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• live stream with ffmpeg and IP camera

  23 octobre 2013, par hamilton.lima

  I'm reading a live stream from a DKSEG IP camera, but the results are not good at all ...

  Here is the command line used for this :

  ./ffmpeg -i "http://192.168.0.29/videostream.asf?user=admin&pwd=&resolution=64&rate=0" -vcodec libx264 -r 30 -s 640x480 -threads 2 -vprofile baseline -vpre zoom -strict experimental -acodec aac -ab 96000 -ar 48000 -ac 1  -f rtsp rtsp://xxx.xxx.xxx.xxx:1935/live/_definst_/c3

  the zoom.ffpreset :

  coder=1
  
  level=31
  
  maxrate=1200000
  
  bufsize=200000
  
  wpredp=0
  
  g=60
  
  refs=1
  
  subq=3
  
  trellis=0
  
  bf=0
  
  rc_lookahead=0

  Even when I save to a mp4 file the results are not good, with some gaps in the video.

  Any toughts ? suggestions ?

• Workaround for webbrowsers buffering live webm streams and introducing a delay (how to play live webm stream)

  8 juillet 2014, par CoryG

  I have webm streams coming from an ffserver (16 streams), managed to get all 16 to load at once within a browser, but whether I am playing just one or 16 there is a delay of 3-5 seconds introduced due to either buffering or the handshake plus some kind of compensation to try and sync the timestamps by the browser itself, which leads to my question :

  Is it possible to force a browser playing a webm stream via HTML5’s video tag (I only really care about Chrome, but it’s happening in Firefox too) to play from the actual most recent frame (clients and servers share a common NTP server over the LAN, so they are guaranteed to be synced, if that is a factor) as opposed to maintaining a buffer of several seconds and playing from the start of that buffer ?

• How do I alter my FFMPEG command to make my HTTP Live Streams more efficient ?

  17 octobre 2014, par Robert

  I want to reduce the muxing overhead when creating .ts files using FFMPEG.

  Im using FFMPEG to create a series of transport stream files used for HTTP live streaming.

  ./ffmpeg -i myInputFile.ismv \
  
           -vcodec copy \
  
           -acodec copy \
  
           -bsf h264_mp4toannexb \
  
           -map 0 \
  
           -f segment \
  
           -segment_time 10\
  
           -segment_list_size 999999 \
  
           -segment_list output/myVarientPlaylist.m3u8 \
  
           -segment_format mpegts \
  
           output/myAudioVideoFile-%04d.ts

  My input is in ismv format and contains a video and audio stream :

  Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 320x240, 348 kb/s, 29.97 tbr, 10000k tbn, 59.94 tbc
  
  Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 63 kb/s

  There is an issues related to muxing that is causing a large amout of overhead to be added to the streams. This is how the issue was described to me for the audio :

  

  So for a given aac stream, the overhead will be 88% (since 200 bytes will map to 2 x 188 byte packets).

  For video, the iframe packets are quite large, so they translate nicely into .ts packets, however, the diffs can be as small as an audio packet, therefore they suffer from the same issue.

  The solution is to combine several aac packets into one larger stream before packaging them into .ts. Is this possible out of the box with FFMPEG ?