Reland "Break up rtc_event_log_api to solve circular dependencies."

This is a reland of 001546da953275c7a39eb220592b440c9b47d756
Original change's description:
> Break up rtc_event_log_api to solve circular dependencies.
>
> The original rtc_event_log_api is refactored to a pure API target plus
> multiple targets coupled with WebRTC implementations.
>
> Bug: None
> Change-Id: Iab9eee3f7bf4228c52d94a5f26fc39bb99b5033f
> Reviewed-on: https://webrtc-review.googlesource.com/43247
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#21811}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org

Bug: None
Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647
Reviewed-on: https://webrtc-review.googlesource.com/46900
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21862}
This commit is contained in:
Qingsi Wang
2018-02-01 11:04:46 -08:00
committed by Commit Bot
parent a9c94d5b12
commit 970b088878
17 changed files with 187 additions and 41 deletions

View File

@ -17,18 +17,72 @@ if (is_android) {
group("logging") {
deps = [
":rtc_event_log_impl",
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log_impl_base",
":rtc_event_log_impl_encoder",
":rtc_event_log_impl_output",
":rtc_event_pacing",
":rtc_event_rtp_rtcp",
":rtc_event_video",
]
if (rtc_enable_protobuf) {
deps += [ ":rtc_event_log_parser" ]
}
}
# TODO(mbonadei): Remove when the following CL is landed:
# https://webrtc-review.googlesource.com/c/src/+/46900.
# This is only a backwards compatible target.
rtc_source_set("rtc_event_log_impl") {
visibility = [ "*" ]
deps = [
":rtc_event_log_impl_base",
]
}
rtc_source_set("rtc_event_log_api") {
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder.h",
"rtc_event_log/events/rtc_event.h",
"rtc_event_log/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
]
deps = [
"../api:libjingle_logging_api",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("rtc_stream_config") {
sources = [
"rtc_event_log/rtc_stream_config.cc",
"rtc_event_log/rtc_stream_config.h",
]
deps = [
":rtc_event_log_api",
"..:webrtc_common",
"../api:libjingle_peerconnection_api",
]
}
rtc_source_set("rtc_event_pacing") {
sources = [
"rtc_event_log/events/rtc_event_alr_state.cc",
"rtc_event_log/events/rtc_event_alr_state.h",
]
deps = [
":rtc_event_log_api",
"../:typedefs",
]
}
rtc_source_set("rtc_event_audio") {
sources = [
"rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
"rtc_event_log/events/rtc_event_audio_network_adaptation.h",
"rtc_event_log/events/rtc_event_audio_playout.cc",
@ -37,6 +91,17 @@ rtc_source_set("rtc_event_log_api") {
"rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
"rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
"rtc_event_log/events/rtc_event_audio_send_stream_config.h",
]
deps = [
":rtc_event_log_api",
":rtc_stream_config",
"../modules/audio_coding:audio_network_adaptor_config",
]
}
rtc_source_set("rtc_event_bwe") {
sources = [
"rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
"rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
@ -47,6 +112,16 @@ rtc_source_set("rtc_event_log_api") {
"rtc_event_log/events/rtc_event_probe_result_failure.h",
"rtc_event_log/events/rtc_event_probe_result_success.cc",
"rtc_event_log/events/rtc_event_probe_result_success.h",
]
deps = [
":rtc_event_log_api",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
]
}
rtc_source_set("rtc_event_rtp_rtcp") {
sources = [
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
@ -55,63 +130,53 @@ rtc_source_set("rtc_event_log_api") {
"rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
"rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
]
deps = [
":rtc_event_log_api",
"../api:array_view",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
]
}
rtc_source_set("rtc_event_video") {
sources = [
"rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
"rtc_event_log/events/rtc_event_video_send_stream_config.h",
"rtc_event_log/output/rtc_event_log_output_file.cc",
"rtc_event_log/output/rtc_event_log_output_file.h",
"rtc_event_log/rtc_event_log.h",
"rtc_event_log/rtc_event_log_factory_interface.h",
"rtc_event_log/rtc_stream_config.cc",
"rtc_event_log/rtc_stream_config.h",
]
deps = [
"..:webrtc_common",
"../:typedefs",
"../api:array_view",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor_config",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
":rtc_event_log_api",
":rtc_stream_config",
]
# TODO(eladalon): Remove this.
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("rtc_event_log_impl") {
rtc_static_library("rtc_event_log_impl_encoder") {
visibility = [ "*" ]
sources = [
"rtc_event_log/encoder/rtc_event_log_encoder.h",
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
"rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
"rtc_event_log/rtc_event_log.cc",
"rtc_event_log/rtc_event_log_factory.cc",
"rtc_event_log/rtc_event_log_factory.h",
]
defines = []
deps = [
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log_api",
"..:webrtc_common",
":rtc_event_log_impl_output",
":rtc_event_pacing",
":rtc_event_rtp_rtcp",
":rtc_event_video",
":rtc_stream_config",
"../modules/audio_coding:audio_network_adaptor",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:checks",
"../rtc_base:protobuf_utils",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:sequenced_task_checker",
]
if (rtc_enable_protobuf) {
@ -126,6 +191,56 @@ rtc_static_library("rtc_event_log_impl") {
}
}
rtc_source_set("rtc_event_log_impl_output") {
sources = [
"rtc_event_log/output/rtc_event_log_output_file.cc",
"rtc_event_log/output/rtc_event_log_output_file.h",
]
deps = [
":rtc_event_log_api",
"../api:libjingle_logging_api",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_static_library("rtc_event_log_impl_base") {
visibility = [ "*" ]
sources = [
"rtc_event_log/rtc_event_log_factory.cc",
"rtc_event_log/rtc_event_log_factory.h",
"rtc_event_log/rtc_event_log_impl.cc",
]
defines = []
deps = [
":rtc_event_log_api",
":rtc_event_log_impl_encoder",
":rtc_event_log_impl_output",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue_api",
"../rtc_base:sequenced_task_checker",
]
if (rtc_enable_protobuf) {
defines += [ "ENABLE_RTC_EVENT_LOG" ]
deps += [ ":rtc_event_log_proto" ]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_enable_protobuf) {
proto_library("rtc_event_log_proto") {
sources = [
@ -148,9 +263,11 @@ if (rtc_enable_protobuf) {
]
deps = [
":rtc_event_bwe",
":rtc_event_log2_proto",
":rtc_event_log_api",
":rtc_event_log_proto",
":rtc_stream_config",
"..:webrtc_common",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor",
@ -184,10 +301,17 @@ if (rtc_enable_protobuf) {
"rtc_event_log/rtc_event_log_unittest_helper.h",
]
deps = [
":rtc_event_audio",
":rtc_event_bwe",
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_impl_base",
":rtc_event_log_impl_encoder",
":rtc_event_log_impl_output",
":rtc_event_log_parser",
":rtc_event_log_proto",
":rtc_event_rtp_rtcp",
":rtc_event_video",
":rtc_stream_config",
"../api:libjingle_peerconnection_api",
"../call",
"../call:call_interfaces",
@ -212,7 +336,6 @@ if (rtc_enable_protobuf) {
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
@ -238,7 +361,6 @@ if (rtc_enable_protobuf) {
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_parser",
"../:webrtc_common",
"../call:video_stream_api",
@ -266,7 +388,6 @@ if (rtc_enable_protobuf) {
]
deps = [
":rtc_event_log_api",
":rtc_event_log_impl",
":rtc_event_log_proto",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",