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authorfrosty <frostinator33@airmail.cc>2026-10-26 10:21:57 -0400
committerfrosty <frostinator33@airmail.cc>2026-10-26 10:21:57 -0400
commit74aecd8b0986d07b50b7ce310d64e645b8b54e38 (patch)
tree560e6434f183eb54cd779936f3a4c9575146a066 /plugins/rtp/meson.build
parent3d7e90d401da02d430a87db0ec7e1db3a6f1a68f (diff)
parent4436001bb361ad4f9e4e05819214ecf0ea77c063 (diff)
Merge 'upstream/master'
Diffstat (limited to 'plugins/rtp/meson.build')
-rw-r--r--plugins/rtp/meson.build92
1 files changed, 0 insertions, 92 deletions
diff --git a/plugins/rtp/meson.build b/plugins/rtp/meson.build
deleted file mode 100644
index dc9d409b..00000000
--- a/plugins/rtp/meson.build
+++ /dev/null
@@ -1,92 +0,0 @@
-dependencies = [
- dep_crypto_vala,
- dep_dino,
- dep_gee,
- dep_glib,
- dep_gmodule,
- dep_gstreamer,
- dep_gstreamer_app,
- dep_gstreamer_audio,
- dep_gstreamer_rtp,
- dep_gstreamer_video,
- dep_gtk4,
- dep_m,
- dep_qlite,
- dep_xmpp_vala,
-]
-sources = files(
- 'src/codec_util.vala',
- 'src/device.vala',
- 'src/gst_fixes.c',
- 'src/module.vala',
- 'src/plugin.vala',
- 'src/register_plugin.vala',
- 'src/stream.vala',
- 'src/video_widget.vala',
-)
-c_args = [
- '-DG_LOG_DOMAIN="rtp"',
-]
-vala_args = []
-if dep_webrtc_audio_processing.found() and get_option('plugin-rtp-webrtc-audio-processing').allowed()
- voice_dependencies = [
- dep_gstreamer_audio,
- dep_webrtc_audio_processing,
- ]
- voice_sources = files(
- 'src/voice_processor_native.cpp',
- )
- voice_c_args = c_args
- if dep_webrtc_audio_processing.version().version_compare('>=2.0')
- voice_c_args += ['-DWEBRTC2']
- elif dep_webrtc_audio_processing.version().version_compare('>=1.0')
- voice_c_args += ['-DWEBRTC1']
- else
- voice_c_args += ['-DWEBRTC0']
- endif
- lib_rtp_voice_processor = static_library('rtp-voice-processor', voice_sources, c_args: voice_c_args, cpp_args: voice_c_args, dependencies: voice_dependencies, install: false)
- dep_rtp_voice_processor = declare_dependency(link_with: lib_rtp_voice_processor)
- vala_args += ['-D', 'WITH_VOICE_PROCESSOR']
- dependencies += [dep_rtp_voice_processor]
- sources += files(
- 'src/voice_processor.vala',
- )
-elif get_option('plugin-rtp-webrtc-audio-processing').enabled()
- error('No compatible webrtc-audio-processing found, but plugin-rtp-webrtc-audio-processing option set.')
-endif
-if dep_gstreamer_rtp.version() == 'unknown' or dep_gstreamer_rtp.version().version_compare('>=1.16')
- vala_args += ['-D', 'GST_1_16']
-endif
-if dep_gstreamer_rtp.version() == 'unknown' or dep_gstreamer_rtp.version().version_compare('>=1.18')
- vala_args += ['-D', 'GST_1_18']
-endif
-if dep_gstreamer_rtp.version() == 'unknown' or dep_gstreamer_rtp.version().version_compare('>=1.20')
- vala_args += ['-D', 'GST_1_20']
-endif
-if meson.get_compiler('vala').version().version_compare('<0.56.1')
- vala_args += ['--vapidir', meson.current_source_dir() / 'vapi']
-endif
-if get_option('plugin-rtp-h264').allowed()
- vala_args += ['-D', 'ENABLE_H264']
-endif
-if get_option('plugin-rtp-msdk').allowed()
- vala_args += ['-D', 'ENABLE_MSDK']
-endif
-if get_option('plugin-rtp-vaapi').allowed()
- vala_args += ['-D', 'ENABLE_VAAPI']
-endif
-if get_option('plugin-rtp-vp9').allowed()
- vala_args += ['-D', 'ENABLE_VP9']
-endif
-
-lib_rtp = shared_library('rtp', sources, name_prefix: '', c_args: c_args, vala_args: vala_args, include_directories: include_directories('src'), dependencies: dependencies, install: true, install_dir: get_option('libdir') / get_option('plugindir'), install_rpath: default_install_rpath)
-dep_rtp = declare_dependency(link_with: lib_rtp, include_directories: include_directories('.'))
-summary('Voice/video calls (rtp)', dep_rtp.found(), bool_yn: true, section: 'Plugins')
-
-if dep_rtp.found()
- summary('H264 codec', get_option('plugin-rtp-h264').allowed(), section: 'RTP configuration')
- summary('VP9 codec', get_option('plugin-rtp-vp9').allowed(), section: 'RTP configuration')
- summary('Intel MediaSDK', get_option('plugin-rtp-msdk').allowed(), section: 'RTP configuration')
- summary('Video Acceleration API', get_option('plugin-rtp-vaapi').allowed(), section: 'RTP configuration')
- summary('Voice preprocessing', dep_webrtc_audio_processing.found(), section: 'RTP configuration')
-endif