Recherche avancée

Sur d’autres sites (7212)

• ffmpeg missing blend filter

  13 mars 2014, par user2927954

  I am working on android project using ffmpeg library. I followed instruction from : https://github.com/guardianproject/android-ffmpeg to build ffmpeg library for android. All are OK.

  Now, i am testing filters that corresponding to video filter of this library such as : drawtext, scale, pad, ... Most of them worked fine, but i have problem with "blend" filter.

  I try with commands : ffmpeg -i input1.mp4 -i input2.mp4 -filter_complex blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)' -y out.mp4 and recieve error : no such blend filter.

  Do you know how i configure to enable this filter ?

  Here is the output result :

  ffmpeg version 0.11.1 Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Feb 27 2014 05:23:20 with gcc 4.6 20120106 (prerelease)
  
    configuration: --arch=arm --cpu=cortex-a8 --target-os=linux --enable-runtime-cpudetect --prefix=/data/data/info.guardianproject.ffmpeg/app_opt --enable-pic --disable-shared --enable-static --cross-prefix=/home/admin/workspace/android-ndk-r9c/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86/bin/arm-linux-androideabi- --sysroot=/home/admin/workspace/android-ndk-r9c/platforms/android-3/arch-arm --extra-cflags='-I../x264 -mfloat-abi=softfp -mfpu=neon' --extra-ldflags=-L../x264 --enable-version3 --enable-gpl --disable-doc --enable-yasm --enable-decoders --enable-encoders --enable-muxers --enable-demuxers --enable-parsers --enable-protocols --enable-filters --enable-avresample --enable-libfreetype --disable-indevs --enable-indev=lavfi --disable-outdevs --enable-hwaccels --enable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-network --enable-libx264 --enable-zlib --enable-muxer=md5
  
    libavutil      51. 54.100 / 51. 54.100
  
    libavcodec     54. 23.100 / 54. 23.100
  
    libavformat    54.  6.100 / 54.  6.100
  
    libavdevice    54.  0.100 / 54.  0.100
  
    libavfilter     2. 77.100 /  2. 77.100
  
    libswscale      2.  1.100 /  2.  1.100
  
    libswresample   0. 15.100 /  0. 15.100
  
    libpostproc    52.  0.100 / 52.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/videokit/in.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 0
  
      compatible_brands: isom3gp4
  
      creation_time   : 2014-03-04 08:53:01
  
    Duration: 00:00:15.10, start: 0.000000, bitrate: 7055 kb/s
  
      Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1280x720, 6935 kb/s, SAR 65536:65536 DAR 16:9, 17.18 fps, 34.42 tbr, 90k tbn, 180k tbc
  
      Metadata:
  
        creation_time   : 2014-03-04 08:53:01
  
        handler_name    : VideoHandle
  
      Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 16000 Hz, stereo, s16, 128 kb/s
  
      Metadata:
  
        creation_time   : 2014-03-04 08:53:01
  
        handler_name    : SoundHandle
  
  Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '/storage/emulated/0/videokit/7.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2mp41
  
      creation_time   : 1970-01-01 00:00:00
  
      encoder         : Lavf53.12.0
  
      comment         : Courtesy of National Geographic.  Used by Permission.
  
    Duration: 00:00:04.20, start: 0.000000, bitrate: 1601 kb/s
  
      Stream #1:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 1472 kb/s, 10 fps, 10 tbr, 10 tbn, 10 tbc
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : VideoHandler
  
      Stream #1:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, s16, 128 kb/s
  
      Metadata:
  
        creation_time   : 1970-01-01 00:00:00
  
        handler_name    : SoundHandler
  
  No such filter: 'blend'
  
  Error configuring filters.

  Thanks

• ffprobe stream selection for encoding

  22 octobre 2014, par user3652819

  when I run ffmpeg, I can see "default" audio and video stream :

  Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 960x400 [SAR 1:1 DAR 12:5], 3859 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 47.95 tbc 
  
    (default)
  
  Metadata:
  
    creation_time   : 2013-05-03 22:50:47
  
    handler_name    : GPAC ISO Video Handler
  
  Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 92 kb/s 
  
    (default)
  
   Metadata:
  
    creation_time   : 1970-01-01 00:00:00
  
    handler_name    : SoundHandler
  
  Stream #0:2: Video: mjpeg, yuvj420p(pc), 675x1000 [SAR 72:72 DAR 27:40], 90k tbr, 90k tbn, 90k tbc

  As I understand, this stream selected by ffmpeg as input when encoding, if map option not set.

  How can I get "default" stream using ffprobe ?

  Sorry for English

• WebRTC books – a brief review

  1er janvier 2014, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.