mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-16 15:20:42 +01:00
78410ad413
This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
646 lines
15 KiB
Text
646 lines
15 KiB
Text
# 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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"asyncresolverfactory.h",
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"bitrate_constraints.h",
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"candidate.cc",
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"candidate.h",
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"crypto/cryptooptions.cc",
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"crypto/cryptooptions.h",
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"crypto/framedecryptorinterface.h",
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"crypto/frameencryptorinterface.h",
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"cryptoparams.h",
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"datachannelinterface.cc",
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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.cc",
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"jsepicecandidate.h",
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"jsepsessiondescription.h",
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"media_transport_interface.cc",
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"media_transport_interface.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.cc",
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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.cc",
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"rtpsenderinterface.h",
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"rtptransceiverinterface.cc",
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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.h",
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"videosourceproxy.h",
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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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":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:encoded_image",
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"video:video_frame",
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"//third_party/abseil-cpp/absl/types:optional",
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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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"../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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"../rtc_base/system:rtc_export",
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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("video_quality_test_fixture_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/video_quality_test_fixture.h",
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]
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deps = [
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":fec_controller_api",
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":libjingle_peerconnection_api",
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":simulated_network_api",
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"../call:fake_network",
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"../call:rtp_interfaces",
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"../test:test_common",
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"../test:video_test_common",
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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("test_dependency_factory") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/test_dependency_factory.cc",
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"test/test_dependency_factory.h",
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]
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deps = [
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":video_quality_test_fixture_api",
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"../rtc_base:thread_checker",
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"//third_party/abseil-cpp/absl/memory",
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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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if (rtc_include_tests) {
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rtc_source_set("create_video_quality_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_video_quality_test_fixture.cc",
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"test/create_video_quality_test_fixture.h",
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]
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deps = [
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":fec_controller_api",
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":video_quality_test_fixture_api",
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"../rtc_base:ptr_util",
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"../video:video_quality_test",
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"//third_party/abseil-cpp/absl/memory",
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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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}
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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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"..:webrtc_common",
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"../rtc_base:rtc_base",
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"//third_party/abseil-cpp/absl/types:optional",
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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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"../rtc_base/system:rtc_export",
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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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"../rtc_base:stringutils",
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"//third_party/abseil-cpp/absl/types:optional",
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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("simulated_network_api") {
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visibility = [ "*" ]
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sources = [
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"test/simulated_network.h",
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]
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deps = [
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"../rtc_base:criticalsection",
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"../rtc_base:rtc_base",
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"//third_party/abseil-cpp/absl/types:optional",
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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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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("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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rtc_source_set("neteq_simulator_api") {
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visibility = [ "*" ]
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sources = [
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"test/neteq_simulator.cc",
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"test/neteq_simulator.h",
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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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rtc_source_set("neteq_simulator_factory") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/neteq_simulator_factory.cc",
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"test/neteq_simulator_factory.h",
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]
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deps = [
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":neteq_simulator_api",
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"../modules/audio_coding:neteq_test_factory",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"//third_party/abseil-cpp/absl/memory",
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]
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}
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}
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rtc_source_set("simulcast_test_fixture_api") {
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visibility = [ "*" ]
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testonly = true
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sources = [
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"test/simulcast_test_fixture.h",
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]
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}
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rtc_source_set("create_simulcast_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_simulcast_test_fixture.cc",
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"test/create_simulcast_test_fixture.h",
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]
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deps = [
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":simulcast_test_fixture_api",
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"../modules/video_coding:simulcast_test_fixture_impl",
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"../rtc_base:rtc_base_approved",
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"video_codecs:video_codecs_api",
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"//third_party/abseil-cpp/absl/memory",
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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("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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"../rtc_base:stringutils",
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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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|
"//third_party/abseil-cpp/absl/memory",
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]
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|
if (!build_with_chromium && is_clang) {
|
|
# 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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|
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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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|
|
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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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|
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rtc_source_set("mock_frame_encryptor") {
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testonly = true
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sources = [
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|
"test/mock_frame_encryptor.cc",
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|
"test/mock_frame_encryptor.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_frame_decryptor") {
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|
testonly = true
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|
sources = [
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|
"test/mock_frame_decryptor.cc",
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|
"test/mock_frame_decryptor.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("fake_frame_encryptor") {
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|
testonly = true
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|
sources = [
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|
"test/fake_frame_encryptor.cc",
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|
"test/fake_frame_encryptor.h",
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|
]
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|
deps = [
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|
":array_view",
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|
":libjingle_peerconnection_api",
|
|
"..:webrtc_common",
|
|
"../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("fake_frame_decryptor") {
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|
testonly = true
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|
sources = [
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"test/fake_frame_decryptor.cc",
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"test/fake_frame_decryptor.h",
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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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|
"..:webrtc_common",
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"../rtc_base:checks",
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|
"../rtc_base:rtc_base_approved",
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|
]
|
|
}
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|
|
rtc_source_set("mock_peerconnectioninterface") {
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_peerconnectioninterface.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection_api",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("mock_rtp") {
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_rtpreceiver.h",
|
|
"test/mock_rtpsender.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection_api",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("mock_video_bitrate_allocator") {
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_video_bitrate_allocator.h",
|
|
]
|
|
|
|
deps = [
|
|
"../api/video:video_bitrate_allocator",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("mock_video_codec_factory") {
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_video_decoder_factory.h",
|
|
"test/mock_video_encoder_factory.h",
|
|
]
|
|
|
|
deps = [
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("mock_video_decoder") {
|
|
visibility = [ "*" ]
|
|
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_video_decoder.cc",
|
|
"test/mock_video_decoder.h",
|
|
]
|
|
|
|
deps = [
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("mock_video_encoder") {
|
|
visibility = [ "*" ]
|
|
|
|
testonly = true
|
|
sources = [
|
|
"test/mock_video_encoder.cc",
|
|
"test/mock_video_encoder.h",
|
|
]
|
|
|
|
deps = [
|
|
"../api/video_codecs:video_codecs_api",
|
|
"../test:test_support",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("fake_media_transport") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/fake_media_transport.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection_api",
|
|
"../rtc_base:checks",
|
|
"../third_party/abseil-cpp/absl/memory:memory",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("loopback_media_transport") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"test/loopback_media_transport.h",
|
|
]
|
|
|
|
deps = [
|
|
":libjingle_peerconnection_api",
|
|
"../rtc_base:checks",
|
|
]
|
|
}
|
|
|
|
rtc_source_set("rtc_api_unittests") {
|
|
testonly = true
|
|
|
|
sources = [
|
|
"array_view_unittest.cc",
|
|
"ortc/mediadescription_unittest.cc",
|
|
"ortc/sessiondescription_unittest.cc",
|
|
"rtcerror_unittest.cc",
|
|
"rtpparameters_unittest.cc",
|
|
"test/loopback_media_transport_unittest.cc",
|
|
]
|
|
|
|
if (!build_with_chromium && is_clang) {
|
|
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
|
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
|
}
|
|
|
|
deps = [
|
|
":array_view",
|
|
":libjingle_peerconnection_api",
|
|
":loopback_media_transport",
|
|
":ortc_api",
|
|
"../rtc_base:checks",
|
|
"../rtc_base:rtc_base_approved",
|
|
"../rtc_base:rtc_base_tests_utils",
|
|
"../test:test_support",
|
|
"units:units_unittests",
|
|
]
|
|
}
|
|
}
|