Moved RtcEventLog files from call/ to logging/

The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.

After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).

The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).

This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.

BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2380683005 .

Cr-Commit-Position: refs/heads/master@{#14485}

webrtc/BUILD.gn

@@ -283,7 +283,7 @@ if (!is_ios || !build_with_chromium) {
 
     if (rtc_enable_protobuf) {
       defines += [ "ENABLE_RTC_EVENT_LOG" ]
-      deps += [ ":rtc_event_log_proto" ]
+      deps += [ "logging:rtc_event_log_proto" ]
     }
   }
 }
@@ -305,35 +305,6 @@ rtc_static_library("webrtc_common") {
   }
 }
 
-if (rtc_enable_protobuf) {
-  proto_library("rtc_event_log_proto") {
-    sources = [
-      "call/rtc_event_log.proto",
-    ]
-    proto_out_dir = "webrtc/call"
-  }
-}
-
-if (rtc_enable_protobuf) {
-  rtc_static_library("rtc_event_log_parser") {
-    sources = [
-      "call/rtc_event_log_parser.cc",
-      "call/rtc_event_log_parser.h",
-    ]
-
-    public_deps = [
-      ":rtc_event_log_proto",
-      ":webrtc_common",
-    ]
-
-    if (is_clang && !is_nacl) {
-      # Suppress warnings from Chrome's Clang plugins.
-      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
-  }
-}
-
 if (use_libfuzzer || use_drfuzz || use_afl) {
   # This target is only here for gn to discover fuzzer build targets under
   # webrtc/test/fuzzers/.
@@ -529,7 +500,7 @@ if (rtc_include_tests) {
     ]
 
     if (rtc_enable_protobuf) {
-      deps += [ "call:rtc_event_log_tests" ]
+      deps += [ "logging:rtc_event_log_tests" ]
     }
 
     if (is_android) {

webrtc/audio/webrtc_audio.gypi

@@ -12,7 +12,7 @@
       '<(webrtc_root)/common.gyp:webrtc_common',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
     ],
     'webrtc_audio_sources': [
       'audio/audio_receive_stream.cc',

webrtc/call/BUILD.gn

@@ -23,11 +23,11 @@ rtc_static_library("call") {
   }
 
   deps = [
-    ":rtc_event_log",
     "..:webrtc_common",
     "../api:call_api",
     "../audio",
     "../base:rtc_task_queue",
+    "../logging:rtc_event_log_impl",
     "../modules/congestion_controller",
     "../modules/rtp_rtcp",
     "../system_wrappers",
@@ -35,32 +35,6 @@ rtc_static_library("call") {
   ]
 }
 
-rtc_static_library("rtc_event_log") {
-  sources = [
-    "rtc_event_log.cc",
-    "rtc_event_log.h",
-    "rtc_event_log_helper_thread.cc",
-    "rtc_event_log_helper_thread.h",
-  ]
-
-  defines = []
-
-  deps = [
-    "..:webrtc_common",
-    "../modules/rtp_rtcp",
-  ]
-
-  if (rtc_enable_protobuf) {
-    defines += [ "ENABLE_RTC_EVENT_LOG" ]
-    deps += [ "..:rtc_event_log_proto" ]
-  }
-  if (is_clang && !is_nacl) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-  }
-}
-
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
@@ -69,7 +43,6 @@ if (rtc_include_tests) {
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "packet_injection_tests.cc",
-      "ringbuffer_unittest.cc",
     ]
     deps = [
       ":call",
@@ -82,24 +55,4 @@ if (rtc_include_tests) {
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
-  rtc_source_set("rtc_event_log_tests") {
-    testonly = true
-    sources = [
-      "rtc_event_log_unittest.cc",
-      "rtc_event_log_unittest_helper.cc",
-    ]
-    deps = [
-      ":rtc_event_log",
-      "..:rtc_event_log_parser",
-      "../modules/rtp_rtcp",
-      "../system_wrappers:metrics_default",
-      "//testing/gmock",
-      "//testing/gtest",
-    ]
-    if (is_clang) {
-      # Suppress warnings from the Chromium Clang plugin.
-      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
-    }
-  }
 }

webrtc/call/DEPS

@@ -1,6 +1,7 @@
 include_rules = [
   "+webrtc/audio",
   "+webrtc/base",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_coding",
   "+webrtc/modules/bitrate_controller",
   "+webrtc/modules/congestion_controller",

webrtc/call/call.cc

@@ -27,8 +27,8 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
 #include "webrtc/call/bitrate_allocator.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/config.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"

webrtc/call/mock/mock_rtc_event_log.h

@@ -13,7 +13,7 @@
 
 #include <string>
 
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/test/gmock.h"
 
 namespace webrtc {

webrtc/call/webrtc_call.gypi

@@ -12,7 +12,7 @@
       '<(webrtc_root)/modules/modules.gyp:congestion_controller',
       '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log_impl',
     ],
     'webrtc_call_sources': [
       'call/bitrate_allocator.cc',

webrtc/logging/BUILD.gn

@@ -0,0 +1,100 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+import("//third_party/protobuf/proto_library.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("rtc_event_log_api") {
+  sources = [
+    "rtc_event_log/rtc_event_log.h",
+  ]
+}
+
+rtc_static_library("rtc_event_log_impl") {
+  sources = [
+    "rtc_event_log/ringbuffer.h",
+    "rtc_event_log/rtc_event_log.cc",
+    "rtc_event_log/rtc_event_log_helper_thread.cc",
+    "rtc_event_log/rtc_event_log_helper_thread.h",
+  ]
+
+  defines = []
+
+  deps = [
+    ":rtc_event_log_api",
+    "..:webrtc_common",
+    "../modules/rtp_rtcp",
+  ]
+
+  if (rtc_enable_protobuf) {
+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
+    deps += [ ":rtc_event_log_proto" ]
+  }
+  if (is_clang && !is_nacl) {
+    # Suppress warnings from Chrome's Clang plugins.
+    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+if (rtc_include_tests) {
+  rtc_source_set("rtc_event_log_tests") {
+    testonly = true
+    sources = [
+      "rtc_event_log/ringbuffer_unittest.cc",
+      "rtc_event_log/rtc_event_log_unittest.cc",
+      "rtc_event_log/rtc_event_log_unittest_helper.cc",
+    ]
+    deps = [
+      ":rtc_event_log_impl",
+      ":rtc_event_log_parser",
+      "../modules/rtp_rtcp",
+      "../system_wrappers:metrics_default",
+      "//testing/gmock",
+      "//testing/gtest",
+    ]
+    if (is_clang) {
+      # Suppress warnings from the Chromium Clang plugin.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}
+
+if (rtc_enable_protobuf) {
+  proto_library("rtc_event_log_proto") {
+    sources = [
+      "rtc_event_log/rtc_event_log.proto",
+    ]
+    proto_out_dir = "webrtc/logging/rtc_event_log"
+  }
+}
+
+if (rtc_enable_protobuf) {
+  rtc_static_library("rtc_event_log_parser") {
+    sources = [
+      "rtc_event_log/rtc_event_log_parser.cc",
+      "rtc_event_log/rtc_event_log_parser.h",
+    ]
+
+    public_deps = [
+      ":rtc_event_log_proto",
+      "..:webrtc_common",
+    ]
+
+    if (is_clang && !is_nacl) {
+      # Suppress warnings from Chrome's Clang plugins.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}

webrtc/logging/OWNERS

@@ -0,0 +1,3 @@
+skvlad@webrtc.org
+stefan@webrtc.org
+terelius@webrtc.org

webrtc/logging/rtc_event_log/DEPS

@@ -0,0 +1,5 @@
+include_rules = [
+  "+webrtc/base",
+  "+webrtc/modules/rtp_rtcp",
+  "+webrtc/system_wrappers",
+]

webrtc/logging/rtc_event_log/ringbuffer.h

@@ -9,8 +9,8 @@
  *
  */
 
-#ifndef WEBRTC_CALL_RINGBUFFER_H_
+#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_
-#define WEBRTC_CALL_RINGBUFFER_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_
 
 #include <memory>
 #include <utility>
@@ -97,4 +97,4 @@ class RingBuffer {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CALL_RINGBUFFER_H_
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_

webrtc/logging/rtc_event_log/ringbuffer_unittest.cc

@@ -12,7 +12,7 @@
 #include <list>
 
 #include "webrtc/base/random.h"
-#include "webrtc/call/ringbuffer.h"
+#include "webrtc/logging/rtc_event_log/ringbuffer.h"
 #include "webrtc/test/gtest.h"
 
 namespace {

webrtc/logging/rtc_event_log/rtc_event_log.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 
 #include <limits>
 #include <vector>
@@ -19,7 +19,7 @@
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log_helper_thread.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@@ -30,9 +30,9 @@
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 #endif
 
@@ -129,9 +129,7 @@ RtcEventLogImpl::RtcEventLogImpl(const Clock* clock)
     : message_queue_(kControlMessagesPerSecond),
       event_queue_(kEventsPerSecond),
       clock_(clock),
-      helper_thread_(&message_queue_,
+      helper_thread_(&message_queue_, &event_queue_, clock),
-                     &event_queue_,
-                     clock),
       thread_checker_() {
   thread_checker_.DetachFromThread();
 }

webrtc/logging/rtc_event_log/rtc_event_log.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_
+#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
-#define WEBRTC_CALL_RTC_EVENT_LOG_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 
 #include <memory>
 #include <string>
@@ -139,4 +139,4 @@ class RtcEventLogNullImpl final : public RtcEventLog {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_

webrtc/logging/rtc_event_log/rtc_event_log.proto

@@ -2,7 +2,6 @@ syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 package webrtc.rtclog;
 
-
 enum MediaType {
   ANY = 0;
   AUDIO = 1;
@@ -10,7 +9,6 @@ enum MediaType {
   DATA = 3;
 }
 
-
 // This is the main message to dump to a file, it can contain multiple event
 // messages, but it is possible to append multiple EventStreams (each with a
 // single event) to a file.
@@ -19,7 +17,6 @@ message EventStream {
   repeated Event stream = 1;
 }
 
-
 message Event {
   // required - Elapsed wallclock time in us since the start of the log.
   optional int64 timestamp_us = 1;
@@ -70,7 +67,6 @@ message Event {
   optional AudioSendConfig audio_sender_config = 11;
 }
 
-
 message RtpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
@@ -87,7 +83,6 @@ message RtpPacket {
   // Do not add code to log user payload data without a privacy review!
 }
 
-
 message RtcpPacket {
   // required - True if the packet is incoming w.r.t. the user logging the data
   optional bool incoming = 1;
@@ -150,7 +145,6 @@ message VideoReceiveConfig {
   repeated DecoderConfig decoders = 7;
 }
 
-
 // Maps decoder names to payload types.
 message DecoderConfig {
   // required
@@ -160,7 +154,6 @@ message DecoderConfig {
   optional int32 payload_type = 2;
 }
 
-
 // Maps RTP header extension names to numerical IDs.
 message RtpHeaderExtension {
   // required
@@ -170,7 +163,6 @@ message RtpHeaderExtension {
   optional int32 id = 2;
 }
 
-
 // RTX settings for incoming video payloads that may be received.
 // RTX is disabled if there's no config present.
 message RtxConfig {
@@ -181,7 +173,6 @@ message RtxConfig {
   optional int32 rtx_payload_type = 2;
 }
 
-
 message RtxMap {
   // required
   optional int32 payload_type = 1;
@@ -190,7 +181,6 @@ message RtxMap {
   optional RtxConfig config = 2;
 }
 
-
 message VideoSendConfig {
   // Synchronization source (stream identifier) for outgoing stream.
   // One stream can have several ssrcs for e.g. simulcast.
@@ -210,7 +200,6 @@ message VideoSendConfig {
   optional EncoderConfig encoder = 5;
 }
 
-
 // Maps encoder names to payload types.
 message EncoderConfig {
   // required
@@ -220,7 +209,6 @@ message EncoderConfig {
   optional int32 payload_type = 2;
 }
 
-
 message AudioReceiveConfig {
   // required - Synchronization source (stream identifier) to be received.
   optional uint32 remote_ssrc = 1;
@@ -232,7 +220,6 @@ message AudioReceiveConfig {
   repeated RtpHeaderExtension header_extensions = 3;
 }
 
-
 message AudioSendConfig {
   // required - Synchronization source (stream identifier) for outgoing stream.
   optional uint32 ssrc = 1;

webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc

@@ -16,8 +16,8 @@
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/rtp_file_writer.h"
 

webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/call/rtc_event_log_helper_thread.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
 
 #include <algorithm>
 

webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
+#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_HELPER_THREAD_H_
-#define WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_HELPER_THREAD_H_
 
 #include <limits>
 #include <memory>
@@ -22,7 +22,7 @@
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/swap_queue.h"
-#include "webrtc/call/ringbuffer.h"
+#include "webrtc/logging/rtc_event_log/ringbuffer.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
@@ -30,9 +30,9 @@
 // Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif
@@ -129,4 +129,4 @@ class RtcEventLogHelperThread final {
 
 #endif  // ENABLE_RTC_EVENT_LOG
 
-#endif  // WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_HELPER_THREAD_H_

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 
 #include <string.h>
 
@@ -19,7 +19,7 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 

webrtc/logging/rtc_event_log/rtc_event_log_parser.h

@@ -7,23 +7,23 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
-#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/ignore_wundef.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 // Files generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 
@@ -120,4 +120,4 @@ class ParsedRtcEventLog {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_

webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc

@@ -18,9 +18,9 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
-#include "webrtc/call/rtc_event_log_unittest_helper.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
@@ -33,9 +33,9 @@
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 
 namespace webrtc {

webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/call/rtc_event_log_unittest_helper.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
 
 #include <string.h>
 
@@ -21,9 +21,9 @@
 
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #else
-#include "webrtc/call/rtc_event_log.pb.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 
 namespace webrtc {

webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
+#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
-#define WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
 
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 
 namespace webrtc {
 
@@ -55,4 +55,4 @@ class RtcEventLogTestHelper {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_

webrtc/modules/audio_coding/BUILD.gn

@@ -117,7 +117,7 @@ rtc_static_library("audio_coding") {
            ":audio_network_adaptor",
            ":neteq",
            ":rent_a_codec",
-           "../../call:rtc_event_log",
+           "../../logging:rtc_event_log_api",
          ]
   defines = audio_coding_defines
 }
@@ -1021,10 +1021,10 @@ if (rtc_include_tests) {
       }
 
       deps = [
-        "../../:rtc_event_log_parser",
+        "../../logging:rtc_event_log_parser",
       ]
       public_deps = [
-        "../../:rtc_event_log_proto",
+        "../../logging:rtc_event_log_proto",
       ]
     }
 

webrtc/modules/audio_coding/DEPS

@@ -2,6 +2,7 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_audio",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/audio_coding/neteq/neteq_unittest.pb.h",  # Different path.
   "+webrtc/system_wrappers",
 ]

webrtc/modules/audio_coding/audio_coding.gypi

@@ -147,7 +147,7 @@
       'dependencies': [
         '<@(audio_coding_dependencies)',
         '<(webrtc_root)/common.gyp:webrtc_common',
-        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+        '<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
         'audio_network_adaptor',
         'neteq',
         'rent_a_codec',

webrtc/modules/audio_coding/neteq/tools/DEPS

@@ -0,0 +1,3 @@
+include_rules = [
+  "+webrtc/logging/rtc_event_log",
+]

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc

@@ -17,7 +17,6 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 

webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h

@@ -15,7 +15,7 @@
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 

webrtc/modules/bitrate_controller/DEPS

@@ -1,5 +1,6 @@
 include_rules = [
   "+webrtc/base",
   "+webrtc/call",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/system_wrappers",
 ]

webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

@@ -15,9 +15,9 @@
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/call/rtc_event_log.h"
 
 namespace webrtc {
 namespace {

webrtc/modules/rtp_rtcp/DEPS

@@ -2,5 +2,6 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_video",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/system_wrappers",
 ]

webrtc/modules/rtp_rtcp/source/rtcp_sender.cc

@@ -19,8 +19,8 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common_types.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

@@ -19,7 +19,7 @@
 #include "webrtc/base/trace_event.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"

webrtc/test/fuzzers/BUILD.gn

@@ -130,7 +130,7 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
     "congestion_controller_feedback_fuzzer.cc",
   ]
   deps = [
-    "../../call:rtc_event_log",
+    "../../logging:rtc_event_log_impl",
     "../../modules/congestion_controller/",
   ]
 }

webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"

webrtc/tools/BUILD.gn

@@ -164,8 +164,8 @@ if (rtc_enable_protobuf) {
     }
     defines = [ "ENABLE_RTC_EVENT_LOG" ]
     deps = [
-      "../:rtc_event_log_parser",
+      "../logging:rtc_event_log_impl",
-      "../call:rtc_event_log",
+      "../logging:rtc_event_log_parser",
       "../modules/congestion_controller:congestion_controller",
       "../modules/rtp_rtcp:rtp_rtcp",
       "../system_wrappers:system_wrappers_default",
@@ -173,7 +173,7 @@ if (rtc_enable_protobuf) {
     ]
     public_deps = [
       ":chart_proto",
-      "../:rtc_event_log_parser",
+      "../logging:rtc_event_log_parser",
     ]
   }
 }
@@ -282,7 +282,7 @@ if (rtc_include_tests) {
         "$root_build_dir/{{source_file_part}}",
       ]
       deps = [
-        "..:rtc_event_log_proto",
+        "../logging:rtc_event_log_proto",
       ]
     }
   }

webrtc/tools/DEPS

@@ -2,6 +2,7 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_video",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_device",
   "+webrtc/modules/audio_processing",
   "+webrtc/modules/bitrate_controller",

webrtc/tools/event_log_visualizer/analyzer.h

@@ -18,7 +18,7 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/tools/event_log_visualizer/plot_base.h"

webrtc/tools/event_log_visualizer/main.cc

@@ -11,7 +11,7 @@
 #include <iostream>
 
 #include "gflags/gflags.h"
-#include "webrtc/call/rtc_event_log_parser.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/tools/event_log_visualizer/analyzer.h"
 #include "webrtc/tools/event_log_visualizer/plot_base.h"
 #include "webrtc/tools/event_log_visualizer/plot_python.h"

webrtc/tools/tools.gyp

@@ -118,7 +118,7 @@
           'target_name': 'event_log_visualizer_utils',
           'type': 'static_library',
           'dependencies': [
-            '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+            '<(webrtc_root)/webrtc.gyp:rtc_event_log_impl',
             '<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
             '<(webrtc_root)/modules/modules.gyp:congestion_controller',
             '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',

webrtc/video/BUILD.gn

@@ -58,8 +58,8 @@ rtc_static_library("video") {
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../base:rtc_task_queue",
-    "../call:rtc_event_log",
     "../common_video",
+    "../logging:rtc_event_log_api",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",

webrtc/video/webrtc_video.gypi

@@ -21,7 +21,7 @@
       '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
       '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
       '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-      '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+      '<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
     ],
     'webrtc_video_sources': [
       'video/call_stats.cc',

webrtc/voice_engine/BUILD.gn

@@ -92,8 +92,8 @@ rtc_static_library("voice_engine") {
     "..:webrtc_common",
     "../api:call_api",
     "../base:rtc_base_approved",
-    "../call:rtc_event_log",
     "../common_audio",
+    "../logging:rtc_event_log_api",
     "../modules/audio_conference_mixer",
     "../modules/audio_device",
     "../modules/audio_processing",
@@ -244,7 +244,7 @@ if (rtc_include_tests) {
       ":voice_engine",
       "//testing/gtest",
       "//third_party/gflags",
-      "//webrtc/call:rtc_event_log",
+      "//webrtc/logging:rtc_event_log_api",
       "//webrtc/system_wrappers",
       "//webrtc/system_wrappers:system_wrappers_default",
       "//webrtc/test:test_support",
@@ -271,7 +271,7 @@ if (rtc_include_tests) {
         "//testing/gmock",
         "//testing/gtest",
         "//third_party/gflags",
-        "//webrtc/call:rtc_event_log",
+        "//webrtc/logging:rtc_event_log_api",
         "//webrtc/modules/video_capture",
         "//webrtc/system_wrappers",
         "//webrtc/system_wrappers/:system_wrappers_default",

webrtc/voice_engine/DEPS

@@ -2,6 +2,7 @@ include_rules = [
   "+webrtc/base",
   "+webrtc/call",
   "+webrtc/common_audio",
+  "+webrtc/logging/rtc_event_log",
   "+webrtc/modules/audio_coding",
   "+webrtc/modules/audio_conference_mixer",
   "+webrtc/modules/audio_device",

webrtc/voice_engine/channel.cc

@@ -20,8 +20,8 @@
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/config.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"

webrtc/voice_engine/test/auto_test/standard/codec_test.cc

@@ -11,7 +11,6 @@
 #include <stdio.h>
 #include <string>
 
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/test/test_suite.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"

webrtc/voice_engine/voice_engine.gyp

@@ -29,7 +29,7 @@
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
-        '<(webrtc_root)/webrtc.gyp:rtc_event_log',
+        '<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
         'level_indicator',
       ],
       'export_dependent_settings': [

webrtc/webrtc.gyp

@@ -32,7 +32,7 @@
         '<@(webrtc_audio_dependencies)',
         '<@(webrtc_call_dependencies)',
         '<@(webrtc_video_dependencies)',
-        'rtc_event_log',
+        'rtc_event_log_impl',
       ],
       'conditions': [
         # TODO(andresp): Chromium should link directly with this and no if
@@ -45,19 +45,27 @@
       ],
     },
     {
-      'target_name': 'rtc_event_log',
+      'target_name': 'rtc_event_log_api',
       'type': 'static_library',
       'sources': [
-        'call/rtc_event_log.cc',
+        'logging/rtc_event_log/rtc_event_log.h',
-        'call/rtc_event_log.h',
+      ],
-        'call/rtc_event_log_helper_thread.cc',
+    },
-        'call/rtc_event_log_helper_thread.h',
+    {
+      'target_name': 'rtc_event_log_impl',
+      'type': 'static_library',
+      'sources': [
+        'logging/rtc_event_log/ringbuffer.h',
+        'logging/rtc_event_log/rtc_event_log.cc',
+        'logging/rtc_event_log/rtc_event_log_helper_thread.cc',
+        'logging/rtc_event_log/rtc_event_log_helper_thread.h',
       ],
       'conditions': [
         # If enable_protobuf is defined, we want to compile the protobuf
         # and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
         ['enable_protobuf==1', {
           'dependencies': [
+            'rtc_event_log_api',
             'rtc_event_log_proto',
           ],
           'defines': [
@@ -79,10 +87,10 @@
           # This target should only be built if enable_protobuf is defined
           'target_name': 'rtc_event_log_proto',
           'type': 'static_library',
-          'sources': ['call/rtc_event_log.proto',],
+          'sources': ['logging/rtc_event_log/rtc_event_log.proto',],
           'variables': {
-            'proto_in_dir': 'call',
+            'proto_in_dir': 'logging/rtc_event_log',
-            'proto_out_dir': 'webrtc/call',
+            'proto_out_dir': 'webrtc/logging/rtc_event_log',
           },
           'includes': ['build/protoc.gypi'],
         },
@@ -90,8 +98,8 @@
           'target_name': 'rtc_event_log_parser',
           'type': 'static_library',
           'sources': [
-            'call/rtc_event_log_parser.cc',
+            'logging/rtc_event_log/rtc_event_log_parser.cc',
-            'call/rtc_event_log_parser.h',
+            'logging/rtc_event_log/rtc_event_log_parser.h',
           ],
           'dependencies': [
             'rtc_event_log_proto',
@@ -107,7 +115,7 @@
         {
           'target_name': 'rtc_event_log2rtp_dump',
           'type': 'executable',
-          'sources': ['call/rtc_event_log2rtp_dump.cc',],
+          'sources': ['logging/rtc_event_log2rtp_dump.cc',],
           'dependencies': [
             '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             'rtc_event_log_parser',