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• How can I improve the up-time of my coffee pot live stream ?

  26 avril 2017, par tww0003

  Some Background on the Project :

  Like most software developers I depend on coffee to keep me running, and so do my coworkers. I had an old iPhone sitting around, so I decided to pay homage to the first webcam and live stream my office coffee pot.

  The stream has become popular within my company, so I want to make sure it will stay online with as little effort possible on my part. As of right now, it will occasionally go down, and I have to manually get it up and running again.

  My Setup :

  I have nginx set up on a digital ocean server (my nginx.conf is shown below), and downloaded an rtmp streaming app for my iPhone.

  The phone is set to stream to example.com/live/stream and then I use an ffmpeg command to take that stream, strip the audio (the live stream is public and I don’t want coworkers to feel like they have to be careful about what they say), and then make it accessible at rtmp://example.com/live/coffee and example.com/hls/coffee.m3u8.

  Since I’m not too familiar with ffmpeg, I had to google around and find the appropriate command to strip the coffee stream of the audio and I found this :
  

  ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee

  Essentially all I know about this command is that the input stream comes from, localhost/live/stream, it strips the audio with -an, and then it outputs to rtmp://localhost/live/coffee.

  I would assume that ffmpeg -i rtmp://localhost/live/stream -an rtmp://localhost/live/coffee would have the same effect, but the page I found the command on was dealing with ffmpeg, and nginx, so I figured the extra parameters were useful.

  What I’ve noticed with this command is that it will error out, taking the live stream down. I wrote a small bash script to rerun the command when it stops, but I don’t think this is the best solution.

  Here is the bash script :

  while true;
  
  do
  
          ffmpeg -i rtmp://localhost/live/stream -vcodec libx264 -vprofile baseline -acodec aac -strict -2 -f flv -an rtmp://localhost/live/coffee
  
          echo 'Something went wrong. Retrying...'
  
          sleep 1
  
  done

  I’m curious about 2 things :

  1. What is the best way to strip audio from an rtmp stream ?
  2. What is the proper configuration for nginx to ensure that my rtmp stream will stay up for as long as possible ?

  Since I have close to 0 experience with nginx, ffmpeg, and rtmp streaming any help, or tips would be appreciated.

  Here is my nginx.conf file :

  worker_processes  1;
  
  
  
  events {
  
      worker_connections  1024;
  
  }
  
  
  
  
  
  http {
  
      include       mime.types;
  
      default_type  application/octet-stream;
  
  
  
      sendfile        on;
  
  
  
      keepalive_timeout  65;
  
  
  
      server {
  
          listen       80;
  
          server_name  localhost;
  
  
  
          location / {
  
              root   html;
  
              index  index.html index.htm;
  
          }
  
  
  
          error_page   500 502 503 504  /50x.html;
  
          location = /50x.html {
  
              root   html;
  
          }
  
  
  
          location /stat {
  
                  rtmp_stat all;
  
                  rtmp_stat_stylesheet stat.xsl;
  
                  allow 127.0.0.1;
  
          }
  
          location /stat.xsl {
  
                  root html;
  
          }
  
          location /hls {
  
                  root /tmp;
  
                  add_header Cache-Control no-cache;
  
          }
  
          location /dash {
  
                  root /tmp;
  
                  add_header Cache-Control no-cache;
  
                  add_header Access-Control-Allow-Origin *;
  
          }
  
      }
  
  }
  
  
  
  rtmp {
  
  
  
      server {
  
  
  
          listen 1935;
  
          chunk_size 4000;
  
  
  
          application live {
  
              live on;
  
  
  
              hls on;
  
              hls_path /tmp/hls;
  
  
  
              dash on;
  
              dash_path /tmp/dash;
  
          }
  
      }
  
  }

  edit :
  I’m also running into this same issue : https://trac.ffmpeg.org/ticket/4401

• VOD Streaming While Live Recording RTMP

  7 février 2019, par Nick Dario

  I need to collect a live stream of audio and offer it as a stream playable from the start, while recording is in progress.

  

It is important the audio is processed from the beginning, and it must be processed within 1 second.

  I am using SRS (simple RTMP server) to handle the rtmp stream, the server allows me to record the stream to a file, but I have not been able to offer the file as a stream.



  What set of tools or methods using SRS, another audio software (possibly ffmpeg), or raw audio manipulation, can achieve this ?

• Android Rendering Live H.264 over RTSP

  4 juin 2013, par Lior Ohana

  I'm trying to decode (and render) live H.264 over RTSP in an Android app.
  Assuming, there are no network latency issues, the latency should not exceed several seconds.

  The first try was to use the MediaPlayer which was fine but the internal buffering of the infrastructure causes delays of 10-15 seconds.

  Right now the main dilemma is between using the new MediaCodec APIs or with FFMPeg.

  I know there are many tutorials/samples out there talking about FFMPeg but I didn't see any comparison.
  I think I understand most of the pros/cons for each but before spending ages on making one of them working I would like to be sure.