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• FFMPEG - can you have comments in script files ?

  25 mai, par rossmcm

  Because of batch's poor ability to handle multi-line strings, I'm using FFMPEG script files more and more for filter_complex arguments. I'm wondering if there is any way to include comments in these files. They seem to be quite tolerant of white space and line breaks. I've tried :

  


  #
#   comment
#


  


  and

  


  ;
;   comment
;


  


  and

  


  `
`   comment
`


  


  and

  


  '
'   comment
'


  


  with no success. It doesn't look as if comments are officially supported, but I'm wondering of there is some syntax quirk that allows them ?

  


  *** CLARIFICATION ***

  


  I'm not talking about comments in batch files here, but comments in filter_complex_script files. FFMPEG supports specifying the filter_complex argument string in a separate file. So instead of :

  


  ffmpeg -i "input.mp4" -filter_complex "[0:v] crop=w=100:h=100:x=12:y=34 " "output.mp4"


  


  you can specify :

  


  ffmpeg -i "input.mp4" -filter_complex_script "crop.txt" "output.mp4"


  


  where crop.txt contains

  


  [0:v] crop=w=100:h=100:x=12:y=34


  


  This is really handy when the filter_complex string gets more complicated (I routinely deal with arguments 10,000 characters in length).

  


• Why does use of H264 in sender/receiver pipelines introduce just HUGE delay ?

  24 janvier 2012, par Serguey Zefirov

  When I try to create pipeline that uses H264 to transmit video, I get some enormous delay, up to 10 seconds to transmit video from my machine to... my machine ! This is unacceptable for my goals and I'd like to consult StackOverflow over what I (or someone else) do wrong.

  I took pipelines from gstrtpbin documentation page and slightly modified them to use Speex :

  This is sender pipeline :
  # !/bin/sh

  gst-launch -v gstrtpbin name=rtpbin \
  
          v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
  
                    rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000                            \
  
                    rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false    \
  
                    udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
  
          pulsesrc ! audioconvert ! audioresample  ! audio/x-raw-int,rate=16000 !    \
  
                    speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1                   \
  
                    rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002                            \
  
                    rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false    \
  
                    udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

  Receiver pipeline :

   !/bin/sh

  gst-launch -v\
  
      gstrtpbin name=rtpbin                                          \
  
      udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H263-1998" \
  
              port=5000 ! rtpbin.recv_rtp_sink_0                                \
  
          rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                    \
  
       udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
  
       rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
  
      udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \
  
              port=5002 ! rtpbin.recv_rtp_sink_1                                \
  
          rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \
  
       udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
  
       rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false

  Those pipelines, a combination of H263 and Speex, work fine enough. I snap my fingers near camera and micropohne and then I see movement and hear sound at the same time.

  Then I changed pipelines to use H264 along the video path.

  The sender becomes :
  # !/bin/sh

  gst-launch -v gstrtpbin name=rtpbin \
  
          v4l2src ! ffmpegcolorspace ! x264enc bitrate=300 ! rtph264pay ! rtpbin.send_rtp_sink_0 \
  
                    rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=5000                            \
  
                    rtpbin.send_rtcp_src_0 ! udpsink host=127.0.0.1 port=5001 sync=false async=false    \
  
                    udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
  
          pulsesrc ! audioconvert ! audioresample  ! audio/x-raw-int,rate=16000 !    \
  
                    speexenc bitrate=16000 ! rtpspeexpay ! rtpbin.send_rtp_sink_1                   \
  
                    rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002                            \
  
                    rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false async=false    \
  
                    udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

  And receiver becomes :
  # !/bin/sh

  gst-launch -v\
  
      gstrtpbin name=rtpbin                                          \
  
      udpsrc caps="application/x-rtp,media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \
  
              port=5000 ! rtpbin.recv_rtp_sink_0                                \
  
          rtpbin. ! rtph264depay ! ffdec_h264 ! xvimagesink                    \
  
       udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
  
       rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
  
      udpsrc caps="application/x-rtp,media=(string)audio, clock-rate=(int)16000, encoding-name=(string)SPEEX, encoding-params=(string)1, payload=(int)110" \
  
              port=5002 ! rtpbin.recv_rtp_sink_1                                \
  
          rtpbin. ! rtpspeexdepay ! speexdec ! audioresample ! audioconvert ! alsasink \
  
       udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
  
       rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5007 sync=false async=false

  This is what happen under Ubuntu 10.04. I didn't noticed such huge delays on Ubuntu 9.04 - the delays there was in range 2-3 seconds, AFAIR.

• rtmp : Follow Flash player numbering for channels.

  16 septembre 2013, par Josh Allmann

  rtmp : Follow Flash player numbering for channels.
  Channel 4 is typically used by the Flash player to transmit
  
  audio, channel 6 for video, and various stream-specific invokes
  
  get sent over channel 8, which is designated the source channel.
  This more closely matches the behavior of the Flash player,
  
  including the transmission of play requests over channel 8.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavformat/rtmppkt.h
  • [DH] libavformat/rtmpproto.c