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rtp: Add a GStreamer-based RTP implementation

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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 <arun@arunraghavan.net>
This commit is contained in:
Arun Raghavan 2016-05-12 19:26:55 +05:30
parent eb912d3605
commit 74f8456acb
12 changed files with 638 additions and 85 deletions
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10
meson.build
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@ -669,6 +669,15 @@ if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
endif
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
endif
# 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()),