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• 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 !!

• ffserver leave original stream size

  28 novembre 2014, par ihnatkuk

  Hope you guys will help me, because I have got stuck and can’t find solution for this problem by myself.
  I am trying to stream video from webcam to users using ffmpeg+ffserver. But I have faced with a problem :

  ffmpeg gets stream from camera and pushes it to feed of ffserver:
  
  ffmpeg -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -y -vcodec libvpx http://127.0.0.1:8090/1.ffm

  ffserver stream options :

  <stream>
  
  Feed 1.ffm
  
  Format webm
  
  NoAudio
  
  #VideoCodec libvpx
  
  #VideoSize 480x320
  
  VideoFrameRate 24
  
  AVOptionVideo flags +global_header
  
  AVOptionVideo cpu-used 0
  
  AVOptionVideo qmin 1
  
  AVOptionVideo qmax 31
  
  AVOptionVideo quality good
  
  PreRoll 0
  
  StartSendOnKey
  
  VideoBitRate 128
  
  </stream>

  (note, videoSize option is commented). But even with default VideoSize (160x128), ffserver doesn’t respond for each request. Browser always gets :

  HTTP/1.0 200 OK
  
  Pragma: no-cache
  
  Content-Type: video/webm

  But sometimes video content is not sent.

  If I uncomment VideoSize option - the same problem but much less successfull requests comparing with default video size.

  ffserver log looks regular with no errors. But as you can see that sometimes it sends only headers to client :

  Thu Nov 27 12:49:11 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
  
  Thu Nov 27 12:49:25 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
  
  Thu Nov 27 12:49:36 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
  
  Thu Nov 27 12:50:52 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
  
  Thu Nov 27 12:53:54 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 459
  
  Thu Nov 27 13:30:19 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
  
  Thu Nov 27 13:30:34 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 385731
  
  Thu Nov 27 13:30:34 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 458752
  
  Thu Nov 27 13:30:36 2014 127.0.0.1 - - [GET] "/1.ffm HTTP/1.1" 200 4175
  
  Thu Nov 27 13:30:58 2014 127.0.0.1 - - [GET] "/1.webm HTTP/1.1" 200 493
  
  Thu Nov 27 13:30:58 2014 127.0.0.1 - - [POST] "/1.ffm HTTP/1.1" 200 622592

  Does anybody know what could it be ? Actually I need to save original VideoSize for stream. I am trying to override ffserver stream options with ffmpeg using the command (passing the same parameters as in ffserver’s stream) :

  ffmpeg -re -override_ffserver -rtsp_transport tcp -i rtsp://admin:admin@192.168.10.76:80 -an -r 24 -qmin 1 -qmax 31 -cpu-used 0 -quality good -flags:v +global_header -b:v 128 -vcodec libvpx -f webm -y http://127.0.0.1:8090/1.ffm

  But at the momment I still have error message ’Output file is empty, nothing was encoded’. Here is ffmpeg’s output :

  ffmpeg version 2.4.2 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Oct  6 2014 17:33:05 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --prefix=/opt/ffmpeg --libdir=/opt/ffmpeg/lib/ --enable-shared --enable-avresample --disable-stripping --enable-gpl --enable-version3 --enable-runtime-cpudetect --build-suffix=.ffmpeg --enable-postproc --enable-x11grab --enable-libcdio --enable-vaapi --enable-vdpau --enable-bzlib --enable-gnutls --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libfaac --enable-libvo-aacenc --enable-nonfree --enable-libmp3lame --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfdk_aac --enable-libopus --enable-pthreads --enable-zlib --enable-libvpx --enable-libfreetype --enable-libpulse --enable-debug=3
  
    libavutil      54.  7.100 / 54.  7.100
  
    libavcodec     56.  1.100 / 56.  1.100
  
    libavformat    56.  4.101 / 56.  4.101
  
    libavdevice    56.  0.100 / 56.  0.100
  
    libavfilter     5.  1.100 /  5.  1.100
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  0.100 /  3.  0.100
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  0.100 / 53.  0.100
  
  Guessed Channel Layout for  Input Stream #0.1 : mono
  
  Input #0, rtsp, from 'rtsp://admin:admin@192.168.10.76:80':
  
    Metadata:
  
      title           : RTSP Session/2.0
  
    Duration: N/A, start: 0.000000, bitrate: 128 kb/s
  
      Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709), 1280x720 [SAR 1:1 DAR 16:9], 25 fps, 100 tbr, 90k tbn, 50 tbc
  
      Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
  
  [swscaler @ 0x197f7a0] deprecated pixel format used, make sure you did set range correctly
  
  [libvpx @ 0x1a0c080] Bitrate 128 is extremely low, maybe you mean 128k
  
  [libvpx @ 0x1a0c080] v1.3.0
  
  The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s
  
  Output #0, webm, to 'http://127.0.0.1:8090/1.ffm':
  
    Metadata:
  
      title           : RTSP Session/2.0
  
      encoder         : Lavf56.4.101
  
      Stream #0:0: Video: vp8 (libvpx), yuv420p, 480x320 [SAR 32:27 DAR 16:9], q=1-31, 0 kb/s, 24 fps, 1k tbn, 24 tbc
  
      Metadata:
  
        encoder         : Lavc56.1.100 libvpx
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  frame=   33 fps= 22 q=0.0 size=       0kB time=00:00:00.00 bitrate=N/A dup=0 droframe=   43 fps= 22 q=0.0 Lsize=       0kB time=00:00:00.00 bitrate=N/A dup=0 drop=1    
  
  video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
  
  Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
  
  Received signal 2: terminating.

  Thanks in advance.

• ffmpeg Join MP4 videos, error (audio) of vine.co

  26 novembre 2014, par user3537531

  I’ve tried everything bucar after searching and I have not managed to unite vine.co videos to upload to youtube. Sometimes youtube error notifies Encryption, sometimes the video is cut audio to half, and in other cases the audio does not match the video.

  things I’ve tried

  ---

  file '/path/to/1.mp4'
  
  file '/path/to/2.mp4'
  
  ffmpeg -f concat -i list.txt -c copy result.mp4

  ---

  ffmpeg -i concat:"1.mp4|2.mp4" -codec copy result.mp4

  ---

  ffmpeg -i "concat:1.mp4|2.mp4" -c copy result.mp4

  The video is created, but always happen any errors or youtube audio conversion error tells me.

  We are talking of joining between 300 and 1,000 videos of between 3 and 6 seconds.

  From what I’ve read all would have to have the same frame rate and the same resolution.

  I have also proven to mp4box and post on youtube gives me trouble conversion

  mp4box-cat 1.mp4 -cat 2.mp4 -new result.mp4

  And not more I can do, I hope you can help me to attach a lot of videos on console. A greeting and thanks.