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• Stream camera/video from python to client using ffmpeg

  31 janvier 2018, par Rahul

  So I’m making an application where I have connected my camera with python code through opencv.

  Now I can process each frame inside my python code but I want to stream this live feed to users.

  I found that FFmpeg can be used to achieve this but I’m wondering how.
  Also please do suggest any solution to achieve this with any other library.

  The only condition is that I should be able to stream from my python code may be via a subprocess.

  thanks in advance :)

• MPEG-DASH Livestreaming using ffmpeg, avconv, MP4Box and Dashjs

  28 avril 2017, par Sushant Mongia

  I’m working on delivering Live Streaming with DASH capabilities. Long story short, it’s a very crude testbed setup so I might be off the mark in some respects. I’m also posting this as a simple setup for the community and people out there struggling to look up a Live Streaming with DASH Tutorial.

  Setup :
  OS : Ubuntu 16.04
  Encoding Tools :
  ffmpeg : To record a livestream using a desktop webcam in mpeg2 format
  avconv : To convert mpeg2 to mpeg4 file format
  MP4Box : To DASH it, i.e. produce the .mpd, some conf files, seg_init and the segments
  Dashjs : Reference Client 2.4.1
  Server : Apache

  Process :
  I’ve written 3 bash scripts, with basically an infinite while loop in them, with ffmpeg , avconv and the MP4Box commands, one in each of them. I first run the ffmpeg script that records a video using the desktop webcam and then I run the avconv script that kills the ffmpeg command and converts the file from mpeg2 format to mpeg 4 format. Now since the ffmpeg command is in an infinite while loop, it restarts. Then I run the MP4Box command that DASH-es the avconv commands’ output. Then everything is sent to the DASHjs client and pretty much the whole setup gets repeated every 5 seconds.

  Commands :

  ---

  ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video0 -f mpegts -codec:v mpeg1video -s 640x480 -b:v 1000k -bf 0 livestream  

  ---

  avconv -i livestream out.mp4  

  ---

  MP4Box command in a loop, MP4Box -dash-live 4000 -fps 24 -frag 6000 -profile dashavc264:live -dynamic -mpd-refresh 5000 -dash-ctx dashtest.txt -time-shift -1 -inter 0 -segment-name output-seg -bs-switching no out.mp4  

  Problem :
  The ffmpeg sends a chunk of 5 seconds (that’s due to the sleep command in my avconv bash script) and the MP4Box reads that 5 second chunk and loops that chunk. So when the next chunk comes in, newer segments are produced, but the player is still playing the older segments, typically just the very first few segments in a loop.

  Questions :
  1) Am I missing out on some core concept here ? Are the commands and their respective attributes with the right parameters and the right values ?
  2) I believe there should be a way to pipeline these processes in a better manner, should I be looking into writing a python script maybe ?

  Happy to provide more info ! Cheers

• Stream opus audio rtp to android device

  27 avril 2017, par Yuriy Aizenberg

  I want to stream audio (opus codec) with ffmpeg directly to android device.

  On PC i start stream :

    ./ffmpeg -re -stream_loop -1 -i akgld-c8mxm.opus -acodec libopus -ac 1 -ab 96k -vn -f rtp rtp://192.168.0.100:6000

  Where 192.168.0.100 - local wifi address of my phone.

  On Android device i tried to play stream :

   public void tryPlayStream() {
  
          String ip = Utils.wifiIpAddress(this);
  
          StrictMode.ThreadPolicy policy = new StrictMode.ThreadPolicy.Builder().permitNetwork().build();
  
          StrictMode.setThreadPolicy(policy);
  
          AudioManager audio = (AudioManager) getSystemService(AUDIO_SERVICE);
  
          audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
  
          audioGroup = new AudioGroup();
  
          audioGroup.setMode(AudioGroup.MODE_ECHO_SUPPRESSION);
  
          InetAddress inetAddress;
  
          try {
  
              inetAddress = InetAddress.getByName(ip);
  
              audioStream = new AudioStream(inetAddress);
  
              audioStream.setCodec(AudioCodec.PCMA);
  
              audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
  
              InetAddress inetAddressRemote = InetAddress.getByName(ip);
  
              audioStream.associate(inetAddressRemote, 6000);
  
              audioStream.join(audioGroup);
  
          } catch (Exception e) {
  
              e.printStackTrace();
  
          }
  
      }

  In logcat i see next lines :

  E/AudioRecord: AudioFlinger could not create record track, status: -1
  
   E/AudioGroup: cannot initialize audio device

  What im doing wrong ? Thanks for any help