Reland "Break up rtc_event_log_api to solve circular dependencies."

This is a reland of 001546d 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 <[email protected]> > Reviewed-by: Björn Terelius <[email protected]> > Reviewed-by: Stefan Holmer <[email protected]> > Commit-Queue: Qingsi Wang <[email protected]> > Cr-Commit-Position: refs/heads/master@{#21811} [email protected],[email protected],[email protected],[email protected] Bug: None Change-Id: I3e7213733741cbfd5dd0076f32209e6bc42a0647 Reviewed-on: https://webrtc-review.googlesource.com/46900 Commit-Queue: Qingsi Wang <[email protected]> Reviewed-by: Mirko Bonadei <[email protected]> Cr-Commit-Position: refs/heads/master@{#21862}
cneyoo · Feb 1, 2018 · 970b088 · 970b088
1 parent a9c94d5
commit 970b088
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Showing 17 changed files with 180 additions and 34 deletions.
diff --git a/audio/BUILD.gn b/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",

diff --git a/call/BUILD.gn b/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",

diff --git a/logging/BUILD.gn b/logging/BUILD.gn
@@ -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",
@@ -55,62 +130,103 @@ 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",
+    ":rtc_event_log_api",
+    ":rtc_stream_config",
+  ]
+}
+
+rtc_static_library("rtc_event_log_impl_encoder") {
+  visibility = [ "*" ]
+  sources = [
+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.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",
+    "../modules/audio_coding:audio_network_adaptor",
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
   ]
 
+  if (rtc_enable_protobuf) {
+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
+    deps += [ ":rtc_event_log_proto" ]
+  }
+
   # 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_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/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",
+    "rtc_event_log/rtc_event_log_impl.cc",
   ]
 
   defines = []
 
   deps = [
     ":rtc_event_log_api",
-    "..:webrtc_common",
-    "../modules/audio_coding:audio_network_adaptor",
-    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
-    "../modules/rtp_rtcp:rtp_rtcp_format",
+    ":rtc_event_log_impl_encoder",
+    ":rtc_event_log_impl_output",
     "../rtc_base:checks",
-    "../rtc_base:protobuf_utils",
     "../rtc_base:rtc_base_approved",
-    "../rtc_base:rtc_task_queue",
+    "../rtc_base:rtc_task_queue_api",
     "../rtc_base:sequenced_task_checker",
   ]
 
@@ -119,7 +235,6 @@ rtc_static_library("rtc_event_log_impl") {
     deps += [ ":rtc_event_log_proto" ]
   }
 
-  # 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" ]
@@ -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",

diff --git a/logging/rtc_event_log/rtc_event_log.cc → logging/rtc_event_log/rtc_event_log_impl.cc b/logging/rtc_event_log/rtc_event_log.cc → logging/rtc_event_log/rtc_event_log_impl.cc
diff --git a/media/BUILD.gn b/media/BUILD.gn
@@ -637,6 +637,7 @@ 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",

diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
@@ -957,6 +957,7 @@ 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",
@@ -2174,6 +2175,7 @@ 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",

diff --git a/modules/bitrate_controller/BUILD.gn b/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",

diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
@@ -103,6 +103,7 @@ rtc_source_set("estimators") {
 
   deps = [
     "../../api:optional",
+    "../../logging:rtc_event_bwe",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
@@ -123,6 +124,7 @@ 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",

diff --git a/modules/pacing/BUILD.gn b/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",