Revert "Break up rtc_event_log_api to solve circular dependencies."
This reverts commit 001546da953275c7a39eb220592b440c9b47d756.
Reason for revert: breaks downstream projects.
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=phoglund@webrtc.org,deadbeef@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com
Change-Id: I82540eac176c4abfb7e50dc51671585b32a1bace
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/46581
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21823}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 84cf6bc..6a359aa 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -58,7 +58,6 @@
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
- "../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
"../modules:module_api",
"../modules/audio_coding",
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 084148b..252c5b2 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -142,11 +142,8 @@
"../api:optional",
"../api:transport_api",
"../audio",
- "../logging:rtc_event_audio",
"../logging:rtc_event_log_api",
- "../logging:rtc_event_rtp_rtcp",
- "../logging:rtc_event_video",
- "../logging:rtc_stream_config",
+ "../logging:rtc_event_log_impl",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
@@ -217,7 +214,6 @@
"../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",
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index edfef15..3f14846 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -17,16 +17,8 @@
group("logging") {
deps = [
- ":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",
+ ":rtc_event_log_impl",
]
-
if (rtc_enable_protobuf) {
deps += [ ":rtc_event_log_parser" ]
}
@@ -34,45 +26,9 @@
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",
@@ -81,17 +37,6 @@
"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",
@@ -102,16 +47,6 @@
"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",
@@ -120,53 +55,63 @@
"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 = [
- ":rtc_event_log_api",
- ":rtc_stream_config",
+ "..: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",
]
+
+ # 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_encoder") {
+rtc_static_library("rtc_event_log_impl") {
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",
- ":rtc_event_log_impl_output",
- ":rtc_event_pacing",
- ":rtc_event_rtp_rtcp",
- ":rtc_event_video",
- ":rtc_stream_config",
+ "..:webrtc_common",
"../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) {
@@ -181,46 +126,6 @@
}
}
-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 = [
@@ -243,11 +148,9 @@
]
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",
@@ -281,17 +184,10 @@
"rtc_event_log/rtc_event_log_unittest_helper.h",
]
deps = [
- ":rtc_event_audio",
- ":rtc_event_bwe",
":rtc_event_log_api",
- ":rtc_event_log_impl_base",
- ":rtc_event_log_impl_encoder",
- ":rtc_event_log_impl_output",
+ ":rtc_event_log_impl",
":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",
@@ -316,6 +212,7 @@
]
deps = [
":rtc_event_log_api",
+ ":rtc_event_log_impl",
":rtc_event_log_parser",
"../modules/rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
@@ -341,6 +238,7 @@
]
deps = [
":rtc_event_log_api",
+ ":rtc_event_log_impl",
":rtc_event_log_parser",
"../:webrtc_common",
"../call:video_stream_api",
@@ -368,6 +266,7 @@
]
deps = [
":rtc_event_log_api",
+ ":rtc_event_log_impl",
":rtc_event_log_proto",
"../rtc_base:checks",
"../rtc_base:rtc_base_approved",
diff --git a/logging/rtc_event_log/rtc_event_log_impl.cc b/logging/rtc_event_log/rtc_event_log.cc
similarity index 98%
rename from logging/rtc_event_log/rtc_event_log_impl.cc
rename to logging/rtc_event_log/rtc_event_log.cc
index de9aae9..2173590 100644
--- a/logging/rtc_event_log/rtc_event_log_impl.cc
+++ b/logging/rtc_event_log/rtc_event_log.cc
@@ -377,10 +377,4 @@
return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
}
-bool RtcEventLogNullImpl::StartLogging(
- std::unique_ptr<RtcEventLogOutput> output,
- int64_t output_period_ms) {
- return false;
-}
-
} // namespace webrtc
diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h
index 79fd39a..3a52480 100644
--- a/logging/rtc_event_log/rtc_event_log.h
+++ b/logging/rtc_event_log/rtc_event_log.h
@@ -57,7 +57,9 @@
class RtcEventLogNullImpl : public RtcEventLog {
public:
bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
- int64_t output_period_ms) override;
+ int64_t output_period_ms) override {
+ return false;
+ }
void StopLogging() override {}
void Log(std::unique_ptr<RtcEvent> event) override {}
};
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 371d2b1..eae7c5c 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -637,7 +637,6 @@
"../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",
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 988555c..8cb659f 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -955,7 +955,6 @@
"../../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",
@@ -2172,7 +2171,6 @@
"../../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",
diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn
index 6c47856..170314d 100644
--- a/modules/bitrate_controller/BUILD.gn
+++ b/modules/bitrate_controller/BUILD.gn
@@ -34,7 +34,6 @@
deps = [
"..:module_api",
- "../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -71,7 +70,6 @@
deps = [
":bitrate_controller",
"../../logging:mocks",
- "../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../test:field_trial",
"../../test:test_support",
diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index bad72a9..0d314c2 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -102,7 +102,6 @@
deps = [
"../../api:optional",
- "../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
@@ -123,7 +122,6 @@
deps = [
":estimators",
"../../:typedefs",
- "../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
diff --git a/modules/pacing/BUILD.gn b/modules/pacing/BUILD.gn
index 853aadb..31610dc 100644
--- a/modules/pacing/BUILD.gn
+++ b/modules/pacing/BUILD.gn
@@ -37,9 +37,7 @@
"../../: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",
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 7180d50..e169363 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -201,9 +201,7 @@
"../../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",
diff --git a/ortc/BUILD.gn b/ortc/BUILD.gn
index bf66c44..2005556 100644
--- a/ortc/BUILD.gn
+++ b/ortc/BUILD.gn
@@ -39,7 +39,6 @@
"../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",
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index e44ae75..47b963b 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -188,7 +188,6 @@
"../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",
@@ -222,7 +221,6 @@
"../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",
@@ -483,8 +481,7 @@
"../api/audio_codecs/L16:audio_encoder_L16",
"../call:call_interfaces",
"../logging:rtc_event_log_api",
- "../logging:rtc_event_log_impl_base",
- "../logging:rtc_event_log_impl_output",
+ "../logging:rtc_event_log_impl",
"../media:rtc_audio_video",
"../media:rtc_data", # TODO(phoglund): AFAIK only used for one sctp constant.
"../media:rtc_media_base",
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index aa897e4..976fa06 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -224,9 +224,8 @@
"../call:call_interfaces",
"../call:video_stream_api",
"../logging:rtc_event_log_api",
- "../logging:rtc_event_log_impl_base",
+ "../logging:rtc_event_log_impl",
"../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",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 7a6c036..4f85834 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -506,8 +506,6 @@
"../../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",
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 41b4bdf..bfc521d 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -613,7 +613,6 @@
"../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",
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index d2716aa..90fb62c 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -231,7 +231,7 @@
]
deps = [
"../../logging:rtc_event_log_api",
- "../../logging:rtc_event_log_impl_base",
+ "../../logging:rtc_event_log_impl",
"../../modules/congestion_controller",
"../../modules/pacing",
"../../modules/remote_bitrate_estimator:remote_bitrate_estimator",
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5e597a8..593e2b9 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -111,7 +111,6 @@
]
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",