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authorArun Raghavan <>2016-05-12 19:26:55 +0530
committerArun Raghavan <>2019-11-08 17:39:40 +0530
commit74f8456acb6e419f01a325655b18f8a45083e26f (patch)
tree5da80693cc214edff7eaf2be6a1b8606feb633ce /
parenteb912d3605e86509b0e69aa37ba2b3fad805df63 (diff)
rtp: Add a GStreamer-based RTP implementation
This adds a GStreamer-based RTP implementation to replace our own. The original implementation is retained for cases where it is not possible to include GStreamer as a dependency. The idea with this is to be able to start supporting more advanced RTP features such as RTCP, non-PCM audio, and potentially synchronised playback. Signed-off-by: Arun Raghavan <>
Diffstat (limited to '')
1 files changed, 10 insertions, 0 deletions
diff --git a/ b/
index 59cba00cb..e24eccc28 100644
--- a/
+++ b/
@@ -669,6 +669,15 @@ if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
+gst_dep = dependency('gstreamer-1.0', required : get_option('gstreamer'))
+gstapp_dep = dependency('gstreamer-app-1.0', required : get_option('gstreamer'))
+gstrtp_dep = dependency('gstreamer-rtp-1.0', required : get_option('gstreamer'))
+have_gstreamer = false
+if gst_dep.found() and gstapp_dep.found() and gstrtp_dep.found()
+ have_gstreamer = true
# These are required for the CMake file generation
cdata.set('PA_LIBDIR', libdir)
cdata.set('PA_INCDIR', includedir)
@@ -815,6 +824,7 @@ summary = [
'Enable OpenSSL (for Airtunes): @0@'.format(openssl_dep.found()),
'Enable FFTW: @0@'.format(fftw_dep.found()),
'Enable ORC: @0@'.format(have_orcc),
+ 'Enable GStreamer: @0@'.format(have_gstreamer),
'Enable Adrian echo canceller: @0@'.format(get_option('adrian-aec')),
'Enable Speex (resampler, AEC): @0@'.format(speex_dep.found()),
'Enable SoXR (resampler): @0@'.format(soxr_dep.found()),