1. 26 Jul, 2020 1 commit
  2. 25 Jun, 2020 1 commit
  3. 22 Jun, 2020 1 commit
  4. 16 Jun, 2020 1 commit
  5. 11 Jun, 2020 1 commit
  6. 24 Apr, 2020 1 commit
  7. 14 Apr, 2020 1 commit
  8. 05 Apr, 2020 2 commits
    • Arun Raghavan's avatar
      Drop gst-plugin- prefix in plugin directory name · dc3c8fd0
      Arun Raghavan authored and Arun Raghavan's avatar Arun Raghavan committed
    • Arun Raghavan's avatar
      Reorganise plugins into directories by function · 205b6040
      Arun Raghavan authored and Arun Raghavan's avatar Arun Raghavan committed
      This should start making navigating the tree a little easier to start
      with, and we can then move to allowing building specific groups of
      plugins as well.
      The plugins are moved into the following hierarchy:
          / gst-plugin-audiofx
          / gst-plugin-claxon
          / gst-plugin-csound
          / gst-plugin-lewton
          / gst-plugin-file
          / gst-plugin-sodium
          / gst-plugin-threadshare
          / gst-plugin-reqwest
          / gst-plugin-rusoto
          / gst-plugin-fallbackswitch
          / gst-plugin-togglerecord
          / gst-plugin-cdg
          / gst-plugin-closedcaption
          / gst-plugin-dav1d
          / gst-plugin-flv
          / gst-plugin-gif
          / gst-plugin-rav1e
  9. 02 Apr, 2020 2 commits
  10. 01 Apr, 2020 1 commit
    • Sebastian Dröge's avatar
      audiofx: Add audioloudnorm filter based on the ffmpeg af_loudnorm filter · 666ec7d5
      Sebastian Dröge authored
      This normalizes the loudness of an audio stream to a target loudness
      with a given maximum peak based on EBU R128.
      Conceptually it keeps a 3s lookahead for calculating the perceived
      loudness and based on that calculates the gain required to reach the
      target loudness. The calculated gains then go through a gaussian filter
      for smoothening and are then applied to the audio in 100ms blocks. Each
      of the 100ms blocks is then passed to a limiter filter to prevent going
      above the maximum peak.
      See http://k.ylo.ph/2016/04/04/loudnorm.html for some more details about
      the algorithm.
      It introduces 3s of latency and currently only works on 192kHz audio.
      Using it with a different sample rate requires resampling before and
      afterwards. The upsampling is required to calculate the true peak.
      Other than the ffmpeg filter it currently does not support two-pass
      processing but only one-pass/live processing.
      Compared to the ffmpeg filter this code was refactored considerably and
      the limiter implementation was fixed to actually work, as well as
      various other bugs in different places that were fixed.