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}

audio/BUILD.gn

@@ -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",

call/BUILD.gn

@@ -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",

logging/BUILD.gn

@@ -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",
+    ":rtc_event_log_api",
-    "../:typedefs",
+    ":rtc_stream_config",
-    "../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") {
+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",

logging/rtc_event_log/rtc_event_log.h

@@ -57,9 +57,7 @@ class RtcEventLog {
 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 {}
 };

logging/rtc_event_log/rtc_event_log_impl.cc

@@ -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

media/BUILD.gn

@@ -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",

modules/audio_coding/BUILD.gn

@@ -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",

modules/bitrate_controller/BUILD.gn

@@ -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",

modules/congestion_controller/BUILD.gn

@@ -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",

modules/pacing/BUILD.gn

@@ -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",

modules/rtp_rtcp/BUILD.gn

@@ -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",

ortc/BUILD.gn

@@ -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",

pc/BUILD.gn

@@ -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",

rtc_tools/BUILD.gn

@@ -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",

sdk/android/BUILD.gn

@@ -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",

test/BUILD.gn

@@ -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",

test/fuzzers/BUILD.gn

@@ -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",

video/BUILD.gn

@@ -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",