Recherche avancée

Sur d’autres sites (4313)

• Best practice to play segments of larger video files in HTML5 video format

  8 octobre 2015, par zx98

  I have to migrate a legacy project from Windows Media to HTML5 video/audio. The system should provide video/audio content to the web user meeting following criteria :

  1. Playout : Segments (TcIn to TcOut) of large video/audio files should be played "on the fly" to the web user. (Cutting the source material into segments does not work because there is a high overlap rate between segments, so 3-4 times more HDD space and a lot of extra programming would be needed. Markered playout of the large source files is not an option either as there are forbidden parts of the source material which should never be available to the user).
  2. Cutting : In general, users are permitted to make copies of the material. Indeed, by using the dowload function, users can download selected segments. I guess, this is not a problem : a service starts a cutting software (probably ffmpeg) that cuts the video, and the user is notified by email to download the product. (That’s the way I do it now but there might be a more simple option).
  3. Formats : what is the preferred video/audio format that most browsers (Chrome, Firefox, Opera, Android) support ?
  4. Server : If possible, I would keep Windows Server because the database behind is MS SQL Server.

  (Windows Media Server and WMEncoder currently works fine for all the above, however WM is not suppored by browsers any more. That is the reason for reengineering).

  I would appreciate any of your suggestions on how to rebuild this video/audio system. (What software elements to use and how).

  Thank you.

  -------- addition -------
  Dear commenters,
  Thank you for your comments. Please note that the above is one single question, the rest is just a detailed description of the issue/conditions, as recommended by stackoverflow.com. I guess, the answer is also very simple, I try to write it myself :

  1. You should keep the large video/audio files in xxx format because...
  2. You create a playlist file like this example... for each of the the segments.
  3. You install the xxx streaming software component on the web/media server to playout the segments.
  4. You can address and play the segments from the client side web browser like this :...
  5. For asynchronous cutting, yes, I recommend ffmpeg, try these profile settings.

  Just a few parts in bold are missing. Thanks to anyone who can fill them.

• Changing video resolution is changing the duration of the output video the input video

  4 septembre 2017, par Sean

  I am trying to change the video resolution on a few videos as well as the frame rate using FFMPEG. I am attempting to copy the original video/audio and change the output video to have a constant frame rate and different resolution.

  Here is the command line I execute :

  `ffmpeg -i "input_video_here" -s 854x480 -c:v libx264 -preset slow -crf 18 -r 30 -refs 3 -movflags +faststart -c:a copy "#{output_video_here}"`

  When I compare the input video with the output video, my output videos are getting cut. The duration of my output video is 51 seconds and my input is 5 minutes. Is something in my ffmpeg command causing my output video to clip ?

• Streaming UDP packets to two different ports (for video and audio). Video works fine, but the audio does not show

  8 avril 2018, par Winston Chen

  I am taking a rtsp stream, split the video and audio out, and stream them to two different ports respectively using gstreamer so that my ffserver would be able to display the stream on my browser.

  My gstreamer pipeline :

  gst-launch-1.0 -v rtspsrc location=rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov latency=300 timeout=0 drop-on-latency=true rtp-blocksize=4096 name=rtsp_source ! \
  
    queue ! capsfilter caps="application/x-rtp,media=video" ! rtph264depay ! h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=127.0.0.1 port=9527 rtsp_source. ! \
  
    queue ! rtpmp4apay pt=97 ! udpsink host=127.0.0.1 port=9327

  Here comes the sdp and my ffmpeg commnad :

  m=video 9527 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  c=IN IP4 127.0.0.1
  
  
  
  m=audio 9327 RTP/AVP 97
  
  a=rtpmap:97 mpeg4-generic/48000/6
  
  c=IN IP4 127.0.0.1

  ffmpeg -protocol_whitelist "file,tcp,rtp,udp" -i ~/test.sdp -max_muxing_queue_size 1024 http://localhost:8090/feed1.ffm

  And finally, this is my ffserver config (the important part) :

  <feed>               # This is the input feed where FFmpeg will send
  
     File ./feed1.ffm            # video stream.
  
     FileMaxSize 1GB             # Maximum file size for buffering video
  
     ACL allow 127.0.0.1         # Allowed IPs
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed feed1.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # NoDefaults
  
     # NoAudio
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 240x160           # Video resolution
  
     VideoFrameRate 10           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
  
  
     PreRoll 0
  
     StartSendOnKey
  
     VideoGopSize 12
  
     VideoBitRate 256
  
  </stream>

  The thing is that if I take away the audio part and apply NoAudio, the video streams fine. However, I could not get the audio to work. Am I doing anything wrong ?