gstreamer/tests/examples/webrtc/meson.build

examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap', 'webrtctransceiver', 'webrtcrenego']

foreach example : examples
  exe_name = example
  src_file = '@0@.c'.format(example)

  executable(exe_name,
    src_file,
    install: false,
    include_directories : [configinc],
    dependencies : [glib_dep, gst_dep, gstwebrtc_dep],
    c_args : ['-DHAVE_CONFIG_H=1', '-DGST_USE_UNSTABLE_API'],
  )
endforeach
webrtc: Initial support for stream addition/removal Limitations: - No transport changes at all (ICE, DTLS) - Codec changes are untested and probably don't work - Stream removal doesn't remove transports (i.e. non-bundled transports will stay around until webrtcbin is shutdown) - Unified Plan SDP only. No Plan-B support. 2018-11-28 06:23:31 +00:00			`examples = ['webrtc', 'webrtcbidirectional', 'webrtcswap', 'webrtctransceiver', 'webrtcrenego']`
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523 2017-01-31 09:56:59 +00:00
			`foreach example : examples`
			`exe_name = example`
			`src_file = '@0@.c'.format(example)`

			`executable(exe_name,`
			`src_file,`
examples: Don't install non-prefixed examples 2018-06-06 18:32:56 +00:00			`install: false,`
webrtcbin: an element that handles the transport aspects of webrtc connections SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523 2017-01-31 09:56:59 +00:00			`include_directories : [configinc],`
			`dependencies : [glib_dep, gst_dep, gstwebrtc_dep],`
			`c_args : ['-DHAVE_CONFIG_H=1', '-DGST_USE_UNSTABLE_API'],`
			`)`
			`endforeach`