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• Using FFMPEG to split a 16 channel audio input source into 4 seperate 4 channel audio feeds for streaming

  30 décembre 2019, par Mathew Knight

  I hope someone can help

  I am currently trying to split a 16ch Dante audio feed from a separate machine into 4 different audio streams that I can use to then TX via RTMP to Wowza for MPEG-DASH encoding, at present i am just trying to split them into files, I will add the RTMP streaming later.

  The biggest issue I am encountering at current is that FFMPEG is returning me this error from my input string

  Filter channelsplit:WR has an unconnected output

  here is my current input string

  ffmpeg -f dshow -i audio="Dante Via Receive (Dante Via)" -filter_complex "[0:a]channelsplit=channel_layout=hexadecagonal[FL][FR][FC][BL][BR][BC][SL][SR][TFL][TFC][TFR][TBL][TBC][TBR][WL][WR]" -map "[FL][FR][FC][BL]" 1-4.wav -map "[BR][BC][SL][SR]" 5-8.wav -map "[TFL][TFC][TFR][TBL]" 9-12.wav -map "[TBC][TBR][WL][WR]" 13-16.wav

  and here is the full FFMPEG output

  ffmpeg version git-2019-12-26-b0d0d7e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 9.2.1 (GCC) 20191125
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  
    libavutil      56. 37.100 / 56. 37.100
  
    libavcodec     58. 65.100 / 58. 65.100
  
    libavformat    58. 35.101 / 58. 35.101
  
    libavdevice    58.  9.101 / 58.  9.101
  
    libavfilter     7. 69.101 /  7. 69.101
  
    libswscale      5.  6.100 /  5.  6.100
  
    libswresample   3.  6.100 /  3.  6.100
  
    libpostproc    55.  6.100 / 55.  6.100
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, dshow, from 'audio=Dante Via Receive (Dante Via)':
  
    Duration: N/A, start: 103082.790000, bitrate: 1411 kb/s
  
      Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  File '1-4.wav' already exists. Overwrite? [y/N] y
  
  File '5-8.wav' already exists. Overwrite? [y/N] y
  
  File '9-12.wav' already exists. Overwrite? [y/N] y
  
  File '13-16.wav' already exists. Overwrite? [y/N] y
  
  Filter channelsplit:WR has an unconnected output 

  I’m also getting the issue where FFMPEG is guessing that the channel count is stereo, which is incorrect but i’m having problems figuring out how to define the input stream as 16ch’s of audio

  Any help with this would be greatly recieved

  Cheers

  M

• FFMPEG detect silence command runs correctly but doses not give the silence duration

  7 janvier 2020, par Aizayousaf

  I have a .wav audio file and I need to extract silence/pause duration in this file. I’m using ffmpeg with silence detect filter but I’m unable to understand why its not giving silence duration with this file while it gives result with other files. Can anyone help me to understand the out given below that why its not showing detected silences.

  Input Command :

  ffmpeg -i "input.wav" -af silencedetect=noise=-30dB:d=0.5 -f null -

  OutPut

  ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  
  built with gcc 9.1.1 (GCC) 20190807
  
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --    enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-
  
  libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable- 
  
  libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-
  
  libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --
  
  enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --
  
  enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid -- 
  
   enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --    enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
  
  
  libavutil      56. 31.100 / 56. 31.100
  
  libavcodec     58. 54.100 / 58. 54.100
  
  libavformat    58. 29.100 / 58. 29.100
  
  libavdevice    58.  8.100 / 58.  8.100
  
  libavfilter     7. 57.100 /  7. 57.100
  
  libswscale      5.  5.100 /  5.  5.100
  
  libswresample   3.  5.100 /  3.  5.100
  
  libpostproc    55.  5.100 / 55.  5.100
  
  
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, wav, from 'D:\Research\PhD\Carolina\AD\wav\media.io_Wakeman_Rhyne_001_01.wav':
  
  Duration: 00:17:38.04, bitrate: 1411 kb/s
  
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, null, to 'pipe:':

• PyQt-thread. Get dynamicly output

  1er janvier 2020, par ZPro

  I use PyQt-thread for parallel conversion of mp3 files to aac via ffmpeg.
  Here is my code :

  class SubprocessThread(QThread):
  
      signal = pyqtSignal('PyQt_PyObject')
  
  
  
      def __init__(self, command, args):
  
          QThread.__init__(self)
  
          self.command = command
  
          self.args = args
  
  
  
      def __del__(self):
  
          self.wait()
  
  
  
      def run(self):
  
          output = subprocess.check_output('{0} {1}'.format(self.command, self.args), shell=True).split()
  
          self.signal.emit(output)

  And here is example of usage :

  threads = []
  
  
  
              for part in parts.keys():
  
                  args = "-i \'{0}.mp3\' -c:a aac -b:a {1}k \'{2}.m4a\'".format(
  
                      os.path.join(tmp_dir, str(part)),
  
                      int(self.bitrate_cbx.currentText()),
  
                      os.path.join(tmp_dir, str(part)))
  
                  print(args)  # debug
  
                  ffmpeg_thread = SubprocessThread('ffmpeg', args)
  
                  ffmpeg_thread.signal.connect(self.on_data_ready)
  
                  threads.append(ffmpeg_thread)
  
                  ffmpeg_thread.start()
  
                  self.threads_count += 1

  I want to make progress bar, based on conversion, but ffmpeg always updates last string in his output (when conversion in progress).
  Here is an example of ffmpeg output while files are converting :

  user@host$ ffmpeg -i '/home/user/001.mp3' -c:a aac -b:a 128k -vn '/home/user/test.m4a' 
  
  ffmpeg version n4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 9.2.0 (GCC)
  
    configuration: --prefix=/usr --disable-debug --disable-static --disable-stripping --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d --enable-libdrm --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libiec61883 --enable-libjack --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxcb --enable-libxml2 --enable-libxvid --enable-nvdec --enable-nvenc --enable-omx --enable-shared --enable-version3
  
    libavutil      56. 31.100 / 56. 31.100
  
    libavcodec     58. 54.100 / 58. 54.100
  
    libavformat    58. 29.100 / 58. 29.100
  
    libavdevice    58.  8.100 / 58.  8.100
  
    libavfilter     7. 57.100 /  7. 57.100
  
    libswscale      5.  5.100 /  5.  5.100
  
    libswresample   3.  5.100 /  3.  5.100
  
    libpostproc    55.  5.100 / 55.  5.100
  
  Input #0, mp3, from '/home/user/001.mp3':
  
    Metadata:
  
      encoder         : Lavf57.41.100
  
      title           : test
  
      artist          : test
  
      album_artist    : test
  
      album           : test
  
      composer        : test
  
      genre           : test
  
      date            : 2018
  
    Duration: 00:12:38.02, start: 0.025056, bitrate: 192 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
  
      Metadata:
  
        encoder         : Lavc57.48
  
      Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc, bt470bg/unknown/unknown), 500x500 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc (attached pic)
  
      Metadata:
  
        comment         : Cover (front)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (mp3float) -> aac (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, ipod, to '/home/user/test.m4a':
  
    Metadata:
  
      date            : test
  
      title           : test
  
      artist          : test
  
      album_artist    : test
  
      album           : test
  
      composer        : test
  
      genre           : test
  
      encoder         : Lavf58.29.100
  
      Stream #0:0: Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 128 kb/s
  
      Metadata:
  
        encoder         : Lavc58.54.100 aac
  
  size=   12107kB time=00:12:38.01 bitrate= 130.8kbits/s speed=79.2x 

  How can I receive this data (string, that begins from "size=...") from my parallel QThreads to calculate overall progress ?