webrtc_m130/api/BUILD.gn

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# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
group("api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
deps = []
if (!build_with_mozilla) {
deps += [ ":libjingle_peerconnection_api" ]
}
}
rtc_source_set("call_api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"call/audio_sink.h",
]
}
rtc_source_set("callfactory_api") {
visibility = [ "*" ]
sources = [
"call/call_factory_interface.h",
]
}
if (!build_with_chromium) {
rtc_static_library("create_peerconnection_factory") {
visibility = [ "*" ]
allow_poison = [ "default_task_queue" ]
sources = [
"create_peerconnection_factory.cc",
"create_peerconnection_factory.h",
]
deps = [
":callfactory_api",
":libjingle_peerconnection_api",
":scoped_refptr",
"../api/rtc_event_log:rtc_event_log_factory",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../modules/audio_device:audio_device_api",
"../modules/audio_processing:api",
"../pc:peerconnection",
"../rtc_base",
"../rtc_base:rtc_base_approved",
"audio:audio_mixer_api",
"audio_codecs:audio_codecs_api",
"task_queue:default_task_queue_factory",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/memory",
]
}
}
rtc_source_set("rtp_headers") {
visibility = [ "*" ]
sources = [
"rtp_headers.cc",
"rtp_headers.h",
]
deps = [
":array_view",
"..:webrtc_common",
"units:timestamp",
"video:video_rtp_headers",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("rtp_packet_info") {
visibility = [ "*" ]
sources = [
"rtp_packet_info.cc",
"rtp_packet_info.h",
"rtp_packet_infos.h",
]
deps = [
":array_view",
":refcountedbase",
":rtp_headers",
":scoped_refptr",
"..:webrtc_common",
"../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_static_library("libjingle_peerconnection_api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
cflags = []
sources = [
"async_resolver_factory.h",
"bitrate_constraints.h",
"candidate.cc",
Reland "Clean up libjingle API dependencies." This is a reland of 9185aca9ce1f66e983d9a5e797cab77a64cc46b0 > Original change's description: > > > Clean up libjingle API dependencies. > > > > > > This CL moves candidate.h into the public API, since it has > > > been implicitly included before. > > > > > > This is a straightforward way of solving the circular > > > dependencies involving that file. For instance, > > > libjingle_peerconnection_api includes candidate.h from > > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which > > > depends on _api. In fact, _api can't depend on much at all > > > since it's a very high level abstraction; instead, things > > > should depend on it. > > > > > > Furthermore, we have the case where deprecated headers > > > include headers in internal modules. I just have to turn > > > off include checking for those, but that's not a big deal. > > > > > > This CL punts the problem of callfactoryinterface.h being > > > implicitly included, and pulling in most of the call > > > module with it. This should be addressed in a follow-up > > > CL. > Bug: webrtc:7504 > Change-Id: Icae0ba1a0488550e2871cc65e66d3661707aa5b6 > Reviewed-on: https://webrtc-review.googlesource.com/6460 > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20156} TBR=deadbeef@webrtc.org Bug: webrtc:7504 Change-Id: Ic6598ac2af9355b60bbd289c86dc75e0ae9fed2e Reviewed-on: https://webrtc-review.googlesource.com/6801 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20167}
2017-10-05 14:53:33 +02:00
"candidate.h",
"congestion_control_interface.h",
"crypto/crypto_options.cc",
"crypto/crypto_options.h",
"crypto/frame_decryptor_interface.h",
"crypto/frame_encryptor_interface.h",
"crypto_params.h",
"data_channel_interface.cc",
"data_channel_interface.h",
"data_channel_transport_interface.cc",
"data_channel_transport_interface.h",
"datagram_transport_interface.h",
"dtls_transport_interface.cc",
"dtls_transport_interface.h",
"dtmf_sender_interface.h",
"ice_transport_interface.h",
"jsep.cc",
"jsep.h",
"jsep_ice_candidate.cc",
"jsep_ice_candidate.h",
"jsep_session_description.h",
"media_stream_interface.cc",
"media_stream_interface.h",
"media_stream_proxy.h",
"media_stream_track_proxy.h",
"media_transport_config.cc",
"media_transport_config.h",
"media_transport_interface.cc",
"media_transport_interface.h",
"notifier.h",
"peer_connection_factory_proxy.h",
"peer_connection_interface.cc",
"peer_connection_interface.h",
"peer_connection_proxy.h",
"proxy.cc",
"proxy.h",
"rtc_error.cc",
"rtc_error.h",
"rtp_receiver_interface.cc",
"rtp_receiver_interface.h",
"rtp_sender_interface.cc",
"rtp_sender_interface.h",
"rtp_transceiver_interface.cc",
"rtp_transceiver_interface.h",
"sctp_transport_interface.cc",
"sctp_transport_interface.h",
"set_remote_description_observer_interface.h",
"stats_types.cc",
"stats_types.h",
"turn_customizer.h",
"uma_metrics.h",
"video_track_source_proxy.h",
]
deps = [
":array_view",
":audio_options_api",
":callfactory_api",
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
":fec_controller_api",
":libjingle_logging_api",
":network_state_predictor_api",
":rtc_stats_api",
":rtp_packet_info",
":rtp_parameters",
":scoped_refptr",
"audio:audio_mixer_api",
"audio_codecs:audio_codecs_api",
"rtc_event_log",
"task_queue",
"transport:bitrate_settings",
"transport:network_control",
"transport/media:audio_interfaces",
"transport/media:video_interfaces",
"units:data_rate",
"units:timestamp",
"video:encoded_image",
"video:video_frame",
"video:video_rtp_headers",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
# Basically, don't add stuff here. You might break sensitive downstream
# targets like pnacl. API should not depend on anything outside of this
# file, really. All these should arguably go away in time.
"..:webrtc_common",
"../media:rtc_media_config",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base",
"../rtc_base:checks",
"../rtc_base:deprecation",
"../rtc_base:rtc_base_approved",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("scoped_refptr") {
visibility = [ "*" ]
sources = [
"scoped_refptr.h",
]
}
rtc_source_set("video_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/video_quality_test_fixture.h",
]
deps = [
":fec_controller_api",
":libjingle_peerconnection_api",
":network_state_predictor_api",
":rtp_parameters",
":simulated_network_api",
"../call:fake_network",
"../call:rtp_interfaces",
"../test:test_common",
"../test:video_test_common",
"transport:network_control",
"video_codecs:video_codecs_api",
]
}
rtc_source_set("video_quality_analyzer_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/video_quality_analyzer_interface.h",
]
deps = [
":stats_observer_interface",
"video:encoded_image",
"video:video_frame",
"video:video_rtp_headers",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("track_id_stream_label_map") {
visibility = [ "*" ]
sources = [
"test/track_id_stream_label_map.h",
]
}
rtc_source_set("rtp_parameters") {
visibility = [ "*" ]
sources = [
"media_types.cc",
"media_types.h",
"rtp_parameters.cc",
"rtp_parameters.h",
]
deps = [
":array_view",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("audio_quality_analyzer_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/audio_quality_analyzer_interface.h",
]
deps = [
":stats_observer_interface",
":track_id_stream_label_map",
]
}
rtc_source_set("stats_observer_interface") {
visibility = [ "*" ]
testonly = true
sources = [
"test/stats_observer_interface.h",
]
deps = [
":libjingle_peerconnection_api",
":rtp_parameters",
]
}
rtc_source_set("peer_connection_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/peerconnection_quality_test_fixture.h",
]
deps = [
":audio_quality_analyzer_api",
":callfactory_api",
":fec_controller_api",
":function_view",
":libjingle_peerconnection_api",
":network_state_predictor_api",
":simulated_network_api",
":stats_observer_interface",
":video_quality_analyzer_api",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
"../rtc_base:rtc_base",
"task_queue",
"transport:network_control",
"units:time_delta",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("test_dependency_factory") {
visibility = [ "*" ]
testonly = true
sources = [
"test/test_dependency_factory.cc",
"test/test_dependency_factory.h",
]
deps = [
":video_quality_test_fixture_api",
"../rtc_base:checks",
"../rtc_base:thread_checker",
"//third_party/abseil-cpp/absl/memory",
]
}
if (rtc_include_tests) {
rtc_source_set("create_video_quality_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_video_quality_test_fixture.cc",
"test/create_video_quality_test_fixture.h",
]
deps = [
":fec_controller_api",
":network_state_predictor_api",
":scoped_refptr",
":video_quality_test_fixture_api",
"../video:video_quality_test",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("create_network_emulation_manager") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_network_emulation_manager.cc",
"test/create_network_emulation_manager.h",
]
deps = [
":network_emulation_manager_api",
"../test/network:emulated_network",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("create_peerconnection_quality_test_fixture") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_peerconnection_quality_test_fixture.cc",
"test/create_peerconnection_quality_test_fixture.h",
]
deps = [
":audio_quality_analyzer_api",
":peer_connection_quality_test_fixture_api",
":video_quality_analyzer_api",
"../test/pc/e2e:peerconnection_quality_test",
"//third_party/abseil-cpp/absl/memory",
]
}
}
rtc_source_set("libjingle_logging_api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"rtc_event_log_output.h",
]
}
rtc_source_set("rtc_event_log_output_file") {
visibility = [ "*" ]
sources = [
"rtc_event_log_output_file.cc",
"rtc_event_log_output_file.h",
]
deps = [
":libjingle_logging_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base/system:file_wrapper",
"rtc_event_log",
]
}
rtc_source_set("rtc_stats_api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
cflags = []
sources = [
"stats/rtc_stats.h",
"stats/rtc_stats_collector_callback.h",
"stats/rtc_stats_report.h",
"stats/rtcstats_objects.h",
]
deps = [
":scoped_refptr",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("audio_options_api") {
visibility = [ "*" ]
sources = [
"audio_options.cc",
"audio_options.h",
]
deps = [
":array_view",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("transport_api") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"call/transport.cc",
"call/transport.h",
]
}
rtc_source_set("bitrate_allocation") {
visibility = [ "*" ]
sources = [
"call/bitrate_allocation.h",
]
deps = [
"units:data_rate",
"units:time_delta",
]
}
rtc_source_set("simulated_network_api") {
visibility = [ "*" ]
sources = [
"test/simulated_network.h",
]
deps = [
"../rtc_base",
"../rtc_base:criticalsection",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("network_emulation_manager_api") {
visibility = [ "*" ]
sources = [
"test/network_emulation_manager.h",
]
deps = [
":simulated_network_api",
"../rtc_base",
"units:data_rate",
"units:data_size",
"units:timestamp",
]
}
rtc_source_set("fec_controller_api") {
visibility = [ "*" ]
sources = [
"fec_controller.h",
"fec_controller_override.h",
]
deps = [
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
"../modules:module_fec_api",
"video:video_frame_type",
]
}
rtc_source_set("network_state_predictor_api") {
visibility = [ "*" ]
sources = [
"network_state_predictor.h",
]
}
rtc_source_set("array_view") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"array_view.h",
]
deps = [
"../rtc_base:checks",
"../rtc_base:type_traits",
]
}
rtc_source_set("refcountedbase") {
Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613. Reason for revert: The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660 Original change's description: > Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" > > This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16. > > Reason for revert: > > Breaks Chrome FYI: > > /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check > -> returned 1 > ERROR at //build/split_static_library.gni:12:5: Dependency not allowed. > static_library(target_name) { > ^---------------------------- > The item //content/renderer:renderer > can not depend on //third_party/webrtc/media:rtc_internal_video_codecs > because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [ > //third_party/webrtc/* > //third_party/webrtc_overrides/* > ] > > https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout > > Original change's description: > > GN rtc_* templates: Set default visibility to webrtc_root + "/*" > > > > This means that by default, targets are visible to everything under > > the WebRTC root, but not visible to anything else. > > > > API targets are manually tagged with visibility "*", so that targets > > outside the WebRTC tree can see them. > > > > BUG=webrtc:8254 > > > > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509 > > Reviewed-on: https://webrtc-review.googlesource.com/24140 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#21548} > > TBR=mbonadei@webrtc.org,kwiberg@webrtc.org > > Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8254 > Reviewed-on: https://webrtc-review.googlesource.com/38760 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#21555} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8254 Reviewed-on: https://webrtc-review.googlesource.com/38860 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:54:53 +00:00
visibility = [ "*" ]
sources = [
"ref_counted_base.h",
]
deps = [
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("ice_transport_factory") {
visibility = [ "*" ]
sources = [
"ice_transport_factory.cc",
"ice_transport_factory.h",
]
deps = [
":libjingle_peerconnection_api",
":scoped_refptr",
"../p2p:rtc_p2p",
"../rtc_base",
"../rtc_base/system:rtc_export",
"rtc_event_log:rtc_event_log",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("neteq_simulator_api") {
visibility = [ "*" ]
sources = [
"test/neteq_simulator.cc",
"test/neteq_simulator.h",
]
}
rtc_source_set("function_view") {
visibility = [ "*" ]
sources = [
"function_view.h",
]
deps = [
"../rtc_base:checks",
]
}
if (rtc_include_tests) {
if (rtc_enable_protobuf) {
rtc_source_set("audioproc_f_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/audioproc_float.cc",
"test/audioproc_float.h",
]
deps = [
"../modules/audio_processing",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
"../modules/audio_processing:audioproc_f_impl",
]
}
rtc_source_set("neteq_simulator_factory") {
visibility = [ "*" ]
testonly = true
sources = [
"test/neteq_simulator_factory.cc",
"test/neteq_simulator_factory.h",
]
deps = [
":neteq_simulator_api",
"../modules/audio_coding:neteq_test_factory",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/flags:flag",
"//third_party/abseil-cpp/absl/flags:parse",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
]
}
}
rtc_source_set("simulcast_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/simulcast_test_fixture.h",
]
}
rtc_source_set("create_simulcast_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_simulcast_test_fixture.cc",
"test/create_simulcast_test_fixture.h",
]
deps = [
":simulcast_test_fixture_api",
"../modules/video_coding:simulcast_test_fixture_impl",
"../rtc_base:rtc_base_approved",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("videocodec_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/videocodec_test_fixture.h",
"test/videocodec_test_stats.cc",
"test/videocodec_test_stats.h",
]
deps = [
"..:webrtc_common",
"../modules/video_coding:video_codec_interface",
"../rtc_base:stringutils",
"video:video_frame_type",
"video_codecs:video_codecs_api",
]
}
rtc_source_set("create_videocodec_test_fixture_api") {
visibility = [ "*" ]
testonly = true
sources = [
"test/create_videocodec_test_fixture.cc",
"test/create_videocodec_test_fixture.h",
]
deps = [
":videocodec_test_fixture_api",
"../modules/video_coding:video_codecs_test_framework",
"../modules/video_coding:videocodec_test_impl",
"../rtc_base:rtc_base_approved",
"video_codecs:video_codecs_api",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("mock_audio_mixer") {
testonly = true
sources = [
"test/mock_audio_mixer.h",
]
deps = [
"../test:test_support",
"audio:audio_mixer_api",
]
}
rtc_source_set("mock_fec_controller_override") {
testonly = true
sources = [
"test/mock_fec_controller_override.h",
]
deps = [
":fec_controller_api",
"../test:test_support",
]
}
rtc_source_set("mock_frame_encryptor") {
testonly = true
sources = [
"test/mock_frame_encryptor.cc",
"test/mock_frame_encryptor.h",
]
deps = [
":libjingle_peerconnection_api",
"../test:test_support",
]
}
rtc_source_set("mock_frame_decryptor") {
testonly = true
sources = [
"test/mock_frame_decryptor.cc",
"test/mock_frame_decryptor.h",
]
deps = [
":libjingle_peerconnection_api",
"../test:test_support",
]
}
rtc_source_set("fake_frame_encryptor") {
testonly = true
sources = [
"test/fake_frame_encryptor.cc",
"test/fake_frame_encryptor.h",
]
deps = [
":array_view",
":libjingle_peerconnection_api",
":rtp_parameters",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("fake_frame_decryptor") {
testonly = true
sources = [
"test/fake_frame_decryptor.cc",
"test/fake_frame_decryptor.h",
]
deps = [
":array_view",
":libjingle_peerconnection_api",
":rtp_parameters",
"..:webrtc_common",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
}
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_bitrate_allocator_factory") {
testonly = true
sources = [
"test/mock_video_bitrate_allocator_factory.h",
]
deps = [
"../api/video:video_bitrate_allocator_factory",
"../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 = [
Reland: Implement true negotiation for DatagramTransport with fallback to RTP. In short, the caller places a x-opaque line in SDP for each m= section that uses datagram transport. If the answerer supports datagram transport, it will parse this line and create a datagram transport. It will then echo the x-opaque line into the answer (to indicate that it accepted use of datagram transport). If the offer and answer contain exactly the same x-opaque line, both peers will use datagram transport. If the x-opaque line is omitted from the answer (or is different in the answer) they will fall back to RTP. Note that a different x-opaque line in the answer means the answerer did not understand something in the negotiation proto. Since WebRTC cannot know what was misunderstood, or whether it's still possible to use the datagram transport, it must fall back to RTP. This may change in the future, possibly by passing the answer to the datagram transport, but it's good enough for now. Negotiation consists of four parts: 1. DatagramTransport exposes transport parameters for both client and server perspectives. The client just echoes what it received from the server (modulo any fields it might not have understood). 2. SDP adds a x-opaque line for opaque transport parameters. Identical to x-mt, but this is specific to datagram transport and goes in each m= section, and appears in the answer as well as the offer. - This is propagated to Jsep as part of the TransportDescription. - SDP files: transport_description.h,cc, transport_description_factory.h,cc, media_session.cc, webrtc_sdp.cc 3. JsepTransport/Controller: - Exposes opaque parameters for each mid (m= section). On offerer, this means pre-allocating a datagram transport and getting its parameters. On the answerer, this means echoing the offerer's parameters. - Uses a composite RTP transport to receive from either default RTP or datagram transport until both offer and answer arrive. - If a provisional answer arrives, sets the composite to send on the provisionally selected transport. - Once both offer and answer are set, deletes the unneeded transports and keeps whichever transport is selected. 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP. Bug: webrtc:9719 Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28198}
2019-06-07 10:28:06 -07:00
"test/fake_datagram_transport.h",
"test/fake_media_transport.h",
]
deps = [
":libjingle_peerconnection_api",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("loopback_media_transport") {
testonly = true
sources = [
"test/loopback_media_transport.cc",
"test/loopback_media_transport.h",
]
deps = [
":libjingle_peerconnection_api",
"../rtc_base",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("rtc_api_unittests") {
testonly = true
sources = [
"array_view_unittest.cc",
"function_view_unittest.cc",
"rtc_error_unittest.cc",
"rtc_event_log_output_file_unittest.cc",
"rtp_packet_info_unittest.cc",
"rtp_packet_infos_unittest.cc",
"rtp_parameters_unittest.cc",
"test/loopback_media_transport_unittest.cc",
]
deps = [
":array_view",
":function_view",
":libjingle_peerconnection_api",
":loopback_media_transport",
":rtc_event_log_output_file",
":rtp_packet_info",
":rtp_parameters",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:rtc_base_approved",
"../test:fileutils",
"../test:test_support",
"task_queue:task_queue_default_factory_unittests",
"units:units_unittests",
"video:video_unittests",
"//third_party/abseil-cpp/absl/memory",
]
}
rtc_source_set("compile_all_headers") {
testonly = true
sources = [
"test/compile_all_headers.cc",
]
deps = [
":fake_frame_decryptor",
":fake_frame_encryptor",
":fake_media_transport",
":loopback_media_transport",
":mock_audio_mixer",
":mock_frame_decryptor",
":mock_frame_encryptor",
":mock_peerconnectioninterface",
":mock_rtp",
":mock_video_bitrate_allocator",
":mock_video_bitrate_allocator_factory",
":mock_video_codec_factory",
":mock_video_decoder",
":mock_video_encoder",
":rtc_api_unittests",
"units:units_unittests",
]
}
}