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• ffmpeg and ffserver, rc buffer underflow ?

  25 février 2018, par Dove Devic

  I am attempting to write a simple streaming server for a project. I have an AWS Linux machine that will be running ffserver. Curently, as it stands, my config file looks like the following :

  #Server Configs
  
  HTTPPort 8090
  
  HTTPBindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 1000
  
  CustomLog -
  
  
  
  #Create a Status Page
  
  <stream>
  
  Format status
  
  ACL allow localhost
  
  ACL allow 255.255.255.255 #Allow everyone to view status, for now
  
  </stream>
  
  
  
  #Creates feed, only allow from self
  
  <feed>
  
  File /tmp/feed1.ffm
  
  FileMaxSize 50M
  
  ACL allow 127.0.0.1
  
  ACL allow 
  
  </feed>
  
  
  
  #Creates stream, allow everyone
  
  <stream>
  
  Format mpeg
  
  Feed feed1.ffm
  
  VideoFrameRate 30
  
  VideoSize 640x480
  
  AudioSampleRate 44100
  
  </stream>

  I then am capturing my Webcam and sending it up to the server using the following command :

  ffmpeg -f dshow 
  
         -i video="Webcam C170":audio="Microphone (Webcam C170)" 
  
         -b:v 1400k  
  
         -maxrate 2400k  
  
         -bufsize 1200k  
  
         -ab 64k  
  
         -s 640x480  
  
         -ac 1  
  
         -ar 44100  
  
         -y http://:8090/feed1.ffm

  When I run this however, I get the following output from my console :

  Guessed Channel Layout for  Input Stream #0.1 : stereo
  
  Input #0, dshow, from 'video=Webcam C170:audio=Microphone (Webcam C170)':
  
    Duration: N/A, start: 12547.408000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 30 tbr, 10000k tbn, 30 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, ffm, to ':8090/feed1.ffm':
  
    Metadata:
  
      creation_time   : 2017-04-26 14:55:27
  
      encoder         : Lavf57.25.100
  
      Stream #0:0: Audio: mp2, 44100 Hz, mono, s16, 64 kb/s
  
      Metadata:
  
        encoder         : Lavc57.24.102 mp2
  
      Stream #0:1: Video: mpeg1video, yuv420p, 640x480, q=2-31, 64 kb/s, 30 fps, 1000k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc57.24.102 mpeg1video
  
      Side data:
  
        unknown side data type 10 (24 bytes)
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (pcm_s16le (native) -> mp2 (native))
  
    Stream #0:0 -> #0:1 (rawvideo (native) -> mpeg1video (native))
  
  Press [q] to stop, [?] for help
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:01.13 bitrate= 404.8kbits/s dup=13 drop=0 speed=2.22x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:01.63 bitrate= 361.1kbits/s dup=13 drop=0 speed=1.61x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:02.13 bitrate= 368.6kbits/s dup=13 drop=0 speed= 1.4x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:02.66 bitrate= 344.1kbits/s dup=13 drop=0 speed=1.32x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:03.16 bitrate= 331.1kbits/s dup=13 drop=0 speed=1.25x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  frame=  117 fps= 36 q=31.0 Lsize=     156kB time=00:00:03.86 bitrate= 330.5kbits/s dup=13 drop=0 speed= 1.2x
  
  video:118kB audio:27kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 7.659440%
  
  Exiting normally, received signal 2.

  And on my viewer, I just get a black screen.

  Is there something I’m missing ? Searching lead to nothing on "increasing qmax" or anything similar to what ffmpeg complained about. There have been questions asked here, but nothing has been done/answered.

  Thanks in advance

• ffmpeg equivalent for sox -t ima

  14 décembre 2022, par LucasM

  I am trying to use ffmpeg to combine 1 audio file (ADPCM) and 1 video file (h264) into single mp4. Video by file conversion works fine but ffmpeg chokes on guessing audio input. I can't figure out how to tell ffmpeg which params to use to decode raw audio file.

  


  Currently I first run sox to convert raw audio to wav :

  


  sox -t ima -r 8000 audio.raw audio.wav


  


  ... then feed audio.wav from sox as ffmpeg input

  


  ffmpeg -i video.raw -i audio.wav movie.mp4


  


  I am trying to avoid sox step and use audio.raw in ffmpeg.

  


  Thank you

  


• Setting a timeout for av_read_frame

  20 décembre 2014, par user3663917

  I am new to FFMPEG and was trying to do HLS streaming using FFMPEG. When i tried using the function "av_read_frame" it returns a negative value whenever data is not available. Is there some method to make this function wait till some data is received or to make this function wait till a timeout is reached ?