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")
}
group("pc") {
deps = [ ":rtc_pc" ]
}
rtc_library("proxy") {
visibility = [ ":*" ]
sources = [
"proxy.cc",
"proxy.h",
]
deps = [
"../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_parameters",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/crypto:options",
"../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:dtls_transport_internal",
"../p2p:rtc_p2p",
"../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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("channel_interface") {
visibility = [ ":*" ]
sources = [ "channel_interface.h" ]
deps = [
":rtp_transport_internal",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
"../media:media_channel",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
rtc_source_set("dtls_srtp_transport") {
visibility = [ ":*" ]
sources = [
"dtls_srtp_transport.cc",
"dtls_srtp_transport.h",
]
deps = [
":srtp_transport",
"../api:dtls_transport_interface",
"../api:libjingle_peerconnection_api",
"../api:rtc_error",
"../p2p:dtls_transport_internal",
"../p2p:packet_transport_internal",
"../p2p:rtc_p2p",
"../rtc_base:buffer",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:ssl",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_source_set("dtls_transport") {
visibility = [ ":*" ]
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",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
"../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:rtc_p2p",
"../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:stringutils",
"../rtc_base:threading",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base/system:no_unique_address",
]
absl_deps = []
}
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",
"../media:rtc_data_sctp_transport_internal",
"../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:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:callback_list",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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 = [
":jsep_transport",
":media_protocol_names",
":rtp_media_utils",
":session_description",
":simulcast_description",
":used_ids",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:rtp_parameters",
"../api:rtp_transceiver_direction",
"../api/crypto:options",
"../media:codec",
"../media:media_constants",
"../media:media_engine",
"../media:rid_description",
"../media:rtc_data_sctp_transport_internal",
"../media:rtc_sdp_video_format_utils",
"../media:stream_params",
"../p2p:ice_credentials_iterator",
"../p2p:p2p_constants",
"../p2p:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:ssl",
"../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",
"../rtc_base/third_party/base64",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../api:libjingle_peerconnection_api",
]
}
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/task_queue:pending_task_safety_flag",
"../api/units:timestamp",
"../call:rtp_receiver",
"../call:video_stream_api",
"../media:rtp_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:packet_transport_internal",
"../p2p:rtc_p2p",
"../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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../p2p:rtc_p2p",
"../rtc_base:callback_list",
"../rtc_base:network_route",
"../rtc_base:ssl",
]
}
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:scoped_refptr",
"../api:sequence_checker",
"../api/transport:datagram_transport_interface",
"../media:rtc_data_sctp_transport_internal",
"../p2p:dtls_transport_internal",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:threading",
"../rtc_base/third_party/sigslot",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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:byte_order",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../system_wrappers:metrics",
]
absl_deps = [
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:strings",
]
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",
"../media:rtp_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:packet_transport_internal",
"../p2p:rtc_p2p",
"../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",
"../rtc_base:zero_memory",
"../rtc_base/third_party/base64",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../p2p:rtc_p2p",
"../rtc_base:ssl",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:socket_address",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory:memory",
"//third_party/abseil-cpp/absl/strings",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
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",
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",
]
Reland "[DataChannel] Send and receive packets on the network thread." This reverts commit 7f16fcda0fd5bb625584b71311dd37b54c096136. Reason for reland: Re-landing after addressing issues in downstream code and hardening the ObserverAdapter from situations where attempted usage of data channel proxies could occur after shutting down the peer connection and terminating the network thread. Original change's description: > Revert "[DataChannel] Send and receive packets on the network thread." > > This reverts commit fe53fec24e02d2d644220f913c3f9ae596bbb2d9. > > Reason for revert: Speculative revert, may be breaking downstream project > > Original change's description: > > [DataChannel] Send and receive packets on the network thread. > > > > This updates sctp channels, including work that happens between the > > data channel controller and the transport, to run on the network > > thread. Previously all network traffic related to data channels was > > routed through the signaling thread before going to either the network > > thread or the caller's thread (e.g. js thread in chrome). Now the > > calls can go straight from the network thread to the JS thread with > > enabling a special flag on the observer (see below) and similarly > > calls to send data, involve 2 threads instead of 3. > > > > * Custom data channel observer adapter implementation that > > maintains compatibility with existing observer implementations in > > that notifications are delivered on the signaling thread. > > The adapter can be explicitly disabled for implementations that > > want to optimize the callback path and promise to not block the > > network thread. > > * Remove the signaling thread copy of data channels in the controller. > > * Remove several PostTask operations that were needed to keep things > > in sync (but the need has gone away). > > * Update tests for the controller to consistently call > > TeardownDataChannelTransport_n to match with production. > > * Update stats collectors (current and legacy) to fetch the data > > channel stats on the network thread where they're maintained. > > * Remove the AsyncChannelCloseTeardown test since the async teardown > > step has gone away. > > * Remove `sid_s` in the channel code since we only need the network > > state now. > > * For the custom observer support (with and without data adapter) and > > maintain compatibility with existing implementations, added a new > > proxy macro that allows an implementation to selectively provide > > its own implementation without being proxied. This is used for > > registering/unregistering a data channel observer. > > * Update the data channel proxy to map most methods to the network > > thread, avoiding the interim jump to the signaling thread. > > * Update a plethora of thread checkers from signaling to network. > > > > Bug: webrtc:11547 > > Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142 > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39760} > > Bug: webrtc:11547 > Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341 > Auto-Submit: Andrey Logvin <landrey@webrtc.org> > Owners-Override: Andrey Logvin <landrey@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@{#39764} Bug: webrtc:11547 Change-Id: I47dfa7e7168be0cd2faab4f8f3ebf110c3728af5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300360 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39786}
2023-04-06 21:21:45 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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("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",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:macromagic",
"../rtc_base:network",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:socket_factory",
"../rtc_base:socket_server",
"../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: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: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",
"../rtc_base:threading",
"../rtc_base:weak_ptr",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../call:call_interfaces",
"../modules/audio_device",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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/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/audio_processing:audio_processing_statistics",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:port",
"../p2p:rtc_p2p",
"../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:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/containers:flat_set",
"../rtc_base/synchronization:mutex",
]
absl_deps = [
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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_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",
"../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",
"../media:codec",
"../media:media_channel",
"../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:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_operations_chain",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory:memory",
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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: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/crypto:options",
"../api/environment",
"../api/rtc_event_log",
"../api/task_queue:pending_task_safety_flag",
"../api/transport:bitrate_settings",
"../api/transport:datagram_transport_interface",
"../api/transport:enums",
"../api/video:video_codec_constants",
"../call:call_interfaces",
"../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:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../rtc_base:event_tracer",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:network",
"../rtc_base:network_constants",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:threading",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
"../system_wrappers:metrics",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("sdp_utils") {
sources = [
"sdp_utils.cc",
"sdp_utils.h",
]
deps = [
":session_description",
"../api:libjingle_peerconnection_api",
"../p2p:rtc_p2p",
"../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_codecs:audio_codecs_api",
"../api/video:video_rtp_headers",
"../call:call_interfaces",
"../media:media_channel",
"../modules/audio_processing:audio_processing_statistics",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:port",
"../p2p:rtc_p2p",
"../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:stringutils",
"../rtc_base:threading",
"../rtc_base:timeutils",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_info",
"../rtc_base:checks",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:network_constants",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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:libjingle_peerconnection_api",
"../api:rtc_error",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/task_queue",
"../p2p:rtc_p2p",
"../p2p:transport_description",
"../p2p:transport_description_factory",
"../rtc_base:checks",
"../rtc_base:logging",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:ssl",
"../rtc_base:stringutils",
"../rtc_base:unique_id_generator",
"../rtc_base:weak_ptr",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/functional:any_invocable",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../p2p:rtc_p2p",
"../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",
]
absl_deps = [ "//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",
"../p2p:rtc_p2p",
"../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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
}
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",
"../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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../media:media_channel",
"../rtc_base:checks",
"../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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
]
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
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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: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",
"../media:audio_source",
"../media:media_channel",
"../media:media_engine",
"../rtc_base:checks",
"../rtc_base:event_tracer",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:ssl",
"../rtc_base:threading",
"../rtc_base/synchronization:mutex",
"../rtc_base/third_party/sigslot",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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",
"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",
]
include_dirs = [ "//third_party/libsrtp/srtp" ]
if (is_win) {
libs = [ "strmiids.lib" ]
}
deps = [
":audio_rtp_receiver",
":channel",
":dtls_srtp_transport",
":dtls_transport",
":ice_transport",
":jsep_transport",
":jsep_transport_controller",
":libjingle_peerconnection",
":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",
":srtp_session",
":srtp_transport",
":used_ids",
":video_rtp_receiver",
"../api:array_view",
"../api:audio_options_api",
"../api:candidate",
"../api:dtls_transport_interface",
"../api:ice_transport_factory",
"../api:libjingle_peerconnection_api",
"../api:make_ref_counted",
"../api:make_ref_counted",
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../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/video:builtin_video_bitrate_allocator_factory",
"../api/video:recordable_encoded_frame",
"../api/video/test:mock_recordable_encoded_frame",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../media:codec",
"../media:media_channel",
"../media:media_constants",
"../media:rid_description",
"../media:rtc_data_sctp_transport_internal",
"../media:rtc_media_tests_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:candidate_pair_interface",
"../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:rtc_p2p",
"../p2p:transport_description",
"../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:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:net_helper",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../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:received_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",
]
absl_deps = [
"//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/types:optional",
]
if (rtc_build_libsrtp) {
deps += [ "//third_party/libsrtp" ]
}
if (is_android) {
use_default_launcher = false
deps += [
"//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
"//testing/android/native_test:native_test_java",
"//testing/android/native_test:native_test_support",
]
}
}
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:create_peerconnection_factory",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:scoped_refptr",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/test/metrics:global_metrics_logger_and_exporter",
"../api/test/metrics:metric",
"../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",
"../media:rtc_media_tests_utils",
"../modules/audio_device:audio_device_api",
"../modules/audio_processing:api",
"../p2p:basic_port_allocator",
"../p2p:connection",
"../p2p:p2p_test_utils",
"../p2p:port_allocator",
"../p2p:port_interface",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../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:ssl",
"../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",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("peerconnection_wrapper") {
testonly = true
sources = [
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
]
deps = [
":pc_test_utils",
":sdp_utils",
"../api:function_view",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:scoped_refptr",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
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:scoped_refptr",
"../api/units:time_delta",
"../p2p:connection",
"../p2p:p2p_server_utils",
"../p2p:p2p_test_utils",
"../p2p:port_allocator",
"../p2p:port_interface",
"../p2p:rtc_p2p",
"../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/time_controller:time_controller",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_test("peerconnection_unittests") {
testonly = true
sources = [
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",
"peer_connection_wrapper.cc",
"peer_connection_wrapper.h",
"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",
":data_channel_controller_unittest",
":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",
":peer_connection",
":peer_connection_factory",
":peer_connection_proxy",
":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:dtmf_sender_interface",
"../api:enable_media",
"../api:enable_media_with_defaults",
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
"../api:field_trials_view",
"../api:function_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:rtc_error",
"../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/adaptation:resource_adaptation_api",
"../api/audio:audio_mixer_api",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../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/transport:datagram_transport_interface",
"../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:scalability_mode",
"../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_data_sctp_transport_internal",
"../media:rtc_media_config",
"../media:stream_params",
"../modules/audio_device:audio_device_api",
"../modules/audio_processing:audio_processing_statistics",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_port_allocator",
"../p2p:connection",
"../p2p:connection_info",
"../p2p:dtls_transport_internal",
"../p2p:fake_port_allocator",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
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
"../p2p:p2p_server_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: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:mdns_responder_interface",
"../rtc_base:net_helper",
"../rtc_base:network",
"../rtc_base:network_constants",
"../rtc_base:null_socket_server",
"../rtc_base:refcount",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:rtc_json",
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:ssl",
"../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/synchronization:mutex",
"../rtc_base/third_party/base64",
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
"../test:field_trial",
"../test:rtc_expect_death",
"../test:run_loop",
"../test:scoped_key_value_config",
"../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/types:optional",
]
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",
"//sdk/android:native_test_jni_onload",
]
shard_timeout = 900
}
deps += [
":libjingle_peerconnection",
":pc_test_utils",
":rtc_pc",
"../api:rtc_event_log_output_file",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../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/video_codecs:builtin_video_decoder_factory",
"../api/video_codecs:builtin_video_encoder_factory",
"../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:rtc_audio_video",
"../media:rtc_media_tests_utils",
"../modules/audio_processing",
Reland "Isolating APM API build target: making :api an actual target." This reverts commit 61c6e5643e7ea058e653956980a90e033249c055. Reason for revert: downstream projects prepared for this change Original change's description: > Revert "Isolating APM API build target: making :api an actual target." > > This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff. > > Reason for revert: breaking downstream > > Original change's description: > > Isolating APM API build target: making :api an actual target. > > > > This CL is part of a refactoring work to unblock other CLs > > that would generate a circular dependency when including > > modules/audio_processing. It will also allow to easily move > > the APM interface part under //api. > > > > More in detail, this change moves the APM interface files from > > the build target modules/audio_processing to > > modules/audio_processing:api. It also adds :api as dependency > > where needed. > > > > Bug: webrtc:9535 > > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd > > Reviewed-on: https://webrtc-review.googlesource.com/c/109501 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#25539} > > TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org > > Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9535 > Reviewed-on: https://webrtc-review.googlesource.com/c/109820 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25540} TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9535 Reviewed-on: https://webrtc-review.googlesource.com/c/109884 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:29:54 +00:00
"../modules/audio_processing:api",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../rtc_base:safe_conversions",
"../test:audio_codec_mocks",
"../test:test_main",
"../test:test_support",
]
}
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",
"../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 = [
"../rtc_base:ssl",
"../sdk/android:internal_jni",
"../sdk/android:libjingle_peerconnection_jni",
"//modules/utility:utility",
"//rtc_base:checks",
"//testing/android/native_test:native_test_support",
]
}
}
rtc_library("integration_test_helpers") {
testonly = true
sources = [
"test/integration_test_helpers.cc",
"test/integration_test_helpers.h",
]
deps = [
":audio_rtp_receiver",
":audio_track",
":dtmf_sender",
":jitter_buffer_delay",
":local_audio_source",
":media_session",
":media_stream",
":pc_test_utils",
":peer_connection",
":peer_connection_factory",
":peer_connection_proxy",
":remote_audio_source",
":rtp_media_utils",
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_sender",
":rtp_transceiver",
":session_description",
":usage_pattern",
":video_rtp_receiver",
":video_rtp_track_source",
":video_track",
":video_track_source",
"../api:array_view",
"../api:audio_options_api",
"../api:candidate",
"../api:create_peerconnection_factory",
"../api:enable_media_with_defaults",
"../api:fake_frame_decryptor",
"../api:fake_frame_encryptor",
"../api:field_trials_view",
"../api:function_view",
"../api:ice_transport_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:mock_async_dns_resolver",
"../api:mock_rtp",
"../api:packet_socket_factory",
"../api:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:rtp_sender_interface",
"../api:rtp_transceiver_direction",
"../api:scoped_refptr",
"../api/audio:audio_mixer_api",
"../api/crypto:frame_decryptor_interface",
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/rtc_event_log",
"../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/transport:field_trial_based_config",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",
"../call:call_interfaces",
"../call/adaptation:resource_adaptation_test_utilities",
"../logging:fake_rtc_event_log",
"../media:media_engine",
"../media:rtc_media_config",
"../media:rtc_media_tests_utils",
"../media:stream_params",
"../modules/audio_device:audio_device_api",
"../modules/audio_processing:api",
"../modules/audio_processing:audio_processing_statistics",
"../modules/audio_processing:audioproc_test_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:basic_port_allocator",
"../p2p:connection",
"../p2p:fake_ice_transport",
"../p2p:fake_port_allocator",
"../p2p:ice_transport_internal",
"../p2p:p2p_constants",
"../p2p:p2p_server_utils",
"../p2p:p2p_test_utils",
"../p2p:port",
"../p2p:port_allocator",
"../p2p:port_interface",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:ip_address",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:mdns_responder_interface",
"../rtc_base:null_socket_server",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_certificate_generator",
"../rtc_base:rtc_event",
"../rtc_base:rtc_json",
"../rtc_base:safe_conversions",
"../rtc_base:socket_address",
"../rtc_base:ssl",
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
"../rtc_base:timeutils",
"../rtc_base/synchronization:mutex",
"../rtc_base/task_utils:repeating_task",
"../rtc_base/third_party/base64",
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
"../test:explicit_key_value_config",
"../test:fileutils",
"../test:rtp_test_utils",
"../test:scoped_key_value_config",
"../test:test_support",
"../test/pc/sctp:fake_sctp_transport",
"../test/time_controller",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
]
absl_deps = [ "//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:rtc_error",
"../api:rtc_stats_api",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
"../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/audio_processing:api",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:connection",
"../p2p:fake_port_allocator",
"../p2p:p2p_test_utils",
"../p2p:port_allocator",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:logging",
"../rtc_base:macromagic",
"../rtc_base:rtc_certificate_generator",
"../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",
]
absl_deps = [
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
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",
"../call:simulated_network",
"../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/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" ]
}
}
}