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• Reduce HLS latency from +30 seconds

  4 juin 2014, par Rick

  Ubuntu 12.04

  nginx 1.2.4

  avconv -version

  avconv version 0.8.10-4:0.8.10-0ubuntu0.12.04.1, Copyright (c) 2000-2013 the Libav developers
  
    built on Feb  6 2014 20:56:59 with gcc 4.6.3
  
  avconv 0.8.10-4:0.8.10-0ubuntu0.12.04.1
  
  libavutil    51. 22. 2 / 51. 22. 2
  
  libavcodec   53. 35. 0 / 53. 35. 0
  
  libavformat  53. 21. 1 / 53. 21. 1
  
  libavdevice  53.  2. 0 / 53.  2. 0
  
  libavfilter   2. 15. 0 /  2. 15. 0
  
  libswscale    2.  1. 0 /  2.  1. 0
  
  libpostproc  52.  0. 0 / 52.  0. 0

  I’m using avconv and nginx to create an HLS stream but right now my latency is regularly well over 30 seconds. After much reading I am aware that HLS has built in latency and that 10s is expected and even preferred but 30s seems quite extreme.

  I’ve seen a lot of discussion on the nginx-rtmp google group, this thread in particular had a lot of suggestions. I have attempted to reduce solve my issue by reducing the hls_fragment and the hls_playlist_length but they haven’t had a significant effect.

  nginx.conf :

  #user  nobody;
  
  worker_processes  1;
  
  
  
  error_log  logs/error.log debug;
  
  
  
  events {
  
      worker_connections  1024;
  
  }
  
  
  
  http {
  
      include       mime.types;
  
      default_type  application/octet-stream;
  
  
  
      sendfile        on;
  
      keepalive_timeout  65;
  
  
  
      server {
  
          listen       8888;
  
          server_name  localhost;
  
  
  
          add_header 'Access-Control-Allow-Origin' "*";
  
  
  
          location /hls {
  
              types {
  
                  application/vnd.apple.mpegurl m3u8;
  
                  video/mp2t ts;
  
              }
  
              root /tmp;
  
          }
  
  
  
          # rtmp stat
  
          location /stat {
  
              rtmp_stat all;
  
              rtmp_stat_stylesheet stat.xsl;
  
          }
  
          location /stat.xsl {
  
              # you can move stat.xsl to a different location
  
              root /usr/build/nginx-rtmp-module;
  
          }
  
  
  
          # rtmp control
  
          location /control {
  
              rtmp_control all;
  
          }
  
  
  
          error_page   500 502 503 504  /50x.html;
  
          location = /50x.html {
  
              root   html;
  
          }
  
      }
  
  }
  
  
  
  rtmp {
  
      server {
  
          listen 1935;
  
          ping 30s;
  
          notify_method get;
  
  
  
          application myapp {
  
              live on;
  
  
  
              hls on;
  
              hls_path /tmp/hls;
  
              hls_base_url http://x.x.x.x:8888/hls/;
  
              hls_sync 2ms;
  
              hls_fragment 2s;
  
  
  
              #hls_variant _low BANDWIDTH=160000;
  
              #hls_variant _mid BANDWIDTH=320000;
  
              #hls_variant _hi BANDWIDTH=640000;
  
          }
  
      }
  
  }

  avconv command :

  avconv -r 30 -y -f image2pipe -codec:v mjpeg -i - -f flv -codec:v libx264 -profile:v baseline -preset ultrafast -tune zerolatency -an -f flv rtmp://127.0.0.1:1935/myapp/mystream

• Error in converting audio file format from ogg to wav [on hold]

  9 juin 2014, par Sumit Bisht

  I am trying to convert an ogg format file that was created using webrtc (html5 usermedia content generated on firefox) and transferred and decoded on the server into a wav file through ffmpeg but am getting this error on cmmand line while trying to convert :

  $ ffmpeg -i 2014-6-5_16-17-54.ogg res1.wav
  
  ffmpeg version 2.0.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on May  1 2014 13:12:12 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-4)
  
    configuration: --enable-gpl --enable-version3 --enable-shared --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
  
    libavutil      52. 38.100 / 52. 38.100
  
    libavcodec     55. 18.102 / 55. 18.102
  
    libavformat    55. 12.100 / 55. 12.100
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 79.101 /  3. 79.101
  
    libswscale      2.  3.100 /  2.  3.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Guessed Channel Layout for  Input Stream #0.0 : mono
  
  Input #0, ogg, from '2014-6-5_16-17-54.ogg':
  
    Duration: 00:00:01.84, start: 0.000000, bitrate: 18 kb/s
  
      Stream #0:0: Audio: opus, 48000 Hz, mono
  
      Metadata:
  
        ENCODER         : Mozilla29.0.1
  
  [graph 0 input from stream 0:0 @ 0x18dca20] Invalid sample format (null)
  
  Error opening filters!

  Although, I am able to play the file on server and using the same command, am able to convert .ogg files generated somewhere else. What might be I missing ?

  Edit :
  Here’s the source code that is used to write to the file :

  1) During startup - use the methods of getUserMedia API.

  navigator.getUserMedia({
  
          audio: true,
  
          video: false
  
      }, function(stream) {
  
          audioStream = RecordRTC(stream, {
  
              bufferSize: 16384
  
          });
  
          audioStream.startRecording();

  2) During stopping of the recording - extracting the recorded information.

  function(audioDataURL) {
  
       var audioFile = {};
  
       audioFile = {
  
         contents: audioDataURL
  
       **strong text**};

  On server end, the following code is creating a file from this data :

  dataURL = dataURL.split(',').pop(); // dataURL is the audioDataURL as defined above
  
  fileBuffer = new Buffer(dataURL, 'base64');
  
  fs.writeFileSync(filePath, fileBuffer);

• How to scale and position watermark to scale ?

  26 avril 2013, par Cobra_Fast

  I'm scaling a video and applying a watermark like so :

  ffmpeg -ss 0:0:0.000 -i video.mp4 -y -an -t 0:0:10.000
  
   -vf \"[in]scale=400:316[middle]\" -b:v 2000k -r 20
  
   -vf 'movie=watermark.png,pad=400:316:0:0:0x00000000 [watermark];[middle] [watermark]overlay=0:0[out]'
  
   out.flv

  However, the applied watermark seems to be scaled to the original video size rather than the smaller scaled video size.

  This command line worked on ffmpeg version 0.8.6.git and now behaves differently after an upgrade to version N-52381-g2288c77.

  How do I get it to work again ?

  ---

  Update 2013-04-26 :
  I now have tried to use the overlay filter's X and Y parameters instead of padding without success.