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• Receive multiple RTP, mix, output RTSP stream

  30 octobre 2017, par John

  I’m currently trying to receive multiple RTP audio stream, mixing them, and output RTSP stream by using ffmpeg or ffserver.

  RTP audio stream is send by Android AudioStream.

  Here is code Android side.

  AudioManager audio = (AudioManager) getSystemService(Context.AUDIO_SERVICE);
  
  audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
  
  
  
  audioGroup = new AudioGroup();
  
  audioGroup.setMode(AudioGroup.MODE_ECHO_SUPPRESSION);
  
  
  
  audioStream = new AudioStream(InetAddress.getByAddress(getLocalIPAddress()));
  
  audioStream.setCodec(AudioCodec.PCMU);
  
  audioStream.setMode(RtpStream.MODE_NORMAL);
  
  audioStream.associate(InetAddress.getByName(SipStackAndroid.getRemoteIp()), REMOTE_PORT);
  
  audioStream.join(audioGroup);

  Then I prepare server side.

  Here is ffserver.conf

  HTTPPort 5555
  
  HTTPBindAddress 0.0.0.0
  
  RTSPPort 5454
  
  RTSPBindAddress 0.0.0.0
  
  MaxHTTPConnections 100
  
  MaxClients 1000
  
  MaxBandwidth 10000
  
  CustomLog -
  
  
  
  <feed>
  
    File /tmp/feed1.ffm
  
    FileMaxSize 500M
  
  </feed>
  
  
  
  <stream>
  
    Format rtp
  
    Feed feed1.ffm
  
    AudioBitRate 13
  
    AudioChannels 1
  
    AudioSampleRate 8000
  
    AudioCodec pcm_mulaw
  
    # AudioCodec libmp3lame
  
    NoVideo
  
  </stream>

  And here is ffserver or ffmpeg command.

  ffserver -d -f ffserver.conf
  
  ffmpeg -i "rtp://192.168.150.10:12345" -acodec auto http://127.0.0.1:5555/feed1.ffm

  I can’t solve how to receive multiple rtp stream and how to mix them.

  Please give some ideas and actual links where I can find an answer.

• FFMPEG to fluent ffmpeg [on hold]

  16 janvier 2018, par Minh

  I use ffmpeg command on VPS but I want cover to github . Help me
  This is code

  ffmpeg - re - i http: //storage.livefb.com.vn/assets/video/video_307_1516075122469280.mp4 -i assets/bg/13.png -i assets/logo.png -filter_complex "[0:v][1:v] overlay=x=4:y=H-h:enable='between(t,0,t)' [tmp]; [tmp][2:v] overlay=x=14:y=H-h-10:enable='between(t,0,t)',drawtext=fontfile=assets/thanh.ttf:y=h-60:x=158:text='TDL AUTOFB INBOXFB LIVEFB
  
    0965367062 MINH THU ':fontcolor=white:fontsize=36,drawbox=enable='
  
  between(t, 2, 10)
  
  ':y=ih-ih/2:color=black@0.8:width=iw:height=50:t=max,drawtext=enable='
  
  between(t, 2, 10)
  
  ':fontfile=assets/thanh.ttf:y=h-h/2+10:x=(w-text_w)/2:text='
  
  được phát bởi livefb liên hệ minh thư để được hỗ trợ ':fontcolor=white:fontsize=30" -vcodec libx264 -pix_fmt yuv420p -r 30 -g 60 -b:v 1400k -profile:v main -level 3.1 -acodec libmp3lame -b:a 128k -ar 44100 -metadata title="" -metadata artist="" -metadata album_artist="" -metadata album="" -metadata date="" -metadata track="" -metadata genre="" -metadata publisher="" -metadata encoded_by="" -metadata copyright="" -metadata composer="" -metadata performer="" -metadata TIT1="" -metadata TIT3="" -metadata disc="" -metadata TKEY="" -metadata TBPM="" -metadata language="eng" -metadata encoder="" -threads 0 -preset superfast -f flv "rtmp://live-api.facebook.com:80/rtmp/187884241800146?ds=1&a=ATgAYCPa1TJeWa2t"

• rtsp stream transcode with ffserver/ffmpeg

  30 juin 2018, par Torbjørn

  I’m having a unifi dome webcam. Connected thrue a USG Pro4. I’m sending a rtsp live stream at 1024x576px. Until now I have been using ffserver to transcode the stream into a new mpjpeg stream that I can show in a html file.
  But after I updated ffserver to the newest version. This does not work.

  Using debian.

  ffserver.conf file :

      HTTPPort 8090
  
      # bind to all IPs aliased or not
  
      HTTPBindAddress 0.0.0.0
  
      # max number of simultaneous clients
  
      MaxClients 1000
  
      # max bandwidth per-client (kb/s)
  
      MaxBandwidth 100000
  
      # Suppress that if you want to launch ffserver as a daemon.
  
      # NoDaemon
  
      NoDefaults
      File /tmp/feed1.ffm
  
      FileMaxSize 500M
      Feed feed1.ffm
  
      Format mpjpeg
  
      VideoCodec mjpeg
  
      VideoFrameRate 1
  
      #VideoBufferSize 80000
  
      VideoQMin 1
  
      VideoQMax 5
  
      VideoSize 1024x576
  
      PreRoll 0
  
      Noaudio
  

  ffmpeg startup :

  `ffmpeg -rtsp_transport tcp -re -i "rtsp ://user:pass@193.213.XXX.XXX:554/s2" -b:v 1016k http://localhost:8090/feed1.ffm`

  I’ve tried with different startups after some google search. But I can get 1-10 fps before it shuts down.
  Sometimes I get :

  `rtsp ://user:pass@193.213.XXX.XXX:554/s2 : Invalid data found when processing input`

      Input #0, rtsp, from ’rtsp ://user:pass@193.213.XXX.XXX:554/s2’ :
  
        Metadata :
  
          title : s2
  
        Duration : N/A, start : 0.000000, bitrate : N/A
  
          Stream #0:0 : Audio : aac (LC), 11025 Hz, mono, fltp
  
          Stream #0:1 : Video : h264 (Main), yuvj420p(pc, bt709, progressive), 1024x576 [SAR 189:190 DAR 168:95], 1 fps, 1 tbr, 90k tbn, 2 tbc
  
      Output #0, ffm, to ’http://localhost:8090/feed1.ffm’ :
  
        Metadata :
  
          title : s2
  
          creation_time : now
  
          encoder : Lavf57.56.101
  
          Stream #0:0 : Video : mjpeg, yuvj420p(pc), 1024x576 [SAR 189:190 DAR 168:95], q=1-5, 1016 kb/s, 1 fps, 1000k tbn, 1 tbc
  
          Metadata :
  
            encoder : Lavc57.64.101 mjpeg
  
          Side data :
  
            cpb : bitrate max/min/avg : 0/0/1016000 buffer size : 0 vbv_delay : -1
  
      Stream mapping :
  
        Stream #0:1 -> #0:0 (h264 (native) -> mjpeg (native))
  
      Press [q] to stop, [?] for help
  
      frame=    4 fps=0.7 q=2.1 Lsize=     564kB time=00:00:03.00 bitrate=1540.1kbits/s dup=1 drop=0 speed=0.54x
  
      video:558kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead : 1.089245%
  

  Is there an easy way to get this working, or is there a much better way of doing it ?
  What i want out of this is an live web cam stream in an html site that works on all platforms without any plugins on clients end.