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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

• Setting start date in gource

  5 juin 2014, par rfc1484

  I’m trying to run gource from a certain date, but it reports me a ’codec not found error’.

  This is the command without a start date :

  gource -1280x720 --seconds-per-day 1 --stop-at-end --hide filenames --hide files --git-branch master --camera-mode track --output-ppm-stream - | ffmpeg -y -b 3000k -r 25 -f image2pipe -vcodec ppm -i - test.mp4

  Which works fine and generates the .mp4 file as expected.

  This is the command with a start date :

  gource -1280x720 --start-date '2014-01-01' --seconds-per-day 1 --stop-at-end --hide filenames --hide files --git-branch master --camera-mode track --output-ppm-stream - | ffmpeg -y -b 3000k -r 25 -f image2pipe -vcodec ppm -i - test.mp4

  Which doesn’t work and generates the following error message :

  [image2pipe @ 0x78a600] Could not find codec parameters (Video : ppm)

  Any suggestions on how to fix this ?

• 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);