56d460902e
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM. Bug: webrtc:8563, b/67926135 Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24 Reviewed-on: https://webrtc-review.googlesource.com/25621 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20877}
395 lines
9.5 KiB
Plaintext
395 lines
9.5 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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public_deps = [
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":libjingle_peerconnection_api",
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]
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}
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rtc_source_set("call_api") {
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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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":audio_mixer_api",
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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_codecs:audio_codecs_api",
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]
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}
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rtc_static_library("libjingle_peerconnection_api") {
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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.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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"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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"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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"rtpparameters.cc",
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"rtpparameters.h",
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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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public_deps = [
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":libjingle_api_deprecated_headers",
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":mediastream_interface_and_implicit_video_frame_api",
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":peerconnection_and_implicit_call_api",
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]
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deps = [
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":optional",
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":rtc_stats_api",
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"audio_codecs:audio_codecs_api",
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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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"..:webrtc_common",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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]
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# This is needed until bugs.webrtc.org/7504 is removed so this target can
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# properly depend on ../media:rtc_media_base
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# TODO(kjellander): Remove this dependency.
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if (is_nacl) {
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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("peerconnection_and_implicit_call_api") {
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# The peerconnectioninterface.h file pulls in call/callfactoryinterface.h
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# and the entire call module with it. We need to either get rid of this
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# dependency or pull most of call/ into the API. For now, silence the warnings
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# this creates since it creates a circular dependency (call very much depends
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# on API). See bugs.webrtc.org/7504.
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check_includes = false
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sources = [
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"peerconnectioninterface.h",
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]
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}
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rtc_source_set("mediastream_interface_and_implicit_video_frame_api") {
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# The mediastreaminterface.h file pulls in in video_frame.h, but the
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# system_wrappers dependency that comes with that breaks pnacl downstream.
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# TODO(phoglund): solve this (see bugs.webrtc.org/7504).
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check_includes = false
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sources = [
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"mediastreaminterface.h",
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]
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deps = [
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"../modules/audio_processing:audio_processing_statistics",
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]
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}
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rtc_source_set("libjingle_api_deprecated_headers") {
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# We need to include headers from undeclared targets here, since they cause
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# circular dependencies. These deprecated headers are going away anyway.
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# See http://bugs.webrtc.org/5883.
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check_includes = false
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sources = [
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"datachannel.h",
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"mediastream.h",
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"mediastreamtrack.h",
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"rtpsender.h",
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"streamcollection.h",
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"videotracksource.h",
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"webrtcsdp.h",
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]
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}
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rtc_source_set("libjingle_logging_api") {
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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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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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# TODO(ossu): Remove once downstream projects have updated.
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rtc_source_set("libjingle_peerconnection") {
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public_deps = [
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"../pc:libjingle_peerconnection",
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]
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}
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rtc_source_set("rtc_stats_api") {
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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:rtc_base_approved",
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]
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}
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rtc_source_set("audio_mixer_api") {
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sources = [
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"audio/audio_mixer.h",
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]
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deps = [
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"../modules:module_api",
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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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sources = [
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"call/transport.h",
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]
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}
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rtc_source_set("video_frame_api") {
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sources = [
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# TODO(phoglund): move i420 files to video_frame_api_i420 after updating
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# downstream. See bugs.webrtc.org/7504.
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"video/i420_buffer.cc",
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"video/i420_buffer.h",
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"video/video_content_type.cc",
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"video/video_content_type.h",
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"video/video_frame.cc",
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"video/video_frame.h",
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"video/video_frame_buffer.cc",
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"video/video_frame_buffer.h",
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"video/video_rotation.h",
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"video/video_timing.cc",
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"video/video_timing.h",
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]
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deps = [
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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]
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# TODO(nisse): This logic is duplicated in multiple places.
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# Define in a single place.
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if (rtc_build_libyuv) {
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deps += [ "$rtc_libyuv_dir" ]
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public_deps = [
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"$rtc_libyuv_dir",
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]
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} else {
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# Need to add a directory normally exported by libyuv.
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include_dirs = [ "$rtc_libyuv_dir/include" ]
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}
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}
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rtc_source_set("video_frame_api_i420") {
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sources = [
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"video/i420_buffer.h",
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]
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deps = [
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":video_frame_api",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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]
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}
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rtc_source_set("array_view") {
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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:rtc_base_approved",
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]
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}
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rtc_source_set("optional") {
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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:rtc_base_approved",
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]
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}
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rtc_source_set("refcountedbase") {
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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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testonly = true
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sources = [
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"test/fakeconstraints.h",
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]
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public_deps = [
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":libjingle_peerconnection_api",
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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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if (rtc_include_tests) {
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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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public_deps = [
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":audio_mixer_api",
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]
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deps = [
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"../test:test_support",
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"//testing/gmock",
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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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public_deps = [
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"../api/video_codecs:video_codecs_api",
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]
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deps = [
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"../test:test_support",
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"//testing/gmock",
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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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":libjingle_peerconnection_api",
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"../api:peerconnection_and_implicit_call_api",
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"../media:rtc_media_base",
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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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}
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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:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../test:test_support",
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]
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}
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}
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