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• ffmpeg converting MOV to MP4 add 5 second still frames to start of video [closed]

  17 février 2013, par Webfox

  I have a website that automatically converts MOV files to MP4. For some reason the client videos (which are exported from their own software) have about 5-8 seconds worth of a still frame added to the beginning of each video when they are converted. I have tried so many different variations trying to sort this but its always the same.

  The function/ffmpeg command that is currently being used is

      public function convertToMP4($file, $destination, $size=null, $qscale='10', $rate=25, $ab='192k', $ar=44100) {
  
      if (substr($destination, -1) == '/') {
  
          $destination = substr($destination, 0, -1);
  
      }
  
  
  
      $filename = end(explode("/", $file));
  
      $outfile = current(explode(".", $filename)).".mp4";
  
  
  
      // Media Size
  
      if (!isset($size) || $size == 'copy') {
  
          if ($dims = $this->getDimensions($file)) {
  
              $size = $dims['width'].'x'.$dims['height'];
  
          }
  
      }
  
      if ($ab == false) {
  
          $cmd = "ffmpeg -i $file -qscale $qscale -an -f mp4 -s $size -aspect 16:9 $destination/$outfile";
  
      } else {
  
          $cmd = "ffmpeg -i $file -acodec ".$this->mp3lib." -qscale $qscale -r $rate -ar $ar -ab $ab -f mp4 -s $size -aspect 16:9 $destination/$outfile";
  
      }
  
  
  
      $ret = shell_exec($cmd);
  
  
  
      if (file_exists("$destination/$outfile")) {
  
          return "$destination/$outfile";
  
      }
  
      return false;
  
  }

  where if $ab is false then remove the sound which is the case for 99% of the videos in scenario

  Thanks

• 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.

• pyav / ffmpeg / libav select number of P-frames and B-frames

  27 mai 2021, par user1315621

  I am streaming from an rtsp source. It looks like half of the frames received are key frames. Is there a way to reduce this percentage and have an higher number of P-frames and B-frames ? If possible, I would like to increase the number of P-frames (not the one of B-frames).
I am using pyav which is a Python wrapper for libav (ffmpeg)

  


  Code :

  


  container = av.open(
    url, 'r',
    options={
        'rtsp_transport': 'tcp',
        'stimeout': '5000000',
        'max_delay': '5000000',
    }
)
stream = container.streams.video[0]
codec_context = stream.codec_context
codec_context.export_mvs = True
codec_context.gop_size = 25  

for packet in self.container.demux(video=0):
    for video_frame in packet.decode():
        print(video_frame.is_key_frame)


  


  Output :

  


  True
False
True
False
...


  


  Note 1 : I can't edit the source. I can just edit the code used to stream the video.

  


  Note 2 : same solution should apply to pyav, libavi and ffmpeg.

  


  Edit : it seems that B-frames are disabled : codec_context.has_b_frames is False