mirror of
https://github.com/klzgrad/naiveproxy.git
synced 2024-11-24 14:26:09 +03:00
473 lines
11 KiB
Plaintext
473 lines
11 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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group("api") {
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visibility = [ "*" ]
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deps = []
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if (!build_with_mozilla) {
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deps += [ ":libjingle_peerconnection_api" ]
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}
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}
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rtc_source_set("call_api") {
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visibility = [ "*" ]
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sources = [
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"call/audio_sink.h",
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]
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deps = [
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# TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
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":transport_api",
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"..:webrtc_common",
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"../rtc_base:rtc_base_approved",
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"audio:audio_mixer_api",
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"audio_codecs:audio_codecs_api",
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]
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}
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rtc_source_set("callfactory_api") {
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visibility = [ "*" ]
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sources = [
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"call/callfactoryinterface.h",
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]
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}
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rtc_static_library("libjingle_peerconnection_api") {
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visibility = [ "*" ]
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cflags = []
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sources = [
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"candidate.cc",
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"candidate.h",
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"cryptoparams.h",
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"datachannelinterface.h",
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"dtmfsenderinterface.h",
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"jsep.cc",
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"jsep.h",
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"jsepicecandidate.h",
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"jsepsessiondescription.h",
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"mediaconstraintsinterface.cc",
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"mediaconstraintsinterface.h",
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"mediastreaminterface.cc",
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"mediastreaminterface.h",
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"mediastreamproxy.h",
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"mediastreamtrackproxy.h",
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"mediatypes.cc",
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"mediatypes.h",
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"notifier.h",
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"peerconnectionfactoryproxy.h",
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"peerconnectioninterface.h",
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"peerconnectionproxy.h",
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"proxy.cc",
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"proxy.h",
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"rtcerror.cc",
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"rtcerror.h",
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"rtp_headers.cc",
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"rtp_headers.h",
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"rtpparameters.cc",
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"rtpparameters.h",
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"rtpreceiverinterface.cc",
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"rtpreceiverinterface.h",
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"rtpsenderinterface.h",
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"rtptransceiverinterface.h",
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"setremotedescriptionobserverinterface.h",
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"statstypes.cc",
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"statstypes.h",
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"turncustomizer.h",
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"umametrics.cc",
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"umametrics.h",
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"videosourceproxy.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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":array_view",
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":audio_options_api",
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":callfactory_api",
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":fec_controller_api",
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":libjingle_logging_api",
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":optional",
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":rtc_stats_api",
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"audio:audio_mixer_api",
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"audio_codecs:audio_codecs_api",
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"transport:bitrate_settings",
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"transport:network_control",
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"video:video_frame",
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# Basically, don't add stuff here. You might break sensitive downstream
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# targets like pnacl. API should not depend on anything outside of this
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# file, really. All these should arguably go away in time.
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"..:typedefs",
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"..:webrtc_common",
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"../logging:rtc_event_log_api",
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"../media:rtc_media_config",
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"../modules/audio_processing:audio_processing_statistics",
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"../rtc_base:checks",
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"../rtc_base:deprecation",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:stringutils",
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]
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if (is_nacl) {
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# This is needed by .h files included from rtc_base.
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deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
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}
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}
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rtc_source_set("libjingle_logging_api") {
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visibility = [ "*" ]
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sources = [
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"rtceventlogoutput.h",
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]
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}
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rtc_source_set("ortc_api") {
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visibility = [ "*" ]
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sources = [
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"ortc/mediadescription.cc",
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"ortc/mediadescription.h",
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"ortc/ortcfactoryinterface.h",
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"ortc/ortcrtpreceiverinterface.h",
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"ortc/ortcrtpsenderinterface.h",
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"ortc/packettransportinterface.h",
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"ortc/rtptransportcontrollerinterface.h",
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"ortc/rtptransportinterface.h",
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"ortc/sessiondescription.cc",
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"ortc/sessiondescription.h",
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"ortc/srtptransportinterface.h",
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"ortc/udptransportinterface.h",
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]
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# For mediastreaminterface.h, etc.
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# TODO(deadbeef): Create a separate target for the common things ORTC and
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# PeerConnection code shares, so that ortc_api can depend on that instead of
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# libjingle_peerconnection_api.
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deps = [
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":libjingle_peerconnection_api",
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":optional",
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"..:webrtc_common",
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"../rtc_base:rtc_base",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("rtc_stats_api") {
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visibility = [ "*" ]
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cflags = []
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sources = [
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"stats/rtcstats.h",
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"stats/rtcstats_objects.h",
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"stats/rtcstatscollectorcallback.h",
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"stats/rtcstatsreport.h",
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]
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deps = [
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("audio_options_api") {
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visibility = [ "*" ]
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sources = [
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"audio_options.cc",
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"audio_options.h",
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]
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deps = [
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":optional",
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("transport_api") {
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visibility = [ "*" ]
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sources = [
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"call/transport.cc",
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"call/transport.h",
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]
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}
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rtc_source_set("fec_controller_api") {
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visibility = [ "*" ]
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sources = [
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"fec_controller.h",
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]
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deps = [
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"..:webrtc_common",
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"../modules:module_fec_api",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_frame.
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# Delete after downstream users are updated.
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rtc_source_set("video_frame_api") {
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visibility = [ "*" ]
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sources = [
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"videosinkinterface.h",
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"videosourceinterface.h",
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]
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:video_frame",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:encoded_frame.
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# Delete after downstream users are updated.
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rtc_source_set("encoded_frame_api") {
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visibility = [ "*" ]
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:encoded_frame",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_stream_decoder.
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# Delete after downstream users are updated.
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rtc_source_set("video_stream_decoder") {
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visibility = [ "*" ]
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:video_stream_decoder",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_stream_decoder_create.
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# Delete after downstream users are updated.
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rtc_source_set("video_stream_decoder_create") {
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visibility = [ "*" ]
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allow_poison = [ "software_video_codecs" ] # TODO(bugs.webrtc.org/7925): Remove.
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:video_stream_decoder_create",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_frame_i420.
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# Delete after downstream users are updated.
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rtc_source_set("video_frame_api_i420") {
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visibility = [ "*" ]
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:video_frame_i420",
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]
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}
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rtc_source_set("array_view") {
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visibility = [ "*" ]
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sources = [
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"array_view.h",
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]
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deps = [
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"../rtc_base:checks",
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"../rtc_base:type_traits",
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]
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}
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rtc_source_set("optional") {
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visibility = [ "*" ]
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sources = [
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"optional.cc",
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"optional.h",
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]
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deps = [
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":array_view",
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"../rtc_base:checks",
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"../rtc_base:sanitizer",
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]
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}
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rtc_source_set("refcountedbase") {
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visibility = [ "*" ]
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sources = [
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"refcountedbase.h",
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]
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deps = [
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"../rtc_base:rtc_base_approved",
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]
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}
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rtc_source_set("libjingle_peerconnection_test_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/fakeconstraints.h",
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]
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deps = [
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":libjingle_peerconnection_api",
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"../rtc_base:rtc_base_approved",
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]
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}
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# TODO(bugs.webrtc.org/9253): Deprecated, replaced by video:video_bitrate_allocation.
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# Delete after downstream users are updated.
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rtc_source_set("video_bitrate_allocation") {
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visibility = [ "*" ]
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public_deps = [ # no-presubmit-check TODO(webrtc:8603)
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"video:video_bitrate_allocation",
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]
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}
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if (rtc_include_tests) {
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if (rtc_enable_protobuf) {
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rtc_source_set("audioproc_f_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/audioproc_float.cc",
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"test/audioproc_float.h",
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]
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deps = [
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"../modules/audio_processing:audio_processing",
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"../modules/audio_processing:audioproc_f_impl",
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]
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}
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}
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rtc_source_set("videocodec_test_fixture_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/videocodec_test_fixture.h",
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"test/videocodec_test_stats.cc",
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"test/videocodec_test_stats.h",
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]
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deps = [
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"..:webrtc_common",
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"../modules/video_coding:video_codec_interface",
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"video_codecs:video_codecs_api",
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]
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}
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rtc_source_set("create_videocodec_test_fixture_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/create_videocodec_test_fixture.cc",
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"test/create_videocodec_test_fixture.h",
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]
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deps = [
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":videocodec_test_fixture_api",
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"../modules/video_coding:video_codecs_test_framework",
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"../modules/video_coding:videocodec_test_impl",
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"../rtc_base:rtc_base_approved",
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"video_codecs:video_codecs_api",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("mock_audio_mixer") {
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testonly = true
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sources = [
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"test/mock_audio_mixer.h",
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]
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deps = [
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"../test:test_support",
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"audio:audio_mixer_api",
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]
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}
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rtc_source_set("mock_rtp") {
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testonly = true
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sources = [
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"test/mock_rtpreceiver.h",
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"test/mock_rtpsender.h",
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]
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deps = [
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":libjingle_peerconnection_api",
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"../test:test_support",
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]
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}
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rtc_source_set("mock_video_codec_factory") {
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testonly = true
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sources = [
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"test/mock_video_decoder_factory.h",
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"test/mock_video_encoder_factory.h",
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]
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deps = [
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"../api/video_codecs:video_codecs_api",
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"../test:test_support",
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]
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}
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rtc_source_set("fakemetricsobserver") {
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testonly = true
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sources = [
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"fakemetricsobserver.cc",
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"fakemetricsobserver.h",
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]
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deps = [
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"../media:rtc_media_base",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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if (!build_with_mozilla) {
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deps += [ ":libjingle_peerconnection_api" ]
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}
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}
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rtc_source_set("rtc_api_unittests") {
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testonly = true
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sources = [
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"array_view_unittest.cc",
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"optional_unittest.cc",
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"ortc/mediadescription_unittest.cc",
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"ortc/sessiondescription_unittest.cc",
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"rtcerror_unittest.cc",
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"rtpparameters_unittest.cc",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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":array_view",
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":libjingle_peerconnection_api",
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":libjingle_peerconnection_test_api",
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":optional",
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":ortc_api",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../test:test_support",
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"units:units_unittests",
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]
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}
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}
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