Use speexdsp's jitter buffer in the AudioBridge (see #3214) (#3233)

.github/workflows/janus-ci.yml

@@ -58,6 +58,7 @@ jobs:
         libopus-dev
         librabbitmq-dev
         libsofia-sip-ua-dev
+        libspeexdsp-dev
         libssl-dev
         libtool
         libvorbis-dev

conf/janus.plugin.audiobridge.jcfg.sample

@@ -10,7 +10,6 @@
 # audiolevel_event = true|false (whether to emit event to other users or not, default=false)
 # audio_active_packets = 100 (number of packets with audio level, default=100, 2 seconds)
 # audio_level_average = 25 (average value of audio level, 127=muted, 0='too loud', default=25)
-# default_prebuffering = number of packets to buffer before decoding each particiant (default=6)
 # default_expectedloss = percent of packets we expect participants may miss, to help with FEC (default=0, max=20; automatically used for forwarders too)
 # default_bitrate = default bitrate in bps to use for the all participants (default=0, which means libopus decides; automatically used for forwarders too)
 # record = true|false (whether this room should be recorded, default=false)

configure.ac

@@ -788,14 +788,17 @@ AC_SUBST([SOFIA_CFLAGS])
 AC_SUBST([SOFIA_LIBS])
 
 PKG_CHECK_MODULES([OPUS],
-                  [opus],
+                  [
+                   opus
+                   speexdsp
+                  ],
                   [
                     AS_IF([test "x$enable_plugin_audiobridge" = "xmaybe"],
                           [enable_plugin_audiobridge=yes])
                   ],
                   [
                     AS_IF([test "x$enable_plugin_audiobridge" = "xyes"],
-                          [AC_MSG_ERROR([libopus not found. See README.md for installation instructions or use --disable-plugin-audiobridge])])
+                          [AC_MSG_ERROR([libopus or libspeexdsp not found. See README.md for installation instructions or use --disable-plugin-audiobridge])])
                   ])
 AC_SUBST([OPUS_CFLAGS])
 AC_SUBST([OPUS_LIBS])