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• Video error while processing

  19 décembre 2012, par joelmaranhao

  Trying out paperclip-ffmpeg gem with Rails 3.1

  In my model

  class Video < ActiveRecord::Base
  
    belongs_to :creative_service
  
  
  
    has_attached_file :asset, :styles => { 
  
      :medium => { :geometry => "640x480", :format => 'flv' },
  
      :thumb => { :geometry => "100x100#", :format => 'jpg', :time => 10 }
  
    }, :processors => [:ffmpeg]
  
  end

  The error message when uploading a MPG file :

  Video error while processing video for MOV0652220121219-19138-1ybt1gp: Command 'ffmpeg -i :source -y -s 640x480 :dest' returned 1. Expected 0

  I am a total newbie with video processing. Any hint is welcome.

  j.

• 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

• Need a workaround for error in xuggler todo with vorbis

  25 septembre 2012, par Chris

  I have been at this for days, I am using xuggler, the only version i can use is 3.4 because upto 4.4 writing to outputstream directly causes jvm crash.

  I need to encode audio using vorbis, however, when i run code which works with any other audio codec I get

  2012-09-25 13:54:16,578 [Thread-4] ERROR org.ffmpeg - [libvorbis @ 0308a010] libvorbis : buffer overflow.
  2012-09-25 13:54:16,578 [Thread-4] WARN com.xuggle.xuggler - error : avcodec_encode_audio failed (../../../../../../../csrc/com/xuggle/xuggler/StreamCoder.cpp:1586)

  I think a workaround for found for this is in later versions of StreamCoder.cpp

  if (mCodecContext->codec->id == CODEC_ID_FLAC
  
            || mCodecContext->codec->id == CODEC_ID_VORBIS)
  
    {
  
      // FLAC & VORBIS audio for some reason gives an error if your output buffer isn't
  
      // over double the frame size, so we fake it here.  This could be further optimized
  
      // to only require an exact number, but this math is simpler and will always
  
      // be large enough.
  
      bufferSize = (64 + getAudioFrameSize() * (bytesPerSample + 1)) * 2;
  
    }

  But if I use the following code to make the IBuffer,

      int bufferSize = aCoder.getAudioFrameSize() * (aCoder.getBitRate());
  
      final byte[] audioBuf = new byte[bufferSize];
  
      //read audio bytes into audiobuf....
  
  
  
      buf = IBuffer.make(null, audioBuf, 0, (64 + aCoder.getAudioFrameSize() *    (aCoder.getBitRate() + 1)) * 2);
  
      System.out.println("afterbufmake");
  
      smp = IAudioSamples.make(buf,1,IAudioSamples.Format.FMT_S16);
  
              if (smp == null) {
  
                    System.out.println("smp == null");
  
                 // return;
  
              }

  smp always ends up null.

  Can anyone think of what might be happening with buffer overflow for just Vorbis and a workaround as I have only 3.4 version of Xuggler that works for what I need which is writing to outputstream.

  Thank you.