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Use speexdsp's jitter buffer in the AudioBridge (1.x) #3233

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Aug 4, 2023
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1 change: 1 addition & 0 deletions .github/workflows/janus-ci.yml
Original file line number Diff line number Diff line change
Expand Up @@ -58,6 +58,7 @@ jobs:
libopus-dev
librabbitmq-dev
libsofia-sip-ua-dev
libspeexdsp-dev
libssl-dev
libtool
libvorbis-dev
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1 change: 0 additions & 1 deletion conf/janus.plugin.audiobridge.jcfg.sample
Original file line number Diff line number Diff line change
Expand Up @@ -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)
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7 changes: 5 additions & 2 deletions configure.ac
Original file line number Diff line number Diff line change
Expand Up @@ -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])
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