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• Revision a32a086d23 : Enable sse2 implmentation of 8x8 ADST/DCT This commit makes use of the butterfl

  20 juin 2013, par Jingning Han

  Changed Paths :
   Modify /vp9/common/vp9_rtcd_defs.sh

  
   Modify /vp9/encoder/x86/vp9_dct_sse2.c

  
  
  Enable sse2 implmentation of 8x8 ADST/DCT

  This commit makes use of the butterfly structure to enable the sse2
  version implementation of 8x8 ADST/DCT hybrid transform coding.

  The runtime of hybrid transform module goes down from 1170 cycles
  to 245 cycles. Overall speed-up around 1.5%.

  Change-Id : Ic808ffd21ece8a9d0410d8c0243d7b6c28ac3b3f

• Use fluent-ffmpeg to tell if a file is a video or audio

  11 août 2016, par afterglowlee

  I am using node-fluent-ffmpeg module in NodeJS. It is very good that fluent-ffmpeg provides functions to get the metadata of a video and audio file.

  https://github.com/schaermu/node-fluent-ffmpeg#reading-video-metadata

  I have tried on Mac OS to use the "resolution" attribute in the metadata to tell if a file is audio only or video, i.e. if both resolution.w and resolution.h are 0, then this file is an audio. This work fine on Mac OS. But some strange things happened that this doesn’t work on Windows platform (I have tried Windows 7 64bit and Windows 2008) using the latest ffmpeg. Even though I put a .mp3 file through fluent-ffmpeg,the result looks something like this :

  video:
  
  {
  
    container:'mp3',
  
    ...
  
    resolution: {w:300,h:300},
  
    resolutionSquare: {w:300,h:300},
  
    aspectString: '1:1',
  
    ...
  
  }
  
  audio:
  
  {
  
    codec:'mp3',
  
    bitrate:64,
  
    sample_rate:44100,
  
    stream:0,
  
    channels:1
  
  }

  I am not why there is a "resolution" since it is a pure audio file. So is there any solid way to find out if the file is audio only or video from the metadata ? Or should I use ffmpeg commandline to find it out ?

• Streaming live video with avconv/ffmped on nginx rtmp server player error

  6 juin 2014, par user3715403

  Good morning,
  My current project involves playing live video (testing using a usb webcam for now). I compiled and installed nginx server with the rtmp module (obtained from https://github.com/arut/nginx-rtmp-module). The nginx configuration file is as :

  worker_processes  1;
  
  
  
  error_log  logs/error.log debug;
  
  
  
  events {
  
      worker_connections  1024;
  
  }
  
  
  
  rtmp {
  
      server {
  
          listen 1935;
  
  
  
          application myapp {
  
              live on;
  
  
  
              #record keyframes;
  
              #record_path /tmp;
  
              #record_max_size 128K;
  
              #record_interval 30s;
  
              #record_suffix .this.is.flv;
  
  
  
              #on_publish http://localhost:8080/publish;
  
              #on_play http://localhost:8080/play;
  
              #on_record_done http://localhost:8080/record_done;
  
          }
  
      }
  
  }
  
  
  
  http {
  
      server {
  
          listen      8080;
  
  
  
          location /stat {
  
              rtmp_stat all;
  
              rtmp_stat_stylesheet stat.xsl;
  
          }
  
  
  
          location /stat.xsl {
  
              root /path/to/nginx-rtmp-module/;
  
          }
  
  
  
          location /control {
  
              rtmp_control all;
  
          }
  
  
  
          #location /publish {
  
          #    return 201;
  
          #}
  
  
  
          #location /play {
  
          #    return 202;
  
          #}
  
  
  
          #location /record_done {
  
          #    return 203;
  
          #}
  
  
  
          location /rtmp-publisher {
  
              root /path/to/nginx-rtmp-module/test;
  
          }
  
  
  
          location / {
  
              root /path/to/nginx-rtmp-module/test/www;
  
          }
  
      }
  
  }

  The command line to start streaming is as :

  avconv -s vga -f video4linux2 -i /dev/video0 -vcodec libx264 -an -f flv -b 800k -r 30 rtmp://localhost/myapp/live

  While streaming, the following stats are displayed :

  avconv version 10.1, Copyright (c) 2000-2014 the Libav developers
  
    built on May 13 2014 15:13:23 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  
  [video4linux2 @ 0x2f06220] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, video4linux2, from '/dev/video0':
  
    Duration: N/A, start: 112393.305806, bitrate: 73728 kb/s
  
      Stream #0.0: Video: rawvideo, yuyv422, 640x480, 73728 kb/s, 15 fps, 1000k tbn
  
  [libx264 @ 0x2f187a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 AVX
  
  [libx264 @ 0x2f187a0] profile High, level 3.0
  
  [libx264 @ 0x2f187a0] 264 - core 120 r2151 a3f4407 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=800 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  
  Output #0, flv, to 'rtmp://localhost/myapp/live':
  
    Metadata:
  
      encoder         : Lavf55.12.0
  
      Stream #0.0: Video: libx264, yuv420p, 640x480, q=-1--1, 800 kb/s, 1k tbn, 30 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo -> libx264)
  
  Press ctrl-c to stop encoding
  
  frame= 3283 fps= 30 q=26.0 size=   10592kB time=108.33 bitrate= 810.5kbits/s

  Before embedding the stream onto a web page using, say videojs, I am using avplay which returns the error : rtmp ://localhost/myapp/live : Input/output error while VLC returns the error
  Your input can’t be opened :
  VLC is unable to open the MRL ’rtmp ://localhost/myapp/live’. Check the log for details.

  Any suggestions about what would cause the errors displayed by VLC/avplay ?

  Thanks
  Daniel