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• compile the ffmpeg3.3 for Mac OS show ERROR : libmp3lame >= 3.98.3 not found

  4 mars 2018, par MR.PJ

  I use my mac to compile the ffmpeg3.3 for Mac OS according to https://trac.ffmpeg.org/wiki/CompilationGuide/macOS.I tried the first(which through Homebrew) and third(which builds it yourself) method,the first method is ok.

  But the third, compiling FFmpeg myself and Installing dependencies with Homebrew.I don’t know the meaning of this sentence enter image description here

  after this step :

  ./configure  --prefix=/usr/local --enable-gpl --enable-nonfree --enable-libass \
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame \
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libopus --enable-libxvid \
--samples=fate-suite/

  terminal show :

  RROR: libmp3lame >= 3.98.3 not found

  when I use locate libmp3lame.a,it is at ’/usr/local/Cellar/lame/3.99.5/lib/libmp3lame.a’.

  I also try --enable-libmp3lame --extra-ldflags=-L/usr/local/lib to solve,but it is no effect.

  How to solve thisERROR: libmp3lame >= 3.98.3 not found

• Streaming UDP packets to two different ports (for video and audio). Video works fine, but the audio does not show

  8 avril 2018, par Winston Chen

  I am taking a rtsp stream, split the video and audio out, and stream them to two different ports respectively using gstreamer so that my ffserver would be able to display the stream on my browser.

  My gstreamer pipeline :

  gst-launch-1.0 -v rtspsrc location=rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov latency=300 timeout=0 drop-on-latency=true rtp-blocksize=4096 name=rtsp_source ! \
  
    queue ! capsfilter caps="application/x-rtp,media=video" ! rtph264depay ! h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=127.0.0.1 port=9527 rtsp_source. ! \
  
    queue ! rtpmp4apay pt=97 ! udpsink host=127.0.0.1 port=9327

  Here comes the sdp and my ffmpeg commnad :

  m=video 9527 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  c=IN IP4 127.0.0.1
  
  
  
  m=audio 9327 RTP/AVP 97
  
  a=rtpmap:97 mpeg4-generic/48000/6
  
  c=IN IP4 127.0.0.1

  ffmpeg -protocol_whitelist "file,tcp,rtp,udp" -i ~/test.sdp -max_muxing_queue_size 1024 http://localhost:8090/feed1.ffm

  And finally, this is my ffserver config (the important part) :

  <feed>               # This is the input feed where FFmpeg will send
  
     File ./feed1.ffm            # video stream.
  
     FileMaxSize 1GB             # Maximum file size for buffering video
  
     ACL allow 127.0.0.1         # Allowed IPs
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed feed1.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # NoDefaults
  
     # NoAudio
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 240x160           # Video resolution
  
     VideoFrameRate 10           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
  
  
     PreRoll 0
  
     StartSendOnKey
  
     VideoGopSize 12
  
     VideoBitRate 256
  
  </stream>

  The thing is that if I take away the audio part and apply NoAudio, the video streams fine. However, I could not get the audio to work. Am I doing anything wrong ?

• fftools/ffmpeg : change fps progress log message to show two decimal digits

  28 avril 2018, par André Camargo

  fftools/ffmpeg : change fps progress log message to show two decimal digits
  Useful when transcoding videos at 29.97 fps because delivers a more accurate result for monitoring.
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] fftools/ffmpeg.c