Sur d’autres sites (1635)

• WebVTT Audio Descriptions for Elephants Dream

  10 mars 2015, par silvia

  When I set out to improve accessibility on the Web and we started developing WebSRT – later to be renamed to WebVTT – I needed an example video to demonstrate captions / subtitles, audio descriptions, transcripts, navigation markers and sign language.

  I needed a freely available video with spoken text that either already had such data available or that I could create it for. Naturally I chose “Elephants Dream” by the Orange Open Movie Project , because it was created under the Creative Commons Attribution 2.5 license.

  As it turned out, the Blender Foundation had already created a collection of SRT files that would represent the English original as well as the translated languages. I was able to reuse them by merely adding a WEBVTT header.

  Then there was a need for a textual audio description. I read up on the plot online and finally wrote up a time-alignd audio description. I’m hereby making that file available under the Create Commons Attribution 4.0 license. I’ve added a few lines to the medadata headers so it doesn’t confuse players. Feel free to reuse at will – I know there are others out there that have a similar need to demonstrate accessibility features.

• Muxing in audio to gstreamer RTMP stream kills both video and Audio

  1er avril 2015, par Adam

  I need some genius help here - I’m trying to set up a live stream for my upcoming wedding... and I have it ALMOST working - audio seems to be the problem.

  This is my setup

  • Raspberry Pi Model B+
  • Logitech C920 (with onboard h264 encoding that I am utilising)
  • on-camera (C920) microphone
  • USB wifi to iPhone 4G connection
  • gstreamer1.0
  • Amazon EC2 Wowza RTMP server

  I have it all set up, but as soon as I mux in the audio, the streams wont play by any player.

  What Works :
  - my gstreamer pipeline WITHOUT the audio muxed in
  - Wowza receives a consistent stream, no failures
  - The various Flash players / iOS / Android and VLC all play back the video

  What doesnt :
  - enabling audio in the mux (using the pipeline below)
  - BUT gstreamer doesnt complain
  - BUT Wowza receives a consistent stream, no failures
  - The various flash players fail to play both Audio and Video. some just display the first video frame
  - VLC plays 1 video frame, and about 100ms of audio, then stops

  Ideally I’d like the muxed audio/video FLV stored on the SD card too in case the network goes down - but if the ’tee’ needs to be sacrificed to make it work, so be it.

  This is my current FAILING pipeline - I assume there’s something really stupid in it because I know practically nothing about gstreamer.... The first frame loads in all the players (except iOS.. which never shows anything)

  # set camera resolution to 720p, and the data format to H264 (alternatives are YUV and JPG)
  
  v4l2-ctl --device=/dev/video0 --set-fmt-video=width=1280,height=720,pixelformat=1
  
  # set the frame rate
  
  v4l2-ctl --device=/dev/video0 --set-parm=10
  
  
  
  gst-launch-1.0 -v -e uvch264src initial-bitrate=300000 average-bitrate=300000 device=/dev/video0 name=src auto-start=true src.vidsrc \
  
                  ! queue \
  
                  ! video/x-h264,width=1280,height=720,framerate=10/1 \
  
                  ! h264parse \
  
                  ! flvmux streamable=true name=mux \
  
                  ! queue \
  
                  ! tee name=t \
  
                  ! queue \
  
                  ! filesink location=/home/pi/wedding.flv t. \
  
                  ! queue \
  
                  ! rtmpsink location='rtmp://wowzaserver/live/wedding live=1' >>/home/pi/wedding.log 2>&1

  Some of the things I can’t really afford to change at this late stage are the encapsulation (FLV) and wowza RTMP because I’ve built everything around that...

  Please Help !! Thanks !

  UPDATE

  Given that I am also saving the FLV file, I have found that if I use ffmpeg to send that FLV file (using audio copy, video copy) to the RTMP server, everything works (but obviously its not live) ! So I am now starting to believe this is a problem with the way Gstreamer encapsulates RTMP - and by putting ffmpeg in the middle it fixes it... but it’s not live of course.
  Is it possible to pipe my output to ffmpeg and using ffmpeg’s RTMP ?

• An H264 encoded MP4 file cannot be played by VLC/HTML5 browsers, is there any header or meta data I can add to fix it ? [migrated]

  5 avril 2015, par user534498

  I have an MP4 file, which is H264 encoded. I saved the file from RTP streams, with sprop-property-sets in front, and followed by frames. (Note. VLC can directly play the RTP stream without problem.)

  The file can be played by "Videos" program in Linux. If I play it with VLC 2.1.4 in Linux, it shows the following error.

  No suitable decoder module : VLC does not support the audio or video
  format "undf". Unfortunately there is no way for you to fix this.

  I then open VLC, from preference menu, I change the Demuxer (demultiplexer) from automatic to "H264 Video Demuxer". VLC can then play it.

  So it seems that the problem is VLC cannot automatically detect a demuxer from the video file. However, I am sure that my video file has only video data, and it doesn’t even have audio (may not even need a demuxer ?)

  I put this file in HTTP server, and use "video" tag for testing, all IE, chrome and firefox browsers cannot play this file, and I guess it’s the same reason as VLC cannot play it.

  So is there a way to fix it ? For example, is there any place I can add a header to tell VLC or similar players to choose H264 video demuxer ?

  Thanks.