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Break up rtc_event_log_api to solve circular dependencies.

Browse Source 
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}
This commit is contained in:
Qingsi Wang
2018-01-24 16:23:20 -08:00
committed by Commit Bot
parent 98bf720f97
commit 001546da95
18 changed files with 174 additions and 44 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
audio/BUILD.gn
View File

@ -58,6 +58,7 @@ rtc_static_library("audio") {
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding",

6
call/BUILD.gn
View File

@ -142,8 +142,11 @@ rtc_static_library("call") {
"../api:optional",
"../api:transport_api",
"../audio",
"../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../logging:rtc_event_rtp_rtcp",
"../logging:rtc_event_video",
"../logging:rtc_stream_config",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
@ -214,6 +217,7 @@ if (rtc_include_tests) {
"../api/audio_codecs:builtin_audio_decoder_factory",
"../audio:audio",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/audio_mixer:audio_mixer_impl",

175
logging/BUILD.gn
View File

@ -17,8 +17,16 @@ 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" ]
}
@ -26,9 +34,45 @@ group("logging") {
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 +81,17 @@ 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 +102,16 @@ 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 +120,53 @@ 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 +181,46 @@ 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",
]
}
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 (rtc_enable_protobuf) {
proto_library("rtc_event_log_proto") {
sources = [
@ -148,9 +243,11 @@ 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 +281,17 @@ 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 +316,6 @@ 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 +341,6 @@ 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 +368,6 @@ 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",

4
logging/rtc_event_log/rtc_event_log.h
View File

@ -57,9 +57,7 @@ class RtcEventLog {
class RtcEventLogNullImpl : public RtcEventLog {
public:
bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) override {
return false;
}
int64_t output_period_ms) override;
void StopLogging() override {}
void Log(std::unique_ptr<RtcEvent> event) override {}
};

6
logging/rtc_event_log/rtc_event_log.cc → logging/rtc_event_log/rtc_event_log_impl.cc
View File

@ -377,4 +377,10 @@ std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
}
bool RtcEventLogNullImpl::StartLogging(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return false;
}
} // namespace webrtc

1
media/BUILD.gn
View File

@ -608,6 +608,7 @@ if (rtc_include_tests) {
"../call:call_interfaces",
"../common_video:common_video",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../modules/audio_device:mock_audio_device",
"../modules/audio_processing:audio_processing",
"../modules/video_coding:simulcast_test_utility",

2
modules/audio_coding/BUILD.gn
View File

@ -955,6 +955,7 @@ rtc_static_library("audio_network_adaptor") {
"../../api:optional",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../logging:rtc_event_audio",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",
@ -2171,6 +2172,7 @@ if (rtc_include_tests) {
"../../common_audio",
"../../common_audio:mock_common_audio",
"../../logging:mocks",
"../../logging:rtc_event_audio",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:protobuf_utils",

2
modules/bitrate_controller/BUILD.gn
View File

@ -34,6 +34,7 @@ rtc_static_library("bitrate_controller") {
deps = [
"..:module_api",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -70,6 +71,7 @@ if (rtc_include_tests) {
deps = [
":bitrate_controller",
"../../logging:mocks",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../test:field_trial",
"../../test:test_support",

2
modules/congestion_controller/BUILD.gn
View File

@ -102,6 +102,7 @@ rtc_source_set("estimators") {
deps = [
"../../api:optional",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@ -122,6 +123,7 @@ rtc_source_set("delay_based_bwe") {
deps = [
":estimators",
"../../:typedefs",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",

2
modules/pacing/BUILD.gn
View File

@ -37,7 +37,9 @@ rtc_static_library("pacing") {
"../../:typedefs",
"../../:webrtc_common",
"../../api:optional",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_pacing",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base/experiments:alr_experiment",

2
modules/rtp_rtcp/BUILD.gn
View File

@ -201,7 +201,9 @@ rtc_static_library("rtp_rtcp") {
"../../api:transport_api",
"../../api/audio_codecs:audio_codecs_api",
"../../common_video",
"../../logging:rtc_event_audio",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_rtp_rtcp",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:gtest_prod",

1
ortc/BUILD.gn
View File

@ -39,6 +39,7 @@ rtc_static_library("ortc") {
"../call:call_interfaces",
"../call:rtp_sender",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media",
"../media:rtc_media_base",

5
pc/BUILD.gn
View File

@ -188,6 +188,7 @@ rtc_static_library("peerconnection") {
"../call:call_interfaces",
"../common_video:common_video",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../media:rtc_data",
"../media:rtc_media_base",
"../p2p:rtc_p2p",
@ -221,6 +222,7 @@ rtc_static_library("create_pc_factory") {
"../call",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_audio_video",
"../media:rtc_media_base",
"../modules/audio_device:audio_device",
@ -481,7 +483,8 @@ if (rtc_include_tests) {
"../api/audio_codecs/L16:audio_encoder_L16",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../logging:rtc_event_log_impl_base",
"../logging:rtc_event_log_impl_output",
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",

3
rtc_tools/BUILD.gn
View File

@ -224,8 +224,9 @@ if (!build_with_chromium) {
"../call:call_interfaces",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../logging:rtc_event_log_impl_base",
"../logging:rtc_event_log_parser",
"../logging:rtc_stream_config",
"../modules:module_api",
"../modules/audio_coding:ana_debug_dump_proto",
"../modules/audio_coding:audio_network_adaptor",

2
sdk/android/BUILD.gn
View File

@ -438,6 +438,8 @@ rtc_static_library("peerconnection_jni") {
"../../api:libjingle_peerconnection_api",
"../../api:peerconnection_and_implicit_call_api",
"../../api/video_codecs:video_codecs_api",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_log_impl_base",
"../../media:rtc_data",
"../../media:rtc_media_base",
"../../modules/audio_device:audio_device",

1
test/BUILD.gn
View File

@ -613,6 +613,7 @@ rtc_source_set("test_common") {
"../call:video_stream_api",
"../common_video",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_base",
"../media:rtc_media_base",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",

2
test/fuzzers/BUILD.gn
View File

@ -231,7 +231,7 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
]
deps = [
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_log_impl",
"../../logging:rtc_event_log_impl_base",
"../../modules/congestion_controller",
"../../modules/pacing",
"../../modules/remote_bitrate_estimator:remote_bitrate_estimator",

1
video/BUILD.gn
View File

@ -112,6 +112,7 @@ if (rtc_include_tests) {
]
deps = [
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl_output",
"../media:rtc_audio_video",
"../media:rtc_internal_video_codecs",
"../modules/audio_mixer:audio_mixer_impl",