webrtc_m130/pc/BUILD.gn

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# Copyright (c) 2016 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.
# Visibility considerations:
#
# Most targets in this file should have visibility ":*", as they are only
# used internally.
# Some functions are cleared for wider webrtc usage; these have default
# visibility (set to "//*", not the gn default of "*").
# These are:
# - rtc_pc
# - session_description
# - simulcast_description
# - sdp_utils
# - media_stream_observer
# - video_track_source
# - libjingle_peerconnection
#
# Some targets are depended on by external users for historical reasons,
# and are therefore marked with visibility "*". This is in the process
# of being removed.
#
# Some targets are only publicly visible in Chrome builds.
# These are marked up as such.
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")
}
if (rtc_build_libsrtp) {
import("//third_party/libsrtp/options.gni")
assert(rtc_build_ssl == libsrtp_build_boringssl,
"Mismatch ssl build settings detected")
assert(rtc_ssl_root == libsrtp_ssl_root, "Mismatch in ssl root detected")
}
group("pc") {
deps = [ ":rtc_pc" ]
}
rtc_source_set("proxy") {
visibility = [ ":*" ]
sources = [ "proxy.h" ]
deps = [
"../api:make_ref_counted",
"../api:scoped_refptr",
"../api/task_queue",
"../rtc_base:event_tracer",
"../rtc_base:rtc_event",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("channel") {
visibility = [
":*",
"../test/peer_scenario",
]
sources = [
"channel.cc",
"channel.h",
]
deps = [
":channel_interface",
":rtp_media_utils",
":rtp_transport_internal",
":session_description",
"../api:libjingle_peerconnection_api",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/units:timestamp",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../media:codec",
"../media:media_channel",
"../media:media_channel_impl",
"../media:rid_description",
"../media:rtp_utils",
"../media:stream_params",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_packet_socket_factory",
"../p2p:dtls_transport_internal",
"../rtc_base:async_packet_socket",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:network_route",
"../rtc_base:socket",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base/containers:flat_set",
"../rtc_base/network:sent_packet",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("channel_interface") {
visibility = [ ":*" ]
sources = [ "channel_interface.h" ]
deps = [
":rtp_transport_internal",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
"../media:media_channel",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("dtls_srtp_transport") {
visibility = [ ":*" ]
sources = [
"dtls_srtp_transport.cc",
"dtls_srtp_transport.h",
]
deps = [
":srtp_transport",
"../api:dtls_transport_interface",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../p2p:dtls_transport_internal",
"../p2p:packet_transport_internal",
"../rtc_base:buffer",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:ssl_adapter",
]
}
rtc_source_set("dtls_transport") {
visibility = [
":*",
"../test/*",
]
sources = [
"dtls_transport.cc",
"dtls_transport.h",
]
deps = [
":ice_transport",
"../api:dtls_transport_interface",
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:scoped_refptr",
"../api:sequence_checker",
"../p2p:dtls_transport",
"../p2p:dtls_transport_internal",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl_adapter",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
]
}
rtc_source_set("external_hmac") {
visibility = [ ":*" ]
sources = [
"external_hmac.cc",
"external_hmac.h",
]
deps = [
"../rtc_base:logging",
"../rtc_base:zero_memory",
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
}
rtc_source_set("ice_transport") {
visibility = [ ":*" ]
sources = [
"ice_transport.cc",
"ice_transport.h",
]
deps = [
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:sequence_checker",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:threading",
]
}
rtc_source_set("jsep_transport") {
visibility = [ ":*" ]
sources = [
"jsep_transport.cc",
"jsep_transport.h",
]
deps = [
":dtls_srtp_transport",
":dtls_transport",
":rtcp_mux_filter",
":rtp_transport",
":rtp_transport_internal",
":sctp_transport",
":session_description",
":srtp_transport",
":transport_stats",
"../api:array_view",
"../api:candidate",
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/transport:datagram_transport_interface",
"../call:payload_type_picker",
"../media:rtc_data_sctp_transport_internal",
"../p2p:dtls_transport",
"../p2p:dtls_transport_internal",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
]
}
rtc_source_set("jsep_transport_collection") {
visibility = [ ":*" ]
sources = [
"jsep_transport_collection.cc",
"jsep_transport_collection.h",
]
deps = [
":jsep_transport",
":session_description",
"../api:libjingle_peerconnection_api",
"../api:sequence_checker",
"../p2p:p2p_constants",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
]
}
rtc_source_set("jsep_transport_controller") {
visibility = [
":*",
"../test/peer_scenario:*",
]
sources = [
"jsep_transport_controller.cc",
"jsep_transport_controller.h",
]
deps = [
":channel",
":dtls_srtp_transport",
":dtls_transport",
":jsep_transport",
":jsep_transport_collection",
":rtp_transport",
":rtp_transport_internal",
":sctp_transport",
":session_description",
":srtp_transport",
":transport_stats",
"../api:async_dns_resolver",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:ice_transport_factory",
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
"../api/environment",
"../api/rtc_event_log",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/transport:sctp_transport_factory_interface",
"../call:payload_type",
"../call:payload_type_picker",
"../media:codec",
"../media:rtc_data_sctp_transport_internal",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:connection",
"../p2p:dtls_transport",
"../p2p:dtls_transport_factory",
"../p2p:dtls_transport_internal",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:packet_transport_internal",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:callback_list",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:crypto_random",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("media_factory") {
sources = [ "media_factory.h" ]
deps = [
"../api/environment",
"../call:call_interfaces",
"../media:media_engine",
]
}
rtc_source_set("media_session") {
visibility = [ "*" ] # Used by Chrome
sources = [
"media_session.cc",
"media_session.h",
]
deps = [
":codec_vendor",
":media_options",
":media_protocol_names",
":rtp_media_utils",
":session_description",
":simulcast_description",
":used_ids",
"../api:field_trials_view",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../call:payload_type",
"../media:codec",
"../media:codec_list",
"../media:media_constants",
"../media:media_engine",
"../media:rid_description",
"../media:rtc_data_sctp_transport_internal",
"../media:stream_params",
"../p2p:ice_credentials_iterator",
"../p2p:p2p_constants",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
"../rtc_base:unique_id_generator",
Reland "Don't create channel_manager++ when media_engine is not set" This reverts commit c6c02efb56b24df04ed9ab61252c14c7bddcca93. Reason for revert: Test now passes (and channel manager is gone) Original change's description: > Revert "Don't create channel_manager when media_engine is not set" > > This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f. > > Reason for revert: breaks downstream project > > Original change's description: > > Don't create channel_manager when media_engine is not set > > > > Also remove a bunch of functions in ChannelManager that were just > > forwarding to MediaEngineInterface. > > > > Bug: webrtc:13931 > > Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36801} > > Bug: webrtc:13931 > Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#36811} Bug: webrtc:13931 Change-Id: I7b5b45b46095c18d489b6a9fe4c625971d6b3da6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261661 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36976}
2022-05-23 14:57:47 +00:00
"../rtc_base/memory:always_valid_pointer",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_library("media_options") {
visibility = [ ":*" ]
sources = [
"media_options.cc",
"media_options.h",
]
deps = [
":simulcast_description",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api/crypto:options",
"../media:codec",
"../media:rid_description",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_library("codec_vendor") {
visibility = [ ":*" ]
sources = [
"codec_vendor.cc",
"codec_vendor.h",
]
deps = [
":media_options",
":rtp_media_utils",
":session_description",
":used_ids",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api/video_codecs:video_codecs_api",
"../media:codec",
"../media:codec_list",
"../media:media_constants",
"../media:media_engine",
"../media:rtc_sdp_video_format_utils",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
"../rtc_base:unique_id_generator",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("media_stream_proxy") {
visibility = [ ":*" ]
sources = [ "media_stream_proxy.h" ]
deps = [
":proxy",
"../api:media_stream_interface",
]
}
rtc_source_set("media_stream_track_proxy") {
visibility = [ ":*" ]
sources = [ "media_stream_track_proxy.h" ]
deps = [
":proxy",
"../api:media_stream_interface",
]
}
rtc_source_set("peer_connection_factory_proxy") {
visibility = [ ":*" ]
sources = [ "peer_connection_factory_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
]
}
rtc_source_set("peer_connection_proxy") {
visibility = [ ":*" ]
sources = [ "peer_connection_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
"../api/transport:bandwidth_estimation_settings",
]
}
rtc_source_set("rtcp_mux_filter") {
visibility = [ ":*" ]
sources = [
"rtcp_mux_filter.cc",
"rtcp_mux_filter.h",
]
deps = [
":session_description",
"../rtc_base:logging",
]
}
rtc_source_set("rtp_media_utils") {
visibility = [ ":*" ]
sources = [
"rtp_media_utils.cc",
"rtp_media_utils.h",
]
deps = [
"../api:rtp_transceiver_direction",
"../rtc_base:checks",
]
}
rtc_source_set("rtp_receiver_proxy") {
visibility = [ ":*" ]
sources = [ "rtp_receiver_proxy.h" ]
deps = [
":proxy",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:dtls_transport_interface",
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
]
}
rtc_source_set("rtp_sender_proxy") {
visibility = [ ":*" ]
sources = [ "rtp_sender_proxy.h" ]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
"../api:rtp_sender_interface",
]
}
rtc_source_set("rtp_transport") {
visibility = [ ":*" ]
sources = [
"rtp_transport.cc",
"rtp_transport.h",
]
deps = [
":rtp_transport_internal",
":session_description",
"../api:array_view",
"../api:field_trials_view",
"../api/task_queue:pending_task_safety_flag",
"../api/units:timestamp",
"../call:rtp_receiver",
"../call:video_receive_stream_api",
"../media:rtp_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:packet_transport_internal",
"../rtc_base:async_packet_socket",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:network_route",
"../rtc_base:socket",
"../rtc_base/network:received_packet",
"../rtc_base/network:sent_packet",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("rtp_transport_internal") {
visibility = [
":*",
"../test/peer_scenario",
]
sources = [ "rtp_transport_internal.h" ]
deps = [
":session_description",
"../call:rtp_receiver",
"../p2p:ice_transport_internal",
"../rtc_base:callback_list",
"../rtc_base:network_route",
"../rtc_base:ssl_adapter",
]
}
rtc_source_set("sctp_transport") {
visibility = [ ":*" ]
sources = [
"sctp_transport.cc",
"sctp_transport.h",
]
deps = [
":dtls_transport",
"../api:dtls_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/transport:datagram_transport_interface",
"../media:rtc_data_sctp_transport_internal",
"../p2p:dtls_transport_internal",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
]
}
rtc_source_set("sctp_utils") {
visibility = [
":*",
"../test/fuzzers:*",
]
sources = [
"sctp_utils.cc",
"sctp_utils.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api/transport:datagram_transport_interface",
"../media:media_channel",
"../media:rtc_data_sctp_transport_internal",
"../net/dcsctp/public:types",
"../rtc_base:byte_buffer",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:logging",
"../rtc_base:ssl_adapter",
]
}
rtc_source_set("srtp_session") {
visibility = [ ":*" ]
sources = [
"srtp_session.cc",
"srtp_session.h",
]
deps = [
":external_hmac",
"../api:array_view",
"../api:field_trials_view",
"../api:scoped_refptr",
"../api:sequence_checker",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:buffer",
"../rtc_base:byte_order",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:string_view",
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
}
rtc_source_set("srtp_transport") {
visibility = [ ":*" ]
sources = [
"srtp_transport.cc",
"srtp_transport.h",
]
deps = [
":rtp_transport",
":srtp_session",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api/units:timestamp",
"../call:rtp_receiver",
"../media:rtp_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:packet_transport_internal",
"../rtc_base:async_packet_socket",
"../rtc_base:buffer",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:network_route",
"../rtc_base:safe_conversions",
"../rtc_base:ssl_adapter",
"../rtc_base:zero_memory",
"../rtc_base/network:received_packet",
"//third_party/abseil-cpp/absl/strings",
]
}
rtc_source_set("transport_stats") {
visibility = [ ":*" ]
sources = [
"transport_stats.cc",
"transport_stats.h",
]
deps = [
"../api:dtls_transport_interface",
"../api:libjingle_peerconnection_api",
"../p2p:connection",
"../p2p:dtls_transport_internal",
"../p2p:ice_transport_internal",
"../p2p:port",
"../rtc_base:ssl_adapter",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("used_ids") {
visibility = [ ":*" ]
sources = [ "used_ids.h" ]
deps = [
"../api:rtp_parameters",
"../media:codec",
"../rtc_base:checks",
"../rtc_base:logging",
]
}
rtc_source_set("video_track_source_proxy") {
visibility = [ "*" ] # Used by Chrome
sources = [
"video_track_source_proxy.cc",
"video_track_source_proxy.h",
]
deps = [
":proxy",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:video_track_source_constraints",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../rtc_base:threading",
]
}
rtc_source_set("session_description") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
visibility = [ "*" ] # Used by Chrome and others
sources = [
"session_description.cc",
"session_description.h",
]
deps = [
":media_protocol_names",
":simulcast_description",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../media:codec",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
"../media:stream_params",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:socket_address",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("simulcast_description") {
sources = [
"simulcast_description.cc",
"simulcast_description.h",
]
deps = [
"../rtc_base:checks",
"../rtc_base:socket_address",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("rtc_pc") {
if (build_with_chromium) {
visibility = [ "*" ]
}
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
deps = [ "../media:rtc_audio_video" ]
}
rtc_library("media_protocol_names") {
visibility = [ ":*" ]
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
sources = [
"media_protocol_names.cc",
"media_protocol_names.h",
]
deps = [ "//third_party/abseil-cpp/absl/strings:string_view" ]
Reland "Reland "Version 2 "Refactoring DataContentDescription class""" This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1. Reason for revert: Tightened protocol name handling. Original change's description: > Revert "Reland "Version 2 "Refactoring DataContentDescription class""" > > This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e. > > Reason for revert: fuzzer failures > > Original change's description: > > Reland "Version 2 "Refactoring DataContentDescription class"" > > > > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c > > > > Original change's description: > > > Version 2 "Refactoring DataContentDescription class" > > > > > > (substantial changes since version 1) > > > > > > This CL splits the cricket::DataContentDescription class into > > > two classes: cricket::RtpDataContentDescription (used for RTP data) > > > and cricket::SctpDataContentDescription (used for SCTP only). > > > > > > SctpDataContentDescription no longer inherits from > > > MediaContentDescriptionImpl, and no longer contains "codecs". > > > > > > Due to usage of internal interfaces by consumers, shimming the old > > > DataContentDescription API is needed. > > > > > > A new cricket::DataContentDescription class is defined, which is > > > a shim over RtpDataContentDescription and SctpDataContentDescription. > > > It exposes as little functionality as possible, but supports the > > > concerned consumer's usage > > > > > > Design document: > > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit# > > > > > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700 > > > Bug: webrtc:10358 Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27933}
2019-05-13 13:36:16 +02:00
}
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
rtc_library("sctp_data_channel") {
visibility = [ ":*" ]
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
sources = [
"sctp_data_channel.cc",
"sctp_data_channel.h",
]
deps = [
":data_channel_utils",
":proxy",
":sctp_utils",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue:pending_task_safety_flag",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../api/transport:datagram_transport_interface",
"../media:media_channel",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../media:rtc_data_sctp_transport_internal",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl_adapter",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:threading",
"../rtc_base:weak_ptr",
"../rtc_base/containers:flat_set",
"../rtc_base/system:no_unique_address",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base/system:unused",
]
}
rtc_library("data_channel_utils") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
visibility = [ "*" ] # Known to be used externally
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
sources = [
"data_channel_utils.cc",
"data_channel_utils.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../media:media_engine",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
"../rtc_base:checks",
]
}
rtc_library("connection_context") {
visibility = [ ":*" ]
sources = [
"connection_context.cc",
"connection_context.h",
]
deps = [
":media_factory",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:refcountedbase",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/environment",
"../api/neteq:neteq_api",
"../api/transport:sctp_transport_factory_interface",
"../media:media_engine",
"../media:rtc_data_sctp_transport_factory",
"../p2p:basic_packet_socket_factory",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:macromagic",
"../rtc_base:network",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:socket_factory",
"../rtc_base:socket_server",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/memory:always_valid_pointer",
]
}
rtc_source_set("data_channel_controller") {
visibility = [ ":*" ]
sources = [
"data_channel_controller.cc",
"data_channel_controller.h",
]
deps = [
":data_channel_utils",
":peer_connection_internal",
":sctp_data_channel",
":sctp_utils",
"../api:libjingle_peerconnection_api",
"../api:priority",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:datagram_transport_interface",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl_adapter",
"../rtc_base:threading",
"../rtc_base:weak_ptr",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_source_set("peer_connection_internal") {
visibility = [ ":*" ]
sources = [ "peer_connection_internal.h" ]
deps = [
":jsep_transport_controller",
":peer_connection_message_handler",
":rtp_transceiver",
":rtp_transmission_manager",
":sctp_data_channel",
"../api:libjingle_peerconnection_api",
"../api/audio:audio_device",
"../call:call_interfaces",
"../modules/audio_device",
]
}
rtc_source_set("rtc_stats_collector") {
visibility = [
":*",
"../api:*",
]
sources = [
"rtc_stats_collector.cc",
"rtc_stats_collector.h",
]
deps = [
":channel",
":channel_interface",
":data_channel_utils",
":peer_connection_internal",
":rtc_stats_traversal",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transceiver",
":sctp_data_channel",
":track_media_info_map",
":transport_stats",
":webrtc_sdp",
"../api:array_view",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_device",
"../api/audio:audio_processing_statistics",
"../api/task_queue:task_queue",
"../api/units:time_delta",
"../api/video:video_rtp_headers",
"../api/video_codecs:scalability_mode",
"../call:call_interfaces",
"../common_video:common_video",
"../media:media_channel",
"../media:media_channel_impl",
"../modules/audio_device",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:port",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:network_constants",
"../rtc_base:refcount",
"../rtc_base:rtc_event",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/containers:flat_set",
"../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("rtc_stats_traversal") {
visibility = [ ":*" ]
sources = [
"rtc_stats_traversal.cc",
"rtc_stats_traversal.h",
]
deps = [
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../rtc_base:checks",
]
}
rtc_source_set("sdp_munging_detector") {
visibility = [ ":*" ]
sources = [
"sdp_munging_detector.cc",
"sdp_munging_detector.h",
]
deps = [
":session_description",
"../api:libjingle_peerconnection_api",
"../media:codec",
"../media:media_constants",
"../media:stream_params",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:logging",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_source_set("sdp_offer_answer") {
visibility = [ ":*" ]
sources = [
"sdp_offer_answer.cc", # TODO: Make separate target when not circular
"sdp_offer_answer.h", # dependent on peerconnection.h
]
deps = [
":channel",
":channel_interface",
":connection_context",
":data_channel_controller",
":dtls_transport",
":jsep_transport_controller",
":legacy_stats_collector",
":media_session",
":media_stream",
":media_stream_observer",
":media_stream_proxy",
":peer_connection_internal",
":peer_connection_message_handler",
":rtp_media_utils",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transceiver",
":rtp_transmission_manager",
":sdp_munging_detector",
":sdp_state_provider",
":session_description",
":simulcast_description",
":stream_collection",
":transceiver_list",
":usage_pattern",
":used_ids",
":webrtc_session_description_factory",
"../api:array_view",
"../api:audio_options_api",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocator_factory",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api/video:video_codec_constants",
"../call:payload_type",
"../media:codec",
"../media:media_channel",
"../media:media_constants",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../media:media_engine",
"../media:rid_description",
"../media:stream_params",
"../p2p:connection",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_operations_chain",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("jsep_ice_candidate") {
visibility = [ ":*" ]
}
rtc_source_set("jsep_session_description") {
visibility = [ ":*" ]
}
rtc_source_set("local_audio_source") {
visibility = [ ":*" ]
sources = [
"local_audio_source.cc",
"local_audio_source.h",
]
deps = [
"../api:audio_options_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
]
}
rtc_source_set("peer_connection") {
visibility = [ ":*" ]
sources = [
"peer_connection.cc",
"peer_connection.h",
]
deps = [
":channel",
":channel_interface",
":connection_context",
":data_channel_controller",
":data_channel_utils",
":dtls_transport",
":ice_server_parsing",
":jsep_transport_controller",
":legacy_stats_collector",
":peer_connection_internal",
":peer_connection_message_handler",
":rtc_stats_collector",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transceiver",
":rtp_transmission_manager",
":rtp_transport_internal",
":sctp_data_channel",
":sctp_transport",
":sdp_offer_answer",
":session_description",
":simulcast_description",
":transceiver_list",
":transport_stats",
":usage_pattern",
":webrtc_session_description_factory",
"../api:async_dns_resolver",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:field_trials_view",
"../api:ice_transport_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:turn_customizer",
"../api/adaptation:resource_adaptation_api",
"../api/audio:audio_device",
"../api/crypto:options",
"../api/environment",
"../api/rtc_event_log",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:bandwidth_estimation_settings",
"../api/transport:bitrate_settings",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/transport:network_control",
"../api/units:time_delta",
"../api/video:video_codec_constants",
"../call:call_interfaces",
"../call:payload_type_picker",
"../media:codec",
"../media:media_channel",
"../media:media_engine",
"../media:rid_description",
"../media:rtc_media_config",
"../media:stream_params",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_async_resolver_factory",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:dtls_transport_internal",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_transport_channel",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:crypto_random",
"../rtc_base:event_tracer",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:net_helpers",
"../rtc_base:network",
"../rtc_base:network_constants",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("simulcast_sdp_serializer") {
visibility = [ ":*" ]
sources = [
"simulcast_sdp_serializer.cc",
"simulcast_sdp_serializer.h",
]
deps = [
":session_description",
":simulcast_description",
"../api:rtc_error",
"../media:rid_description",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:stringutils",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("sdp_utils") {
sources = [
"sdp_utils.cc",
"sdp_utils.h",
]
deps = [
":session_description",
"../api:libjingle_peerconnection_api",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("legacy_stats_collector") {
visibility = [ ":*" ]
sources = [
"legacy_stats_collector.cc",
"legacy_stats_collector.h",
]
deps = [
":channel",
":channel_interface",
":data_channel_utils",
":legacy_stats_collector_interface",
":peer_connection_internal",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender_proxy",
":rtp_transceiver",
":transport_stats",
"../api:candidate",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_processing_statistics",
"../api/audio_codecs:audio_codecs_api",
"../api/video:video_rtp_headers",
"../call:call_interfaces",
"../media:media_channel",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:port",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:network_constants",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("stream_collection") {
visibility = [ ":*" ]
sources = [ "stream_collection.h" ]
deps = [ "../api:libjingle_peerconnection_api" ]
}
rtc_source_set("track_media_info_map") {
visibility = [ ":*" ]
sources = [
"track_media_info_map.cc",
"track_media_info_map.h",
]
deps = [
":rtp_receiver",
":rtp_sender",
"../api:array_view",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../media:media_channel",
"../media:stream_params",
"../rtc_base:checks",
"../rtc_base:refcount",
"../rtc_base:threading",
]
}
rtc_source_set("webrtc_sdp") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
visibility = [ "*" ] # Used by Chrome and more
sources = [
"jsep_ice_candidate.cc",
"jsep_session_description.cc",
"webrtc_sdp.cc",
"webrtc_sdp.h",
]
deps = [
":media_protocol_names",
":media_session",
":session_description",
":simulcast_description",
":simulcast_sdp_serializer",
"../api:candidate",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../media:codec",
"../media:media_constants",
"../media:rid_description",
"../media:rtc_data_sctp_transport_internal",
"../media:rtp_utils",
"../media:stream_params",
"../p2p:candidate_pair_interface",
"../p2p:connection",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:port",
"../p2p:port_interface",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:net_helpers",
"../rtc_base:network_constants",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_source_set("webrtc_session_description_factory") {
visibility = [ ":*" ]
sources = [
"webrtc_session_description_factory.cc",
"webrtc_session_description_factory.h",
]
deps = [
":connection_context",
":media_session",
":sdp_state_provider",
":session_description",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../call:payload_type",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
rtc_library("ice_server_parsing") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
visibility = [ "*" ] # Known to be used externally
sources = [
"ice_server_parsing.cc",
"ice_server_parsing.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../p2p:connection",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:port_interface",
"../rtc_base:checks",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:socket_address",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
]
}
rtc_library("media_stream_observer") {
sources = [
"media_stream_observer.cc",
"media_stream_observer.h",
]
deps = [
"../api:media_stream_interface",
"../api:scoped_refptr",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_source_set("peer_connection_factory") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
visibility = [ "*" ] # Known to be used externally
allow_poison = [ "environment_construction" ]
sources = [
"peer_connection_factory.cc",
"peer_connection_factory.h",
]
deps = [
":local_audio_source",
":media_stream_proxy",
":media_stream_track_proxy",
":peer_connection",
":peer_connection_factory_proxy",
":peer_connection_proxy",
"../api:audio_options_api",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:network_state_predictor_api",
"../api:packet_socket_factory",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/environment",
"../api/environment:environment_factory",
"../api/metronome",
"../api/neteq:neteq_api",
"../api/rtc_event_log:rtc_event_log",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/transport:sctp_transport_factory_interface",
"../api/units:data_rate",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../call:rtp_sender",
"../media:media_engine",
"../p2p:basic_packet_socket_factory",
"../p2p:basic_port_allocator",
"../p2p:connection",
"../p2p:default_ice_transport_factory",
"../p2p:port_allocator",
"../pc:audio_track",
"../pc:connection_context",
"../pc:media_factory",
"../pc:media_stream",
"../pc:rtp_parameters_conversion",
"../pc:session_description",
"../pc:video_track",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:safe_conversions",
"../rtc_base:threading",
"../rtc_base/experiments:field_trial_parser",
"../rtc_base/system:file_wrapper",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_library("peer_connection_message_handler") {
visibility = [ ":*" ]
sources = [
"peer_connection_message_handler.cc",
"peer_connection_message_handler.h",
]
deps = [
":legacy_stats_collector_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../rtc_base:checks",
]
}
rtc_library("usage_pattern") {
visibility = [ ":*" ]
sources = [
"usage_pattern.cc",
"usage_pattern.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../rtc_base:logging",
"../system_wrappers:metrics",
]
}
rtc_library("rtp_transceiver") {
visibility = [ ":*" ]
sources = [
"rtp_transceiver.cc",
"rtp_transceiver.h",
]
deps = [
":channel",
":channel_interface",
":connection_context",
":proxy",
":rtp_media_utils",
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transport_internal",
":session_description",
"../api:array_view",
"../api:audio_options_api",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:options",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/video:video_bitrate_allocator_factory",
Reland "Run IWYU on some files I intend to work on" This reverts commit fe34363ca0ff9d79d7d0943a98ae3a5198e61f75. Reason for revert: Downstream error fixed. Original change's description: > Revert "Run IWYU on some files I intend to work on" > > This reverts commit 827da15f1408a399ed15ce5c9726b6af772fb71a. > > Reason for revert: Breaks downstream project > > Original change's description: > > Run IWYU on some files I intend to work on > > > > and files that broke when I fixed the first set. > > > > Bug: webrtc:42226242 > > Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42429} > > Bug: webrtc:42226242 > Change-Id: I6b18dced08669c6741c6a51768fbb8b9072c6e82 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353580 > Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#42430} Bug: webrtc:42226242 Change-Id: I8ba51da47ea34d6bbf868e5ebc0037c6cffec8ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353660 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42437}
2024-06-04 21:29:14 +00:00
"../api/video_codecs:scalability_mode",
"../media:codec",
"../media:media_channel",
"../media:media_channel_impl",
"../media:media_constants",
"../media:media_engine",
"../media:rtc_media_config",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_library("rtp_transmission_manager") {
visibility = [ ":*" ]
sources = [
"rtp_transmission_manager.cc",
"rtp_transmission_manager.h",
]
deps = [
":audio_rtp_receiver",
":channel",
":channel_interface",
":legacy_stats_collector_interface",
":rtp_receiver",
":rtp_receiver_proxy",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transceiver",
":transceiver_list",
":usage_pattern",
":video_rtp_receiver",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/environment",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../rtc_base:weak_ptr",
]
}
rtc_library("transceiver_list") {
visibility = [ ":*" ]
sources = [
"transceiver_list.cc",
"transceiver_list.h",
]
deps = [
":rtp_transceiver",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
]
}
rtc_library("rtp_receiver") {
visibility = [ ":*" ]
sources = [
"rtp_receiver.cc",
"rtp_receiver.h",
]
deps = [
":media_stream",
":media_stream_proxy",
":video_track_source",
"../api:dtls_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api/crypto:frame_decryptor_interface",
"../api/video:video_frame",
"../media:media_channel",
"../media:video_broadcaster",
"../rtc_base:checks",
"../rtc_base:logging",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
]
}
rtc_library("audio_rtp_receiver") {
visibility = [ ":*" ]
sources = [
"audio_rtp_receiver.cc",
"audio_rtp_receiver.h",
]
deps = [
":audio_track",
":jitter_buffer_delay",
":media_stream",
":media_stream_track_proxy",
":remote_audio_source",
":rtp_receiver",
"../api:dtls_transport_interface",
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:frame_decryptor_interface",
"../api/task_queue:pending_task_safety_flag",
"../api/transport/rtp:rtp_source",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
]
}
rtc_library("video_rtp_receiver") {
visibility = [ ":*" ]
sources = [
"video_rtp_receiver.cc",
"video_rtp_receiver.h",
]
deps = [
":jitter_buffer_delay",
":media_stream",
":media_stream_track_proxy",
":rtp_receiver",
":video_rtp_track_source",
":video_track",
"../api:dtls_transport_interface",
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:frame_decryptor_interface",
"../api/transport/rtp:rtp_source",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
]
}
rtc_library("video_rtp_track_source") {
visibility = [ ":*" ]
sources = [
"video_rtp_track_source.cc",
"video_rtp_track_source.h",
]
deps = [
":video_track_source",
"../api:sequence_checker",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../media:video_broadcaster",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base/synchronization:mutex",
"../rtc_base/system:no_unique_address",
]
}
rtc_library("audio_track") {
visibility = [ ":*" ]
sources = [
"audio_track.cc",
"audio_track.h",
]
deps = [
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../rtc_base:checks",
"../rtc_base/system:no_unique_address",
]
}
rtc_library("video_track") {
visibility = [ ":*" ]
sources = [
"video_track.cc",
"video_track.h",
]
deps = [
":video_track_source_proxy",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/video:video_frame",
"../media:video_source_base",
"../rtc_base:checks",
"../rtc_base:macromagic",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../rtc_base/system:no_unique_address",
]
}
rtc_source_set("sdp_state_provider") {
visibility = [ ":*" ]
sources = [ "sdp_state_provider.h" ]
deps = [ "../api:libjingle_peerconnection_api" ]
}
rtc_library("jitter_buffer_delay") {
visibility = [ ":*" ]
sources = [
"jitter_buffer_delay.cc",
"jitter_buffer_delay.h",
]
deps = [
"../api:sequence_checker",
"../rtc_base:checks",
"../rtc_base:macromagic",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base:safe_conversions",
"../rtc_base:safe_minmax",
VideoRtpReceiver & AudioRtpReceiver threading fixes. For implementations where the signaling and worker threads are not the same thread, this significantly cuts down on Thread::Invoke()s that would block the signaling thread while waiting for the worker thread. For Audio and Video Rtp receivers, the following methods now do not block the signaling thread: * GetParameters * SetJitterBufferMinimumDelay * GetSources * SetFrameDecryptor / GetFrameDecryptor * SetDepacketizerToDecoderFrameTransformer Importantly this change also makes the track() accessor accessible directly from the application thread (bypassing the proxy) since for receiver objects, the track object is const. Other changes: * Remove RefCountedObject inheritance, use make_ref_counted instead. * Every member variable in the rtp receiver classes is now RTC_GUARDED * Stop() now fully clears up worker thread state, and Stop() is consistently called before destruction. This means that there's one thread hop instead of at least 4 before (sometimes more), per receiver. * OnChanged triggered volume for audio tracks is done asynchronously. * Deleted most of the JitterBufferDelay implementation. Turns out that it was largely unnecessary overhead and complexity. It seems that these two classes are copy/pasted to a large extent so further refactoring would be good in the future, as to not have to fix each issue twice. Bug: chromium:1184611 Change-Id: I1ba5c3abbd1b0571f7d12850d64004fd2d83e5e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218605 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34022}
2021-05-17 14:50:10 +02:00
"../rtc_base/system:no_unique_address",
]
}
rtc_library("remote_audio_source") {
visibility = [ ":*" ]
sources = [
"remote_audio_source.cc",
"remote_audio_source.h",
]
deps = [
":channel",
"../api:call_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:safe_conversions",
"../rtc_base:stringutils",
"../rtc_base/synchronization:mutex",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_library("rtp_sender") {
visibility = [ ":*" ]
sources = [
"rtp_sender.cc",
"rtp_sender.h",
]
deps = [
":dtmf_sender",
":legacy_stats_collector_interface",
"../api:audio_options_api",
"../api:dtls_transport_interface",
"../api:dtmf_sender_interface",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:priority",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:frame_encryptor_interface",
"../api/environment",
"../media:audio_source",
"../media:media_channel",
"../media:media_engine",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/sigslot",
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
rtc_library("rtp_parameters_conversion") {
visibility = [ ":*" ]
sources = [
"rtp_parameters_conversion.cc",
"rtp_parameters_conversion.h",
]
deps = [
":session_description",
"../api:array_view",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:rtp_parameters",
"../media:codec",
"../media:media_constants",
"../media:rtp_utils",
"../media:stream_params",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:stringutils",
]
}
rtc_library("dtmf_sender") {
visibility = [ ":*" ]
sources = [
"dtmf_sender.cc",
"dtmf_sender.h",
]
deps = [
":proxy",
"../api:dtmf_sender_interface",
"../api:libjingle_peerconnection_api",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/units:time_delta",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:refcount",
"../rtc_base/third_party/sigslot",
]
}
rtc_library("media_stream") {
visibility = [ ":*" ]
sources = [
"media_stream.cc",
"media_stream.h",
]
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:scoped_refptr",
"../rtc_base:checks",
]
}
rtc_library("video_track_source") {
sources = [
"video_track_source.cc",
"video_track_source.h",
]
deps = [
"../api:media_stream_interface",
"../api:sequence_checker",
"../api/video:recordable_encoded_frame",
"../api/video:video_frame",
"../media:media_channel",
"../rtc_base:checks",
"../rtc_base:macromagic",
Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." This is a reland of 3ed36c0521546881656c73984456485dcab16205 Original change's description: > Reland "Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver." > > This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f > > The difference from the original CL is that a check for > `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed. > This caused a side effect that registering the sink while the source > was in an "initializing" state, failed. The last remaining state > however, is `kEnded` - but since there's no logic in the class around > the expected value of the states, the check inside of AddSink() > doesn't provide an additional value - it's rather a surprise for > developers if it doesn't succeed. So, now removed. > > Original change's description: > > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver. > > > > This simplifies the logic in these classes a bit, which makes upcoming > > change easier. The `stopped_` flag in these classes was essentially > > the same thing as `media_channel_ == nullptr`, which is what's > > consistently used now for the same checks. > > > > Bug: webrtc:13540 > > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35907} > > Bug: webrtc:13540 > Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35958} Bug: webrtc:13540 Change-Id: I6d7d67fddb1ddfc69a302f0f69a9b815f2fd82f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251386 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35967}
2022-02-08 21:12:15 +01:00
"../rtc_base/system:no_unique_address",
"../rtc_base/system:rtc_export",
]
}
rtc_source_set("legacy_stats_collector_interface") {
visibility = [ ":*" ]
sources = [ "legacy_stats_collector_interface.h" ]
deps = [
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
]
}
# This target contains the libraries that are required in order to get an
# usable peerconnection-using binary.
rtc_source_set("libjingle_peerconnection") {
# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
visibility = [ "*" ] # Used by Chrome and others
allow_poison = [ "environment_construction" ]
deps = [
":jsep_session_description",
":peer_connection_factory",
":rtc_stats_collector",
"../api:libjingle_peerconnection_api",
"../stats",
]
}
if (rtc_include_tests && !build_with_chromium) {
rtc_test("rtc_pc_unittests") {
testonly = true
sources = [
"audio_rtp_receiver_unittest.cc",
"channel_unittest.cc",
"codec_vendor_unittest.cc",
"dtls_srtp_transport_integrationtest.cc",
"dtls_srtp_transport_unittest.cc",
"dtls_transport_unittest.cc",
"ice_transport_unittest.cc",
"jsep_transport_controller_unittest.cc",
"jsep_transport_unittest.cc",
"media_session_unittest.cc",
"rtcp_mux_filter_unittest.cc",
"rtp_transport_unittest.cc",
"sctp_transport_unittest.cc",
"session_description_unittest.cc",
"srtp_session_unittest.cc",
"srtp_transport_unittest.cc",
"test/rtp_transport_test_util.h",
"test/srtp_test_util.h",
"used_ids_unittest.cc",
"video_rtp_receiver_unittest.cc",
]
if (is_win) {
libs = [ "strmiids.lib" ]
}
deps = [
":audio_rtp_receiver",
":channel",
":codec_vendor",
":dtls_srtp_transport",
":dtls_transport",
":ice_transport",
":jsep_transport",
":jsep_transport_controller",
":libjingle_peerconnection",
":media_options",
":media_protocol_names",
":media_session",
":pc_test_utils",
":rtc_pc",
":rtcp_mux_filter",
":rtp_media_utils",
":rtp_parameters_conversion",
":rtp_transport",
":rtp_transport_internal",
":sctp_transport",
":session_description",
":simulcast_description",
":srtp_session",
":srtp_transport",
":transport_stats",
":used_ids",
":video_rtp_receiver",
"../api:array_view",
"../api:audio_options_api",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:ice_transport_factory",
"../api:ice_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:make_ref_counted",
"../api:priority",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:options",
"../api/environment:environment",
"../api/environment:environment_factory",
"../api/task_queue:pending_task_safety_flag",
"../api/task_queue:task_queue",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:recordable_encoded_frame",
"../api/video/test:mock_recordable_encoded_frame",
"../api/video_codecs:video_codecs_api",
Reland "Use PayloadTypePicker for video PT assignment" This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95. Reason for revert: Revised codec matching to fix issue. Changes also back out some changes that should not have been included (using PayloadTypePicker for codec list merging). Original change's description: > Revert "Use PayloadTypePicker for video PT assignment" > > This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3. > > Reason for revert: Broke internal tests. > > Original change's description: > > Use PayloadTypePicker for video PT assignment > > > > This includes changes that change the order of codecs. > > It is preparatory to doing late assignment of video PTs. > > > > Bug: webrtc:360058654 > > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400 > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#43489} > > Bug: webrtc:360058654 > Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43490} Bug: webrtc:360058654 Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43554}
2024-12-12 22:15:39 +00:00
"../call:fake_payload_type_suggester",
"../call:payload_type_picker",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../media:codec",
"../media:codec_list",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
"../media:rtc_data_sctp_transport_internal",
"../media:rtc_media_tests_utils",
"../media:stream_params",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:candidate_pair_interface",
"../p2p:dtls_transport",
"../p2p:dtls_transport_factory",
"../p2p:dtls_transport_internal",
"../p2p:fake_ice_transport",
"../p2p:fake_port_allocator",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_test_utils",
"../p2p:packet_transport_internal",
"../p2p:port_allocator",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../p2p:transport_info",
"../rtc_base:async_packet_socket",
"../rtc_base:buffer",
"../rtc_base:byte_order",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:crypto_random",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:network_route",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:socket",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base/containers:flat_set",
"../rtc_base/network:ecn_marking",
"../rtc_base/network:received_packet",
"../rtc_base/network:sent_packet",
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
"../test:explicit_key_value_config",
"../test:run_loop",
"../test:scoped_key_value_config",
"../test:test_main",
"../test:test_support",
"../test:wait_until",
"//testing/gmock",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
include_dirs = []
if (rtc_build_libsrtp) {
include_dirs += [ "//third_party/libsrtp/srtp" ]
deps += [ "//third_party/libsrtp" ]
if (!rtc_build_ssl) {
configs += [ "..:external_ssl_library" ]
}
}
if (is_android) {
use_default_launcher = false
deps += [ "//build/android/gtest_apk:native_test_instrumentation_test_runner_java" ]
}
}
rtc_library("peerconnection_perf_tests") {
testonly = true
sources = [ "peer_connection_rampup_tests.cc" ]
deps = [
":pc_test_utils",
":peer_connection",
":peerconnection_wrapper",
"../api:audio_options_api",
"../api:enable_media_with_defaults",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../api/test/metrics:global_metrics_logger_and_exporter",
"../api/test/metrics:metric",
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
"../p2p:basic_packet_socket_factory",
"../p2p:p2p_test_utils",
"../p2p:port_interface",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:rtc_base_tests_utils",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:socket_address",
"../rtc_base:socket_factory",
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../system_wrappers",
"../test:test_support",
"../test:wait_until",
]
}
rtc_library("peerconnection_wrapper") {
testonly = true
sources = [
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
]
deps = [
":pc_test_utils",
":peer_connection",
":peer_connection_proxy",
":sdp_utils",
"../api:function_view",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../rtc_base:checks",
"../rtc_base:logging",
"../test:test_support",
"../test:wait_until",
]
}
rtc_test("slow_peer_connection_unittests") {
testonly = true
sources = [ "slow_peer_connection_integration_test.cc" ]
deps = [
":integration_test_helpers",
":pc_test_utils",
"../api:dtmf_sender_interface",
"../api:libjingle_peerconnection_api",
"../api:rtc_error_matchers",
"../api:scoped_refptr",
"../api/units:time_delta",
"../p2p:connection",
"../p2p:p2p_server_utils",
"../p2p:p2p_test_utils",
"../p2p:port_allocator",
"../p2p:port_interface",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../test:test_main",
"../test:test_support",
"../test:wait_until",
"../test/time_controller:time_controller",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
]
}
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
"congestion_control_integrationtest.cc",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"data_channel_integrationtest.cc",
"data_channel_unittest.cc",
"dtmf_sender_unittest.cc",
"ice_server_parsing_unittest.cc",
"jitter_buffer_delay_unittest.cc",
"jsep_session_description_unittest.cc",
"legacy_stats_collector_unittest.cc",
"local_audio_source_unittest.cc",
"media_stream_unittest.cc",
"peer_connection_adaptation_integrationtest.cc",
"peer_connection_bundle_unittest.cc",
"peer_connection_crypto_unittest.cc",
"peer_connection_data_channel_unittest.cc",
"peer_connection_encodings_integrationtest.cc",
"peer_connection_end_to_end_unittest.cc",
"peer_connection_factory_unittest.cc",
"peer_connection_field_trial_tests.cc",
"peer_connection_header_extension_unittest.cc",
"peer_connection_histogram_unittest.cc",
"peer_connection_ice_unittest.cc",
"peer_connection_integrationtest.cc",
"peer_connection_interface_unittest.cc",
"peer_connection_jsep_unittest.cc",
"peer_connection_media_unittest.cc",
"peer_connection_rtp_unittest.cc",
"peer_connection_signaling_unittest.cc",
"peer_connection_simulcast_unittest.cc",
"peer_connection_svc_integrationtest.cc",
"proxy_unittest.cc",
"rtc_stats_collector_unittest.cc",
"rtc_stats_integrationtest.cc",
"rtc_stats_traversal_unittest.cc",
"rtp_media_utils_unittest.cc",
"rtp_parameters_conversion_unittest.cc",
"rtp_sender_receiver_unittest.cc",
"rtp_transceiver_unittest.cc",
"sctp_utils_unittest.cc",
"sdp_offer_answer_unittest.cc",
"simulcast_sdp_serializer_unittest.cc",
"test/fake_audio_capture_module_unittest.cc",
"test/test_sdp_strings.h",
"track_media_info_map_unittest.cc",
"video_rtp_track_source_unittest.cc",
"video_track_unittest.cc",
"webrtc_sdp_unittest.cc",
]
deps = [
":audio_rtp_receiver",
":audio_track",
":channel",
":channel_interface",
":dtls_srtp_transport",
":dtls_transport",
":dtmf_sender",
":enable_fake_media",
":ice_server_parsing",
":integration_test_helpers",
":jitter_buffer_delay",
":legacy_stats_collector",
":local_audio_source",
":media_protocol_names",
":media_session",
":media_stream",
":pc_test_utils",
":peer_connection",
":peer_connection_factory",
":peer_connection_internal",
":peer_connection_proxy",
":peerconnection_wrapper",
":proxy",
":rtc_stats_collector",
":rtc_stats_traversal",
":rtp_media_utils",
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_sender",
":rtp_sender_proxy",
":rtp_transceiver",
":rtp_transport_internal",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
":sctp_data_channel",
":sctp_transport",
":sctp_utils",
":sdp_utils",
":session_description",
":simulcast_description",
":simulcast_sdp_serializer",
":stream_collection",
":track_media_info_map",
":transport_stats",
":usage_pattern",
":video_rtp_receiver",
":video_rtp_track_source",
":video_track",
":video_track_source",
":webrtc_sdp",
"../api:array_view",
"../api:audio_options_api",
"../api:candidate",
"../api:create_peerconnection_factory",
"../api:dtls_transport_interface",
"../api:enable_media",
"../api:enable_media_with_defaults",
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
"../api:field_trials",
"../api:field_trials_view",
"../api:ice_transport_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:mock_async_dns_resolver",
"../api:mock_encoder_selector",
"../api:mock_packet_socket_factory",
"../api:mock_video_track",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api:packet_socket_factory",
"../api:priority",
"../api:ref_count",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/adaptation:resource_adaptation_api",
"../api/audio:audio_device",
"../api/audio:audio_mixer_api",
"../api/audio:audio_processing",
"../api/audio:audio_processing_statistics",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/audio_codecs:opus_audio_decoder_factory",
"../api/audio_codecs:opus_audio_encoder_factory",
"../api/audio_codecs/L16:audio_decoder_L16",
"../api/audio_codecs/L16:audio_encoder_L16",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/environment",
"../api/environment:environment_factory",
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:bitrate_settings",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/transport:field_trial_based_config",
"../api/transport:sctp_transport_factory_interface",
"../api/transport/rtp:rtp_source",
"../api/units:data_rate",
"../api/units:time_delta",
"../api/units:timestamp",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:encoded_image",
"../api/video:recordable_encoded_frame",
"../api/video:video_bitrate_allocator_factory",
"../api/video:video_codec_constants",
"../api/video:video_frame",
Reland "Reland "Split peer_connection_integrationtest.cc into pieces"" This reverts commit 89c40e246e39372390f0f843545d4e56aa657040. Reason for revert: Added missing INSTANTIATE Original change's description: > Revert "Reland "Split peer_connection_integrationtest.cc into pieces"" > > This reverts commit 772066bf16b125c1346a4d1b3e28c6e6f21cc1a7. > > Reason for revert: Did not catch all missing INSTANTIATE_TEST_SUITE_P > > Original change's description: > > Reland "Split peer_connection_integrationtest.cc into pieces" > > > > This reverts commit 8644f2b7632cff5e46560c2f5cf7c0dc071aa32d. > > > > Reason for revert: Fixed the bugs > > > > Original change's description: > > > Revert "Split peer_connection_integrationtest.cc into pieces" > > > > > > This reverts commit cae4656d4a7439e25160ff4d94e50949ff87cebe. > > > > > > Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc). > > > > > > Original change's description: > > > > Split peer_connection_integrationtest.cc into pieces > > > > > > > > This creates two integration tests: One for datachannel, the other > > > > for every test that is not datachannel. > > > > > > > > It separates out the common framework to a new file in pc/test. > > > > Also applies some fixes to IWYU. > > > > > > > > Bug: None > > > > Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060 > > > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#33244} > > > > > > TBR=hbos@webrtc.org,hta@webrtc.org > > > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > > > No-Try: True > > > Bug: None > > > Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283 > > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#33255} > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: None > > Change-Id: I1bb6186d7f898de82d26f4cd3d8a88014140c518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207864 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33283} > > Bug: None > Change-Id: I2b09b57c2477e52301ac30ec12ed69f2555ba7f8 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208021 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33286} Bug: None Change-Id: I6e362ac2234ae6c69dc9bbf886ee7dece8484202 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208022 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33289}
2021-02-17 09:05:31 +00:00
"../api/video:video_rtp_headers",
"../api/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../api/video_codecs:scalability_mode",
"../api/video_codecs:video_codecs_api",
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
"../call:call_interfaces",
"../call/adaptation:resource_adaptation_test_utilities",
"../common_video",
"../logging:fake_rtc_event_log",
"../media:codec",
"../media:media_channel",
"../media:media_constants",
"../media:media_engine",
"../media:rid_description",
"../media:rtc_audio_video",
"../media:rtc_data_sctp_transport_internal",
"../media:rtc_media_config",
"../media:rtc_media_tests_utils",
"../media:stream_params",
"../modules/audio_processing:mocks",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_packet_socket_factory",
"../p2p:connection_info",
"../p2p:dtls_transport_internal",
"../p2p:fake_port_allocator",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_test_utils",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:port_interface",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:byte_buffer",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:crypto_random",
"../rtc_base:digest",
"../rtc_base:event_tracer",
"../rtc_base:gunit_helpers",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:network",
"../rtc_base:network_constants",
"../rtc_base:null_socket_server",
"../rtc_base:random",
"../rtc_base:refcount",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:rtc_json",
"../rtc_base:safe_conversions",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:socket_address",
"../rtc_base:socket_server",
"../rtc_base:ssl",
"../rtc_base:ssl_adapter",
"../rtc_base:stringutils",
"../rtc_base:task_queue_for_test",
Reland "Refactor rtc_base build targets." This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 10:41:01 +01:00
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base:unique_id_generator",
"../rtc_base/containers:flat_map",
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/base64",
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
"../test:audio_codec_mocks",
"../test:rtc_expect_death",
"../test:run_loop",
"../test:test_support",
"../test:wait_until",
"../test/pc/sctp:fake_sctp_transport",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/strings:string_view",
]
# These deps are kept separately because they can't be automatically
# regenerated by gn_check_autofix tool
deps += [
":data_channel_controller_unittest",
"../test:test_main",
]
if (is_android) {
use_default_launcher = false
deps += [
":android_black_magic",
# We need to depend on this one directly, or classloads will fail for
# the voice engine BuildInfo, for instance.
"../sdk/android:libjingle_peerconnection_java",
]
shard_timeout = 900
}
}
rtc_library("data_channel_controller_unittest") {
testonly = true
sources = [ "data_channel_controller_unittest.cc" ]
deps = [
":data_channel_controller",
":pc_test_utils",
":peer_connection_internal",
":sctp_data_channel",
"../api:priority",
"../rtc_base:null_socket_server",
"../test:run_loop",
"../test:test_support",
]
}
if (is_android) {
rtc_library("android_black_magic") {
# The android code uses hacky includes to ssl code. Having this in a
# separate target enables us to keep the peerconnection unit tests clean.
testonly = true
sources = [
"test/android_test_initializer.cc",
"test/android_test_initializer.h",
]
deps = [
"../modules/utility:utility",
"../rtc_base:checks",
"../rtc_base:ssl_adapter",
"../sdk/android:internal_jni",
"../sdk/android:libjingle_peerconnection_jni",
]
}
}
rtc_library("integration_test_helpers") {
testonly = true
sources = [
"test/integration_test_helpers.cc",
"test/integration_test_helpers.h",
]
deps = [
":pc_test_utils",
":peer_connection",
":peer_connection_factory",
":peer_connection_proxy",
":session_description",
":video_track_source",
"../api:audio_options_api",
"../api:candidate",
"../api:enable_media_with_defaults",
"../api:field_trials",
"../api:field_trials_view",
"../api:ice_transport_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:mock_async_dns_resolver",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:builtin_audio_processing_builder",
"../api/crypto:options",
"../api/metronome",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/task_queue:pending_task_safety_flag",
"../api/units:time_delta",
"../api/video:video_rtp_headers",
"../logging:fake_rtc_event_log",
"../media:stream_params",
"../p2p:fake_ice_transport",
"../p2p:ice_transport_internal",
"../p2p:p2p_test_utils",
"../p2p:port",
"../p2p:port_interface",
"../rtc_base:checks",
"../rtc_base:crypto_random",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:socket_address",
"../rtc_base:socket_factory",
"../rtc_base:socket_server",
"../rtc_base:ssl_adapter",
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../system_wrappers:metrics",
"../test:test_support",
"../test:wait_until",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
rtc_library("enable_fake_media") {
testonly = true
visibility = [ ":*" ]
sources = [
"test/enable_fake_media.cc",
"test/enable_fake_media.h",
]
deps = [
":media_factory",
"../api:libjingle_peerconnection_api",
"../api/environment",
"../call:call_interfaces",
"../media:rtc_media_tests_utils",
"../rtc_base:checks",
"//third_party/abseil-cpp/absl/base:nullability",
]
}
rtc_library("pc_test_utils") {
testonly = true
sources = [
"test/fake_audio_capture_module.cc",
"test/fake_audio_capture_module.h",
"test/fake_data_channel_controller.h",
"test/fake_peer_connection_base.h",
"test/fake_peer_connection_for_stats.h",
"test/fake_periodic_video_source.h",
"test/fake_periodic_video_track_source.h",
"test/fake_rtc_certificate_generator.h",
"test/fake_video_track_renderer.h",
"test/fake_video_track_source.h",
"test/frame_generator_capturer_video_track_source.h",
"test/mock_channel_interface.h",
"test/mock_data_channel.h",
"test/mock_peer_connection_internal.h",
"test/mock_peer_connection_observers.h",
"test/mock_rtp_receiver_internal.h",
"test/mock_rtp_sender_internal.h",
"test/mock_voice_media_receive_channel_interface.h",
"test/peer_connection_test_wrapper.cc",
"test/peer_connection_test_wrapper.h",
"test/rtc_stats_obtainer.h",
"test/simulcast_layer_util.cc",
"test/simulcast_layer_util.h",
"test/test_sdp_strings.h",
]
deps = [
":channel",
":channel_interface",
":enable_fake_media",
":jitter_buffer_delay",
":libjingle_peerconnection",
":peer_connection_internal",
":rtp_receiver",
":rtp_sender",
Reland "Break out targets from pc/peerconnection build target." This reverts commit e3bf4a67c9eb24edea7bfacc0029e8fd0fa96ca5. Reason for revert: Fixed downstream projects. Original change's description: > Revert "Break out targets from pc/peerconnection build target." > > This reverts commit c9664435944268cd5753eb238bfe9494dd2eec8b. > > Reason for revert: Breaks upstream project > > Original change's description: > > Break out targets from pc/peerconnection build target. > > > > This is part of a project to make sdp_offer_answer be a separate > > compile target from peerconnection. > > This CL affects sctp_data_channel and data_channel_utils. > > > > Bug: webrtc:11995 > > Change-Id: I98244413b7cffdd0c70c56221f0692c2949e0549 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249799 > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35840} > > TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: If2a898f6e573ce347b9858fe8bf29a5a2211bff0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:11995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249946 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35843} # Not skipping CQ checks because this is a reland. Bug: webrtc:11995, webrtc:13634 Change-Id: I751e089da01c682c37953fedac542a67ce77ac9b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249992 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35923}
2022-02-07 02:59:36 +00:00
":sctp_data_channel",
":session_description",
":simulcast_description",
":stream_collection",
":video_track_source",
"../api:audio_options_api",
"../api:call_api",
"../api:create_frame_generator",
"../api:create_peerconnection_factory",
"../api:field_trials_view",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:media_stream_interface",
"../api:priority",
"../api:rtc_error",
"../api:rtc_error_matchers",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_device",
"../api/audio:audio_mixer_api",
"../api/audio:audio_processing",
"../api/audio_codecs:audio_codecs_api",
"../api/environment",
"../api/environment:environment_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:resolution",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
"../api/video_codecs:video_decoder_factory_template",
"../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"../api/video_codecs:video_encoder_factory_template",
"../api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
"../call:call_interfaces",
"../media:media_channel",
"../media:media_channel_impl",
"../media:rtc_media",
"../media:rtc_media_tests_utils",
"../media:rtc_simulcast_encoder_adapter",
"../media:video_broadcaster",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_packet_socket_factory",
"../p2p:connection",
"../p2p:fake_port_allocator",
"../p2p:p2p_test_utils",
"../p2p:port_allocator",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:socket_server",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base:weak_ptr",
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:repeating_task",
"../rtc_base/third_party/sigslot",
"../test:frame_generator_capturer",
"../test:scoped_key_value_config",
"../test:test_support",
"../test:wait_until",
"//testing/gmock",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
]
}
svc_tests_resources = [
"../resources/difficult_photo_1850_1110.yuv",
"../resources/photo_1850_1110.yuv",
"../resources/presentation_1850_1110.yuv",
"../resources/web_screenshot_1850_1110.yuv",
]
if (is_ios) {
bundle_data("svc_tests_bundle_data") {
testonly = true
sources = svc_tests_resources
outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
}
}
rtc_test("svc_tests") {
sources = [ "test/svc_e2e_tests.cc" ]
data = svc_tests_resources
deps = [
"../api:create_network_emulation_manager",
"../api:create_peer_connection_quality_test_frame_generator",
"../api:create_peerconnection_quality_test_fixture",
"../api:frame_generator_api",
"../api:media_stream_interface",
"../api:network_emulation_manager_api",
"../api:peer_connection_quality_test_fixture_api",
"../api:rtc_stats_api",
"../api:simulated_network_api",
"../api:time_controller",
"../api/test/metrics:global_metrics_logger_and_exporter",
"../api/test/pclf:media_configuration",
"../api/test/pclf:media_quality_test_params",
"../api/test/pclf:peer_configurer",
"../api/video_codecs:video_codecs_api",
"../media:media_constants",
"../modules/video_coding:webrtc_vp9",
"../modules/video_coding/svc:scalability_mode_util",
"../rtc_base/containers:flat_map",
"../system_wrappers:field_trial",
"../test:field_trial",
"../test:fileutils",
"../test:test_main",
"../test:test_support",
"../test/network:simulated_network",
"../test/pc/e2e:network_quality_metrics_reporter",
Reland "[DVQA] Create separate BUILD.gn file for video analyzer" This reverts commit 76793c300fdd87fa8fd8be3dd2e5faf8c1916e96. Reason for revert: Can't cleanly revert the old one. A forward fix will be provided. Original change's description: > Revert "[DVQA] Create separate BUILD.gn file for video analyzer" > > This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c. > > Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview > > > Original change's description: > > [DVQA] Create separate BUILD.gn file for video analyzer > > > > Bug: None > > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17 > > No-try: True > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141 > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38662} > > Bug: None > Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000 > Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38672} Bug: None Change-Id: I74506eaa6a1060bf87e651881c86b4f576f447ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284020 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38676}
2022-11-18 09:47:40 +00:00
"../test/pc/e2e/analyzer/video:default_video_quality_analyzer",
]
if (is_ios) {
deps += [ ":svc_tests_bundle_data" ]
}
}
}