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• Raw numpy array from real-time network audio stream in Python

  30 mai 2015, par And

  I would like to get raw data (optimally as a numpy array) from a PCM network audio stream for which I have an URL. The goal is to analyze the signal from the network stream in real-time.

  There seem to be a lot of Python audio modules or wrappers around e.g. FFMPEG, but after quite an extensive search I am yet to find a single example of a complete pipeline.

  For those familiar with OpenCV, I am looking for an audio counterpart of the OpenCV VideoCampture class.

  Any suggestions of modules to look at or code code snippets welcome !

• ffmpeg live rtmp stream does not start to process for long time

  10 août 2013, par user1492502

  I have rtmp stream created by flash player in h264 but when i convert it to video or tumbnail using ffmpeg it some times works after very very long time and some time not work but if I create a stream with Flash Media live encoder on same FMS server the command below works fine. At the same time if I try the stream in player it works well and fine.

  I am using IP so DNS resolving issue is not possible either I think.

  ffmpeg -i rtmp ://xxx.xxx.xx.xx/live/bdeef2c065509361e78fa8cac90aac741cc5ee29 -r 1 -an -updatefirst 1 -y thumbnail.jpg

  Following is when it worked aftre 15 - 20 minutes 
  
  
  
  ffmpeg -i "rtmp://xxx.xxx.xx.xx/live/bdeef2c065509361e78fa8cac90aac741cc5ee29 live=1" -r 1 -an -updatefirst 1 -y thumb.jpg
  
  [root@test ~]# ffmpeg -i rtmp://38.125.41.20/live/bdeef2c065509361e78fa8cac90aac741cc5ee29 -r 1 -an -updatefirst 1 -y thumbnail.jpg
  
  ffmpeg version N-49953-g7d0e3b1-syslint Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Feb 14 2013 15:29:40 with gcc 4.4.6 (GCC) 20120305 (Red Hat 4.4.6-4)
  
    configuration: --prefix=/usr/local/cpffmpeg --enable-shared --enable-nonfree --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-decoder=liba52 --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --extra-cflags=-I/usr/local/cpffmpeg/include/ --extra-ldflags=-L/usr/local/cpffmpeg/lib --enable-version3 --extra-version=syslint
  
    libavutil      52. 17.101 / 52. 17.101
  
    libavcodec     54. 91.103 / 54. 91.103
  
    libavformat    54. 63.100 / 54. 63.100
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 37.101 /  3. 37.101
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  [flv @ 0x14c0100] Stream #1: not enough frames to estimate rate; consider increasing probesize
  
  [flv @ 0x14c0100] Could not find codec parameters for stream 1 (Audio: none, 0 channels): unspecified sample format
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [flv @ 0x14c0100] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, flv, from 'rtmp://xxx.xxx.xx.xx/bdeef2c065509361e78fa8cac90aac741cc5ee29':
  
    Metadata:
  
      keyFrameInterval: 15
  
      quality         : 90
  
      level           : 3.1
  
      bandwith        : 0
  
      codec           : H264Avc
  
      fps             : 15
  
      profile         : baseline
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Baseline), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 15 tbr, 1k tbn, 30 tbc
  
      Stream #0:1: Audio: none, 0 channels
  
  Output #0, image2, to 'thumbnail.jpg':
  
    Metadata:
  
      keyFrameInterval: 15
  
      quality         : 90
  
      level           : 3.1
  
      bandwith        : 0
  
      codec           : H264Avc
  
      fps             : 15
  
      profile         : baseline
  
      encoder         : Lavf54.63.100
  
      Stream #0:0: Video: mjpeg, yuvj420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 1 tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 -> mjpeg)
  
  Press [q] to stop, [?] for help
  
  frame= 2723 fps=1.3 q=1.6 size=N/A time=00:45:23.00 bitrate=N/A dup=8 drop=12044

  and on stopping the stream by closing the browser running the flash player which is publishing the video I get the following

  [flv @ 0x23684e0] Could not find codec parameters for stream 1 (Audio: none, 0 channels): unspecified sample format
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  [flv @ 0x23684e0] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, flv, from 'rtmp://xxx.xxx.xx.xx/live/bdeef2c065509361e78fa8cac90aac741cc5ee29':
  
    Metadata:
  
      keyFrameInterval: 15
  
      quality         : 90
  
      bandwith        : 0
  
      level           : 3.1
  
      codec           : H264Avc
  
      fps             : 15
  
      profile         : baseline
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Baseline), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 15 tbr, 1k tbn, 30 tbc
  
      Stream #0:1: Audio: none, 0 channels

  when if i stop the stream it quickly creates a thumbnail file where as running stream is an issue.

• ffmpeg issue with mp4 play and time convert for webm [on hold]

  19 mai 2015, par ali rah

  I use this code to convert mp4 to mp4 (with watermark) and webm and ogv

  and I view it via a standard html5 code.

  exec("$ffmpegPath  -i $inputPath -i $watermark -filter_complex overlay=10:10 $outPath");

  But I found 2 important problem.

  first : it generate mp4 but when open it on a browser it just play voice (no video with a black screen)

  note : I can play orginall mp4 in browsers.

  Second : I saw it can convert mp4 and ovg in a few second but for webm it takes about 3 min !!!...

  please advice me to solve my problems...