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• Why the result of FFmpeg capture process has no audio to create a webm file ?

  23 mars 2019, par RAM

  The result of my bellow FFmpeg command has no audio (is silent) :

  ffmpeg -f gdigrab -framerate 30 -i desktop -video_size 720x480 -c:v libvpx-vp9 -c:a libopus -b:v 1M -b:a 128K -auto-alt-ref 0 -crf 10 -preset ultrafast output.webm

  But this one has audio :

  ffmpeg -f gdigrab -i desktop -f dshow -i audio="Microphone (4- High Definition Audio Device)" output.mkv

  • How should I capture as webm file by using libopus or libvorbis ?
  • What is the problem in my first command ?

  ---

  My FFmpeg version :

  ffmpeg version N-93439-gb073fb9eea Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20190212
  
    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. 26.100 / 56. 26.100
  
    libavcodec     58. 47.105 / 58. 47.105
  
    libavformat    58. 26.101 / 58. 26.101
  
    libavdevice    58.  7.100 / 58.  7.100
  
    libavfilter     7. 48.100 /  7. 48.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100

• ffmpeg clean all noise background silences in a poscast

  23 mars 2019, par fireDevelop.com

  I have hundreds of podcast without music, just the voice and the room silence.
  In the silences, I have many clicks, respirations, etc...
  I need to clean all silences with a script, keeping intact the voice.

  In this picture you can see my dirty silences

  And here the result I want in all my audios

  When I use some scripts of sox. I don`t get the result I spect because the voice is affected by the script, the room-silence disappear and some clic still in the silences.

  Then in order to keep intact the voice, I want to do this :

  1. Delete all the silences longer than 3 seconds.

  ---

  2. Split all the audio and silences with in a sequence numbers. ie. :

     • 001-Silence-2.0seconds.wav
     • 002-voice.wav
     • 003-Silence-0.25seconds.wav
     • 004-voice.wav
     • 005-Silence-0.75seconds.wav
     • 006-voice.wav
     • ...
     • ...

  ---

  3. Before, run the script I created manually many files with silences of diferents silences I will use :

     • myManuallySilence-0.25seconds.wav
     • myManuallySilence-0.50seconds.wav
     • myManuallySilence-0.75seconds.wav
     • myManuallySilence-0.1seconds.wav
     • myManuallySilence-1.25seconds.wav
     • ...
     • ...
     • myManuallySilence-2.50seconds.wav
     • myManuallySilence-2.75seconds.wav
     • myManuallySilence-3.0seconds.wav

  ---

  4. the script will check the dirty silences duration and replace by the files myManuallySilence-x.xseconds.wav

  ---

  5. merge all files in one wav file, with the original voice and all the silences cleanned.

  ---

  At the moment I have only this script :

  # get the path of Adobe Audition and add timestamp in the output 
  
  filename
  
  fileName=out
  
  current_time=$(date "+%Y.%m.%d-%H.%M.%S")
  
  newFileName=$fileName.$current_time.wav
  
  #yourPathAPP=/Applications/Adobe\ Audition\ CC\ 2019/Adobe\ Audition\ 
  
  CC\ 2019.app
  
  yourPathAPP=/Volumes/6TB/Applications/ocenaudio.app
  
  # # First denoise audio
  
  
  
  # ## Get noise sample
  
   ffmpeg -i in.wav -vn -ss 00:00:00 -t 00:00:01 noise-sample.wav
  
  
  
  # ## Create noise profile
  
   sox noise-sample.wav -n noiseprof noise.prof
  
  
  
  # ## Clean audio from noise
  
   sox in.wav $newFileName noisered noise.prof 0.50
  
  # # Split audio by noise
  
   sox -V3 $newFileName output.wav silence 1 00:00:02.000 - 80d 1 
  
   00:00:02.000 -80d : newfile : restart
  
  
  
  # ####### (these settings worked for my computer mic - maybe we need to 
  
  finetune them later) #######

  Is getting all the voice in separate files like this :
  output001.wav
  output002.wav
  output003.wav
  output004.wav
  ...
  output00x.wav

  Please, any suggestion will be appreciated.
  Thanks so much in advance !

• Transcoding to H264. PTS and DTS sync accross multiple output streams with different bitrates

  25 mars 2019, par timmytimmers

  I have a setup where I am transcoding live feeds from OTA broadcasts to H264 using the Nvidia NVENC encoder. I am also transcoding the audio to AAC. We are trying to output 3 cbr streams and various bitrates. The problem I am running into is that the PTS and DTS on the multiple outputs are not aligning which is critical for our use case. I am hoping there is an easy fix to this but I have not yet been able to locate one. Any thoughts on how to accomplish this ?

  ===> Source Feed <===

  ffprobe udp://@238.224.1.5:59005
  
  ffprobe version N-93005-gd92f06e Copyright (c) 2007-2019 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
  
    configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/circle/ffmpeg_build/include --extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/circle/bin --enable-gpl --enable-libaom --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc
  
    libavutil      56. 26.100 / 56. 26.100
  
    libavcodec     58. 44.100 / 58. 44.100
  
    libavformat    58. 26.100 / 58. 26.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 48.100 /  7. 48.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  [mpeg2video @ 0x558e5a80fa40] Invalid frame dimensions 0x0.
  
      Last message repeated 22 times
  
  Input #0, mpegts, from 'udp://@238.224.1.5:59005:
  
    Duration: N/A, start: 89037.540778, bitrate: N/A
  
    Program 3 
  
      Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), yuv420p(tv, progressive), 1280x720 [SAR 1:1 DAR 16:9], Closed Captions, 59.94 fps, 59.94 tbr, 90k tbn, 119.88 tbc
  
      Stream #0:1[0x34](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, 5.1(side), fltp, 384 kb/s
  
      Stream #0:2[0x35](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, fltp, 192 kb/s

  ===> Command I am currently running to transcode <===

  screen -d -m ffmpeg -i 'udp://@238.224.1.5:59005?fifo_size=1000000&overrun_nonfatal=1' \
  
          -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 6000K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test6000" -metadata service_provider="test" 'udp://@239.1.1.1:59001?pkt_size=1316' \
  
          -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 3500K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test3500" -metadata service_provider="test" 'udp://@239.1.1.2:59002?pkt_size=1316' \
  
          -vcodec h264_nvenc -bf:v 2 -g 120 -rc cbr -b:v 1500K -profile:v high -level 4.0 -acodec aac -ac 2 -b:a 128k -ar 44100 -f mpegts -metadata service_name="test1500" -metadata service_provider="test" 'udp://@239.1.1.3:59003?pkt_size=1316'

  These streams will be eventually mux’d back together for DRM insertion into a ABR stream. Without those values being in sync it will not be ABR compliant.