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',