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gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/meson.build

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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
webrtc_sources = [
'dtlstransport.c',
'icetransport.c',
'rtcsessiondescription.c',
'rtpreceiver.c',
'rtpsender.c',
'rtptransceiver.c',
webrtc: Add GstWebRTCDataChannel to the library API This makes it more discoverable for bindings and allows bindings to generate static API for the signals and functions. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>
2020-06-01 11:46:03 +00:00
'datachannel.c',
webrtc: Split sctptransport into lib and implementation parts GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
2021-05-07 06:12:25 +00:00
'sctptransport.c',
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
]
webrtc_headers = [
'dtlstransport.h',
'icetransport.h',
'rtcsessiondescription.h',
'rtpreceiver.h',
'rtpsender.h',
'rtptransceiver.h',
webrtc: Add GstWebRTCDataChannel to the library API This makes it more discoverable for bindings and allows bindings to generate static API for the signals and functions. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>
2020-06-01 11:46:03 +00:00
'datachannel.h',
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
'webrtc_fwd.h',
'webrtc.h',
webrtc: Split sctptransport into lib and implementation parts GstWebRTCSCTPTransport is now made into into an abstract base class that only contains property specifications matching the RTCSctpTransport interface of the W3C WebRTC specification, see https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This class is put into the WebRTC library to expose it for applications and to allow for generation of bindings for non-dynamic languages using GObject introspection. The actual implementation is moved to the subclass WebRTCSCTPTransport located in the WebRTC plugin. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
2021-05-07 06:12:25 +00:00
'sctptransport.h',
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
]
webrtc_enumtypes_headers = [
'dtlstransport.h',
'icetransport.h',
'rtptransceiver.h',
'webrtc_fwd.h',
]
meson: webrtc: use gnome.mkenums_simple() to generate enumtypes files
2018-03-22 14:07:53 +00:00
webrtc_enums = gnome.mkenums_simple('webrtc-enumtypes',
sources : webrtc_enumtypes_headers,
libs: fix API export/import and 'inconsistent linkage' on MSVC For each lib we build export its own API in headers when we're building it, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. The problem was that we had defined all GST_*_API decorators unconditionally to GST_EXPORT. This was intentional and only supposed to be temporary, but caused linker warnings because we tell the linker that we want to export all symbols even those from externall DLLs, and when the linker notices that they were in external DLLS and not present locally it warns. What we need to do when building each library is: export the library's own symbols and import all other symbols. To this end we define e.g. BUILDING_GST_FOO and then we define the GST_FOO_API decorator either to export or to import symbols depending on whether BUILDING_GST_FOO is set or not. That way external users of each library API automatically get the import. While we're at it, add new GST_API_EXPORT in config.h and use that for GST_*_API decorators instead of GST_EXPORT. The right export define depends on the toolchain and whether we're using -fvisibility=hidden or not, so it's better to set it to the right thing directly than hard-coding a compiler whitelist in the public header. We put the export define into config.h instead of passing it via the command line to the compiler because it might contain spaces and brackets and in the autotools scenario we'd have to pass that through multiple layers of plumbing and Makefile/shell escaping and we're just not going to be *that* lucky. The export define is only used if we're compiling our lib, not by external users of the lib headers, so it's not a problem to put it into config.h Also, this means all .c files of libs need to include config.h to get the export marker defined, so fix up a few that didn't include config.h. This commit depends on a common submodule commit that makes gst-glib-gen.mak add an #include "config.h" to generated enum/marshal .c files for the autotools build. https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 10:52:22 +00:00
body_prefix : '#ifdef HAVE_CONFIG_H\n#include "config.h"\n#endif',
meson: webrtc: use gnome.mkenums_simple() to generate enumtypes files
2018-03-22 14:07:53 +00:00
header_prefix : '#include <gst/webrtc/webrtc_fwd.h>',
decorator: 'GST_WEBRTC_API',
install_header: true,
meson: fix install dir for generated header files Nixos installs into a non-standard includedir, so need to take account of the 'includedir' option instead of just hard-coding 'include' here. https://bugzilla.gnome.org/show_bug.cgi?id=794856
2018-08-10 11:57:47 +00:00
install_dir : join_paths(get_option('includedir'), 'gstreamer-1.0/gst/webrtc/'))
meson: webrtc: use gnome.mkenums_simple() to generate enumtypes files
2018-03-22 14:07:53 +00:00
gstwebrtc_c = webrtc_enums[0]
gstwebrtc_h = webrtc_enums[1]
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
webrtc_gen_sources = [gstwebrtc_h]
webrtc: Add gobject-introspection integration https://bugzilla.gnome.org/show_bug.cgi?id=794347
2018-03-15 10:04:00 +00:00
gstwebrtc_dependencies = [gstbase_dep, gstsdp_dep]
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
gstwebrtc = library('gstwebrtc-' + api_version,
webrtc_sources, gstwebrtc_c, gstwebrtc_h,
libs: fix API export/import and 'inconsistent linkage' on MSVC For each lib we build export its own API in headers when we're building it, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. The problem was that we had defined all GST_*_API decorators unconditionally to GST_EXPORT. This was intentional and only supposed to be temporary, but caused linker warnings because we tell the linker that we want to export all symbols even those from externall DLLs, and when the linker notices that they were in external DLLS and not present locally it warns. What we need to do when building each library is: export the library's own symbols and import all other symbols. To this end we define e.g. BUILDING_GST_FOO and then we define the GST_FOO_API decorator either to export or to import symbols depending on whether BUILDING_GST_FOO is set or not. That way external users of each library API automatically get the import. While we're at it, add new GST_API_EXPORT in config.h and use that for GST_*_API decorators instead of GST_EXPORT. The right export define depends on the toolchain and whether we're using -fvisibility=hidden or not, so it's better to set it to the right thing directly than hard-coding a compiler whitelist in the public header. We put the export define into config.h instead of passing it via the command line to the compiler because it might contain spaces and brackets and in the autotools scenario we'd have to pass that through multiple layers of plumbing and Makefile/shell escaping and we're just not going to be *that* lucky. The export define is only used if we're compiling our lib, not by external users of the lib headers, so it's not a problem to put it into config.h Also, this means all .c files of libs need to include config.h to get the export marker defined, so fix up a few that didn't include config.h. This commit depends on a common submodule commit that makes gst-glib-gen.mak add an #include "config.h" to generated enum/marshal .c files for the autotools build. https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 10:52:22 +00:00
c_args : gst_plugins_bad_args + ['-DGST_USE_UNSTABLE_API', '-DBUILDING_GST_WEBRTC'],
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, libsinc],
version : libversion,
soversion : soversion,
meson: Maintain macOS ABI through dylib versioning Requires Meson 0.48, but the feature will be ignored on older versions so it's safe to add it without bumping the requirement. Documentation: https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2018-08-31 09:07:46 +00:00
darwin_versions : osxversion,
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
install : true,
dependencies : gstwebrtc_dependencies,
)
Meson: Use pkg-config generator
2018-04-07 20:33:47 +00:00
pkgconfig.generate(gstwebrtc,
libraries : [gst_dep, gstbase_dep],
variables : pkgconfig_variables,
subdirs : pkgconfig_subdirs,
name : 'gstreamer-webrtc-1.0',
description : 'GStreamer WebRTC support',
)
webrtc: Add gobject-introspection integration https://bugzilla.gnome.org/show_bug.cgi?id=794347
2018-03-15 10:04:00 +00:00
if build_gir
webrtc_gir = gnome.generate_gir(gstwebrtc,
webrtc: Fix gobject-introspection build with meson
2018-03-15 14:45:46 +00:00
sources : webrtc_sources + webrtc_headers + [gstwebrtc_h],
webrtc: Add gobject-introspection integration https://bugzilla.gnome.org/show_bug.cgi?id=794347
2018-03-15 10:04:00 +00:00
namespace : 'GstWebRTC',
nsversion : api_version,
identifier_prefix : 'Gst',
symbol_prefix : 'gst',
export_packages : 'gstreamer-webrtc-1.0',
includes : ['Gst-1.0', 'GstSdp-1.0'],
install : true,
extra_args : gir_init_section + ['-DGST_USE_UNSTABLE_API'] + ['--c-include=gst/webrtc/webrtc.h'],
dependencies : [gstbase_dep, gstsdp_dep]
)
webrtc_gen_sources += webrtc_gir
endif
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
install_headers(webrtc_headers, subdir : 'gstreamer-1.0/gst/webrtc')
gstwebrtc_dep = declare_dependency(link_with: gstwebrtc,
include_directories : libsinc,
webrtc: Add gobject-introspection integration https://bugzilla.gnome.org/show_bug.cgi?id=794347
2018-03-15 10:04:00 +00:00
sources: webrtc_gen_sources,
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
dependencies: gstwebrtc_dependencies)
Meson: Use pkg-config generator
2018-04-07 20:33:47 +00:00
meson.override_dependency('gstreamer-webrtc-1.0', gstwebrtc_dep)
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