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• 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 ?

• Progress with rtc.io

  12 août 2014, par silvia

  At the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.

  

  Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.

  

  One of the most exciting opportunities is still under-exploited : the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.

  Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.

  

  We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone ?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections :

  

  For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.

  For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.

  

  So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course : enjoy WebRTC ! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team !

  On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating !!

• Is there a set of working P/Invoke declarations for FFMpeg, libavutil, libavformat and libavcodec in .NET ?

  30 août 2011, par casperOne

  I'm currently looking to access libavutil, libavformat and libavcodec (all part of FFMpeg) from .NET.

  Currently, I'm getting the libraries from the automated builds of the shared FFMpeg package performed every night for Windows 32-bit.

  I am also using the code from the ffmpeg-sharp project. In that project, I have removed a number of classes that were not compiling (they are wrapper classes not the P/Invoke declarations).

  The code compiles fine, but I am running into a few issues.

  First, it appears that the build of av*.dll uses the cdecl calling convention, as I was receiving a number of PInvokeStackImbalanceException when trying to call av_open_input_file. This was easy enough to change to get it to work right. The AVFormatContext structure is populated.

  After that, I want to call av_find_stream_info to get information about the streams in the file. However, when calling that with the AVFormatContext retrieved from the call to av_open_input_file, an AccessViolationException is thrown indicating that I am trying to read or write from protected memory.

  Has anyone used P/Invoke to access the libavutil, libavformat and libavcodec dll libraries through P/Invoke and have gotten it to work ?

  I should mention that working with the command-line version of FFMpeg, while a solution, is not a viable solution in this case, access needs to occur through the libraries. The reason for this is that I'd have to thrash the disk way too much to do what I need to do (I have to do a frame-by-frame analysis of some very high definition video) and I want to avoid the disk as much as possible.