspberry Pi and want to stream from my Microsoft Lifecam VX-800 to a web server using HLS protocol and avconv/ffmpeg, preferably without using raspivid because it’s not working (couldn’t figure out why). The stream needs to be fed to jwplayer. I went through a lot of guides but none seem to be fitting my needs. Could you give me detailed instructions on how to do the live streaming or give me links to detailed tutorials that you tried and worked ?

  Thanks.