From cc91d284e43e266f97edb999eb2ebfc8a094beac Mon Sep 17 00:00:00 2001 From: skvlad Date: Mon, 3 Oct 2016 18:31:22 -0700 Subject: [PATCH] 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 | 33 +----- webrtc/audio/webrtc_audio.gypi | 2 +- webrtc/call/BUILD.gn | 49 +-------- webrtc/call/DEPS | 1 + webrtc/call/call.cc | 2 +- webrtc/call/mock/mock_rtc_event_log.h | 2 +- webrtc/call/webrtc_call.gypi | 2 +- webrtc/logging/BUILD.gn | 100 ++++++++++++++++++ webrtc/logging/OWNERS | 3 + webrtc/logging/rtc_event_log/DEPS | 5 + .../rtc_event_log}/ringbuffer.h | 6 +- .../rtc_event_log}/ringbuffer_unittest.cc | 2 +- .../rtc_event_log}/rtc_event_log.cc | 12 +-- .../rtc_event_log}/rtc_event_log.h | 6 +- .../rtc_event_log}/rtc_event_log.proto | 13 --- .../rtc_event_log}/rtc_event_log2rtp_dump.cc | 4 +- .../rtc_event_log_helper_thread.cc | 4 +- .../rtc_event_log_helper_thread.h | 12 +-- .../rtc_event_log}/rtc_event_log_parser.cc | 4 +- .../rtc_event_log}/rtc_event_log_parser.h | 12 +-- .../rtc_event_log}/rtc_event_log_unittest.cc | 10 +- .../rtc_event_log_unittest_helper.cc | 6 +- .../rtc_event_log_unittest_helper.h | 8 +- webrtc/modules/audio_coding/BUILD.gn | 6 +- webrtc/modules/audio_coding/DEPS | 1 + webrtc/modules/audio_coding/audio_coding.gypi | 2 +- webrtc/modules/audio_coding/neteq/tools/DEPS | 3 + .../neteq/tools/rtc_event_log_source.cc | 1 - .../neteq/tools/rtc_event_log_source.h | 2 +- webrtc/modules/bitrate_controller/DEPS | 1 + .../send_side_bandwidth_estimation.cc | 2 +- webrtc/modules/rtp_rtcp/DEPS | 1 + webrtc/modules/rtp_rtcp/source/rtcp_sender.cc | 2 +- webrtc/modules/rtp_rtcp/source/rtp_sender.cc | 2 +- webrtc/test/fuzzers/BUILD.gn | 2 +- .../congestion_controller_feedback_fuzzer.cc | 2 +- webrtc/tools/BUILD.gn | 8 +- webrtc/tools/DEPS | 1 + webrtc/tools/event_log_visualizer/analyzer.h | 2 +- webrtc/tools/event_log_visualizer/main.cc | 2 +- webrtc/tools/tools.gyp | 2 +- webrtc/video/BUILD.gn | 2 +- webrtc/video/webrtc_video.gypi | 2 +- webrtc/voice_engine/BUILD.gn | 6 +- webrtc/voice_engine/DEPS | 1 + webrtc/voice_engine/channel.cc | 2 +- .../test/auto_test/standard/codec_test.cc | 1 - webrtc/voice_engine/voice_engine.gyp | 2 +- webrtc/webrtc.gyp | 32 +++--- 49 files changed, 210 insertions(+), 178 deletions(-) create mode 100644 webrtc/logging/BUILD.gn create mode 100644 webrtc/logging/OWNERS create mode 100644 webrtc/logging/rtc_event_log/DEPS rename webrtc/{call => logging/rtc_event_log}/ringbuffer.h (93%) rename webrtc/{call => logging/rtc_event_log}/ringbuffer_unittest.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log.h (97%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log.proto (99%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log2rtp_dump.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_helper_thread.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_helper_thread.h (90%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_parser.cc (99%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_parser.h (92%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_unittest.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_unittest_helper.cc (98%) rename webrtc/{call => logging/rtc_event_log}/rtc_event_log_unittest_helper.h (89%) create mode 100644 webrtc/modules/audio_coding/neteq/tools/DEPS diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index aaea4e3b9d..a97d508d4d 100644 --- a/webrtc/BUILD.gn +++ b/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) { diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi index 6230057e4f..4439d1fcbb 100644 --- a/webrtc/audio/webrtc_audio.gypi +++ b/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', diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 648bc85e0f..bd2b30e570 100644 --- a/webrtc/call/BUILD.gn +++ b/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" ] - } - } } diff --git a/webrtc/call/DEPS b/webrtc/call/DEPS index 0f9030853c..4256b589ff 100644 --- a/webrtc/call/DEPS +++ b/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", diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 16a6f469d5..dd08d770ee 100644 --- a/webrtc/call/call.cc +++ b/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" diff --git a/webrtc/call/mock/mock_rtc_event_log.h b/webrtc/call/mock/mock_rtc_event_log.h index bb1337e8f7..27623860cb 100644 --- a/webrtc/call/mock/mock_rtc_event_log.h +++ b/webrtc/call/mock/mock_rtc_event_log.h @@ -13,7 +13,7 @@ #include -#include "webrtc/call/rtc_event_log.h" +#include "webrtc/logging/rtc_event_log/rtc_event_log.h" #include "webrtc/test/gmock.h" namespace webrtc { diff --git a/webrtc/call/webrtc_call.gypi b/webrtc/call/webrtc_call.gypi index 455a11a794..59dcef6af6 100644 --- a/webrtc/call/webrtc_call.gypi +++ b/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', diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn new file mode 100644 index 0000000000..7078963d1e --- /dev/null +++ b/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" ] + } + } +} diff --git a/webrtc/logging/OWNERS b/webrtc/logging/OWNERS new file mode 100644 index 0000000000..492fb7ab32 --- /dev/null +++ b/webrtc/logging/OWNERS @@ -0,0 +1,3 @@ +skvlad@webrtc.org +stefan@webrtc.org +terelius@webrtc.org diff --git a/webrtc/logging/rtc_event_log/DEPS b/webrtc/logging/rtc_event_log/DEPS new file mode 100644 index 0000000000..039ebf9507 --- /dev/null +++ b/webrtc/logging/rtc_event_log/DEPS @@ -0,0 +1,5 @@ +include_rules = [ + "+webrtc/base", + "+webrtc/modules/rtp_rtcp", + "+webrtc/system_wrappers", +] diff --git a/webrtc/call/ringbuffer.h b/webrtc/logging/rtc_event_log/ringbuffer.h similarity index 93% rename from webrtc/call/ringbuffer.h rename to webrtc/logging/rtc_event_log/ringbuffer.h index fa5e4227ff..6c0ffda805 100644 --- a/webrtc/call/ringbuffer.h +++ b/webrtc/logging/rtc_event_log/ringbuffer.h @@ -9,8 +9,8 @@ * */ -#ifndef WEBRTC_CALL_RINGBUFFER_H_ -#define WEBRTC_CALL_RINGBUFFER_H_ +#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_ +#define WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_ #include #include @@ -97,4 +97,4 @@ class RingBuffer { } // namespace webrtc -#endif // WEBRTC_CALL_RINGBUFFER_H_ +#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RINGBUFFER_H_ diff --git a/webrtc/call/ringbuffer_unittest.cc b/webrtc/logging/rtc_event_log/ringbuffer_unittest.cc similarity index 98% rename from webrtc/call/ringbuffer_unittest.cc rename to webrtc/logging/rtc_event_log/ringbuffer_unittest.cc index 370f262180..7e672feb22 100644 --- a/webrtc/call/ringbuffer_unittest.cc +++ b/webrtc/logging/rtc_event_log/ringbuffer_unittest.cc @@ -12,7 +12,7 @@ #include #include "webrtc/base/random.h" -#include "webrtc/call/ringbuffer.h" +#include "webrtc/logging/rtc_event_log/ringbuffer.h" #include "webrtc/test/gtest.h" namespace { diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc similarity index 98% rename from webrtc/call/rtc_event_log.cc rename to webrtc/logging/rtc_event_log/rtc_event_log.cc index c022296730..ff31cee1fc 100644 --- a/webrtc/call/rtc_event_log.cc +++ b/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 #include @@ -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_, - &event_queue_, - clock), + helper_thread_(&message_queue_, &event_queue_, clock), thread_checker_() { thread_checker_.DetachFromThread(); } diff --git a/webrtc/call/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h similarity index 97% rename from webrtc/call/rtc_event_log.h rename to webrtc/logging/rtc_event_log/rtc_event_log.h index a3359692eb..910e9a61b9 100644 --- a/webrtc/call/rtc_event_log.h +++ b/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_ -#define WEBRTC_CALL_RTC_EVENT_LOG_H_ +#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ +#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ #include #include @@ -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_ diff --git a/webrtc/call/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto similarity index 99% rename from webrtc/call/rtc_event_log.proto rename to webrtc/logging/rtc_event_log/rtc_event_log.proto index b14306e362..a6d1695796 100644 --- a/webrtc/call/rtc_event_log.proto +++ b/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; diff --git a/webrtc/call/rtc_event_log2rtp_dump.cc b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc similarity index 98% rename from webrtc/call/rtc_event_log2rtp_dump.cc rename to webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc index 5733cfa31d..337b65b80f 100644 --- a/webrtc/call/rtc_event_log2rtp_dump.cc +++ b/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/call/rtc_event_log_parser.h" +#include "webrtc/logging/rtc_event_log/rtc_event_log.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" diff --git a/webrtc/call/rtc_event_log_helper_thread.cc b/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.cc similarity index 98% rename from webrtc/call/rtc_event_log_helper_thread.cc rename to webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.cc index c0f8972631..4d55da3542 100644 --- a/webrtc/call/rtc_event_log_helper_thread.cc +++ b/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 @@ -74,7 +74,7 @@ RtcEventLogHelperThread::~RtcEventLogHelperThread() { } wake_from_hibernation_.Set(); wake_periodically_.Set(); // Wake up the output thread. - thread_.Stop(); // Wait for the thread to terminate. + thread_.Stop(); // Wait for the thread to terminate. } void RtcEventLogHelperThread::WaitForFileFinished() { diff --git a/webrtc/call/rtc_event_log_helper_thread.h b/webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h similarity index 90% rename from webrtc/call/rtc_event_log_helper_thread.h rename to webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h index 1408961ec7..bda27f2f4f 100644 --- a/webrtc/call/rtc_event_log_helper_thread.h +++ b/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_ -#define WEBRTC_CALL_RTC_EVENT_LOG_HELPER_THREAD_H_ +#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_HELPER_THREAD_H_ +#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_HELPER_THREAD_H_ #include #include @@ -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_ diff --git a/webrtc/call/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc similarity index 99% rename from webrtc/call/rtc_event_log_parser.cc rename to webrtc/logging/rtc_event_log/rtc_event_log_parser.cc index a2f95d0a75..362d79e4de 100644 --- a/webrtc/call/rtc_event_log_parser.cc +++ b/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 @@ -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" diff --git a/webrtc/call/rtc_event_log_parser.h b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h similarity index 92% rename from webrtc/call/rtc_event_log_parser.h rename to webrtc/logging/rtc_event_log/rtc_event_log_parser.h index a50ec20391..6a684cb9c1 100644 --- a/webrtc/call/rtc_event_log_parser.h +++ b/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_ -#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_ +#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_ +#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_ #include #include #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_ diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc similarity index 98% rename from webrtc/call/rtc_event_log_unittest.cc rename to webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc index 6c4ec6382e..d6af3e9a12 100644 --- a/webrtc/call/rtc_event_log_unittest.cc +++ b/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/call/rtc_event_log_parser.h" -#include "webrtc/call/rtc_event_log_unittest_helper.h" +#include "webrtc/logging/rtc_event_log/rtc_event_log.h" +#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.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 { diff --git a/webrtc/call/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc similarity index 98% rename from webrtc/call/rtc_event_log_unittest_helper.cc rename to webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc index 566e92b632..b6403014b4 100644 --- a/webrtc/call/rtc_event_log_unittest_helper.cc +++ b/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 @@ -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 { diff --git a/webrtc/call/rtc_event_log_unittest_helper.h b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h similarity index 89% rename from webrtc/call/rtc_event_log_unittest_helper.h rename to webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h index b662c3ccc3..5ffb6f402b 100644 --- a/webrtc/call/rtc_event_log_unittest_helper.h +++ b/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_ -#define WEBRTC_CALL_RTC_EVENT_LOG_UNITTEST_HELPER_H_ +#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_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_ diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index ef5ea96be2..a75fbaa124 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/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", ] } diff --git a/webrtc/modules/audio_coding/DEPS b/webrtc/modules/audio_coding/DEPS index 31aa1c25fb..013cce7a6d 100644 --- a/webrtc/modules/audio_coding/DEPS +++ b/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", ] diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi index 0ed3e08034..5f9eb2cdad 100644 --- a/webrtc/modules/audio_coding/audio_coding.gypi +++ b/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', diff --git a/webrtc/modules/audio_coding/neteq/tools/DEPS b/webrtc/modules/audio_coding/neteq/tools/DEPS new file mode 100644 index 0000000000..0f16a4fcee --- /dev/null +++ b/webrtc/modules/audio_coding/neteq/tools/DEPS @@ -0,0 +1,3 @@ +include_rules = [ + "+webrtc/logging/rtc_event_log", +] diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc index 517458bf94..6e19e93ed1 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc +++ b/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" diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h index 71bf841bde..fad491e8a2 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h @@ -15,7 +15,7 @@ #include #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" diff --git a/webrtc/modules/bitrate_controller/DEPS b/webrtc/modules/bitrate_controller/DEPS index 9a462b6fc5..553c36b973 100644 --- a/webrtc/modules/bitrate_controller/DEPS +++ b/webrtc/modules/bitrate_controller/DEPS @@ -1,5 +1,6 @@ include_rules = [ "+webrtc/base", "+webrtc/call", + "+webrtc/logging/rtc_event_log", "+webrtc/system_wrappers", ] diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc index f306fb255d..f09382bdc2 100644 --- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc +++ b/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 { diff --git a/webrtc/modules/rtp_rtcp/DEPS b/webrtc/modules/rtp_rtcp/DEPS index 0720a15fec..4004651c91 100644 --- a/webrtc/modules/rtp_rtcp/DEPS +++ b/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", ] diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc index 9882d3db51..8fe3334789 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc +++ b/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" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index c7c60b9b78..2aaa58e38f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/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" diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn index 0acf1943ec..eb6104dc26 100644 --- a/webrtc/test/fuzzers/BUILD.gn +++ b/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/", ] } diff --git a/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc b/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc index 6fb5f9ba1f..496af90c69 100644 --- a/webrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc +++ b/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" diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index 08206c1d0b..1ba2018a35 100644 --- a/webrtc/tools/BUILD.gn +++ b/webrtc/tools/BUILD.gn @@ -164,8 +164,8 @@ if (rtc_enable_protobuf) { } defines = [ "ENABLE_RTC_EVENT_LOG" ] deps = [ - "../:rtc_event_log_parser", - "../call:rtc_event_log", + "../logging:rtc_event_log_impl", + "../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", ] } } diff --git a/webrtc/tools/DEPS b/webrtc/tools/DEPS index 507106a063..ac56340ece 100644 --- a/webrtc/tools/DEPS +++ b/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", diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h index 6f959e1ad5..5cbcf79b8a 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.h +++ b/webrtc/tools/event_log_visualizer/analyzer.h @@ -18,7 +18,7 @@ #include #include -#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" diff --git a/webrtc/tools/event_log_visualizer/main.cc b/webrtc/tools/event_log_visualizer/main.cc index bde172a835..b487696f66 100644 --- a/webrtc/tools/event_log_visualizer/main.cc +++ b/webrtc/tools/event_log_visualizer/main.cc @@ -11,7 +11,7 @@ #include #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" diff --git a/webrtc/tools/tools.gyp b/webrtc/tools/tools.gyp index b58b0c779a..3b3a84292d 100644 --- a/webrtc/tools/tools.gyp +++ b/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', diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index d411dad6c0..294d30c9e3 100644 --- a/webrtc/video/BUILD.gn +++ b/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", diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi index af162dead7..dde855278e 100644 --- a/webrtc/video/webrtc_video.gypi +++ b/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', diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index 8450d1dce3..dd5546be13 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/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", diff --git a/webrtc/voice_engine/DEPS b/webrtc/voice_engine/DEPS index 224eeee676..18efd816a0 100644 --- a/webrtc/voice_engine/DEPS +++ b/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", diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index 582bde5f26..ce9770e8a4 100644 --- a/webrtc/voice_engine/channel.cc +++ b/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" diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc index 5a500af2df..dad03baf7f 100644 --- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc +++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc @@ -11,7 +11,6 @@ #include #include -#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" diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 61dd2c038c..c264fe79cd 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/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': [ diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index a4c1c1d45f..98d9498366 100644 --- a/webrtc/webrtc.gyp +++ b/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', - 'call/rtc_event_log.h', - 'call/rtc_event_log_helper_thread.cc', - 'call/rtc_event_log_helper_thread.h', + 'logging/rtc_event_log/rtc_event_log.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_out_dir': 'webrtc/call', + 'proto_in_dir': 'logging/rtc_event_log', + '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', - 'call/rtc_event_log_parser.h', + 'logging/rtc_event_log/rtc_event_log_parser.cc', + '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',