Improvements for audio desync issues

This incorporates only the audio desync related changes from #11626
* Add the aresample filter with async option to fill in timestamp gaps
* Use the libopus decoder for opus audio instead of ffmpeg's builtin
  decoder

record-and-playback/core/lib/recordandplayback/edl/audio.rb

@@ -21,7 +21,7 @@ module BigBlueButton
   module EDL
     module Audio
       FFMPEG_AEVALSRC = "aevalsrc=s=48000:c=stereo:exprs=0|0"
-      FFMPEG_AFORMAT = "aformat=sample_fmts=s16:sample_rates=48000:channel_layouts=stereo"
+      FFMPEG_AFORMAT = "aresample=async=1000,aformat=sample_fmts=s16:sample_rates=48000:channel_layouts=stereo"
       FFMPEG_WF_CODEC = 'libvorbis'
       FFMPEG_WF_ARGS = ['-c:a', FFMPEG_WF_CODEC, '-q:a', '2', '-f', 'ogg']
       WF_EXT = 'ogg'
@@ -145,6 +145,10 @@ module BigBlueButton
           if audioinfo[input[:filename]][:format][:format_name] == 'wav'
             ffmpeg_cmd += ['-ignore_length', '1']
           end
+          # Prefer using the libopus decoder for opus files, it handles discontinuities better
+          if audioinfo[input[:filename]][:audio][:codec_name] == 'opus'
+            ffmpeg_cmd << '-c:a' << 'libopus'
+          end
           ffmpeg_cmd += ['-i', input[:filename]]
         end