diff options
Diffstat (limited to 'third_party/libwebrtc/api/rtc_event_log')
9 files changed, 593 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtc_event_log/BUILD.gn b/third_party/libwebrtc/api/rtc_event_log/BUILD.gn new file mode 100644 index 0000000000..158dc06a7b --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/BUILD.gn @@ -0,0 +1,48 @@ +# Copyright (c) 2019 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("../../webrtc.gni") + +rtc_library("rtc_event_log") { + visibility = [ "*" ] + sources = [ + "rtc_event.cc", + "rtc_event.h", + "rtc_event_log.cc", + "rtc_event_log.h", + "rtc_event_log_factory_interface.h", + ] + + deps = [ + "..:libjingle_logging_api", + "../../rtc_base:checks", + "../../rtc_base:timeutils", + "../task_queue", + ] +} + +rtc_library("rtc_event_log_factory") { + visibility = [ "*" ] + sources = [ + "rtc_event_log_factory.cc", + "rtc_event_log_factory.h", + ] + + deps = [ + ":rtc_event_log", + "../../rtc_base:checks", + "../../rtc_base/system:rtc_export", + "../../system_wrappers:field_trial", + "../task_queue", + ] + + if (rtc_enable_protobuf) { + defines = [ "WEBRTC_ENABLE_RTC_EVENT_LOG" ] + deps += [ "../../logging:rtc_event_log_impl" ] + } +} diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event.cc b/third_party/libwebrtc/api/rtc_event_log/rtc_event.cc new file mode 100644 index 0000000000..631188b915 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event.cc @@ -0,0 +1,19 @@ +/* + * Copyright (c) 2019 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. + */ + +#include "api/rtc_event_log/rtc_event.h" + +#include "rtc_base/time_utils.h" + +namespace webrtc { + +RtcEvent::RtcEvent() : timestamp_us_(rtc::TimeMillis() * 1000) {} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event.h b/third_party/libwebrtc/api/rtc_event_log/rtc_event.h new file mode 100644 index 0000000000..aa74944fe5 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event.h @@ -0,0 +1,88 @@ +/* + * Copyright (c) 2017 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. + */ + +#ifndef API_RTC_EVENT_LOG_RTC_EVENT_H_ +#define API_RTC_EVENT_LOG_RTC_EVENT_H_ + +#include <cstdint> + +namespace webrtc { + +// This class allows us to store unencoded RTC events. Subclasses of this class +// store the actual information. This allows us to keep all unencoded events, +// even when their type and associated information differ, in the same buffer. +// Additionally, it prevents dependency leaking - a module that only logs +// events of type RtcEvent_A doesn't need to know about anything associated +// with events of type RtcEvent_B. +class RtcEvent { + public: + // Subclasses of this class have to associate themselves with a unique value + // of Type. This leaks the information of existing subclasses into the + // superclass, but the *actual* information - rtclog::StreamConfig, etc. - + // is kept separate. + enum class Type : uint32_t { + AlrStateEvent, + RouteChangeEvent, + RemoteEstimateEvent, + AudioNetworkAdaptation, + AudioPlayout, + AudioReceiveStreamConfig, + AudioSendStreamConfig, + BweUpdateDelayBased, + BweUpdateLossBased, + DtlsTransportState, + DtlsWritableState, + IceCandidatePairConfig, + IceCandidatePairEvent, + ProbeClusterCreated, + ProbeResultFailure, + ProbeResultSuccess, + RtcpPacketIncoming, + RtcpPacketOutgoing, + RtpPacketIncoming, + RtpPacketOutgoing, + VideoReceiveStreamConfig, + VideoSendStreamConfig, + GenericPacketSent, + GenericPacketReceived, + GenericAckReceived, + FrameDecoded, + NetEqSetMinimumDelay, + BeginV3Log = 0x2501580, + EndV3Log = 0x2501581, + FakeEvent, // For unit testing. + }; + + RtcEvent(); + virtual ~RtcEvent() = default; + + virtual Type GetType() const = 0; + + virtual bool IsConfigEvent() const = 0; + + // Events are grouped by Type before being encoded. + // Optionally, `GetGroupKey` can be overloaded to group the + // events by a secondary key (in addition to the event type.) + // This can, in some cases, improve compression efficiency + // e.g. by grouping events by SSRC. + virtual uint32_t GetGroupKey() const { return 0; } + + int64_t timestamp_ms() const { return timestamp_us_ / 1000; } + int64_t timestamp_us() const { return timestamp_us_; } + + protected: + explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {} + + const int64_t timestamp_us_; +}; + +} // namespace webrtc + +#endif // API_RTC_EVENT_LOG_RTC_EVENT_H_ diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.cc b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.cc new file mode 100644 index 0000000000..56189c0ff7 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.cc @@ -0,0 +1,21 @@ +/* + * Copyright (c) 2018 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. + */ + +#include "api/rtc_event_log/rtc_event_log.h" + +namespace webrtc { + +bool RtcEventLogNull::StartLogging( + std::unique_ptr<RtcEventLogOutput> /*output*/, + int64_t /*output_period_ms*/) { + return false; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.h b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.h new file mode 100644 index 0000000000..7b42cdc028 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.h @@ -0,0 +1,69 @@ +/* + * Copyright (c) 2015 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. + */ + +#ifndef API_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ +#define API_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ + +#include <cstddef> +#include <cstdint> +#include <functional> +#include <memory> + +#include "api/rtc_event_log/rtc_event.h" +#include "api/rtc_event_log_output.h" +#include "api/task_queue/task_queue_factory.h" + +namespace webrtc { + +class RtcEventLog { + public: + enum : size_t { kUnlimitedOutput = 0 }; + enum : int64_t { kImmediateOutput = 0 }; + + // TODO(eladalon): Get rid of the legacy encoding and this enum once all + // clients have migrated to the new format. + enum class EncodingType { Legacy, NewFormat, ProtoFree }; + + virtual ~RtcEventLog() = default; + + // Starts logging to a given output. The output might be limited in size, + // and may close itself once it has reached the maximum size. + virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> output, + int64_t output_period_ms) = 0; + + // Stops logging to file and waits until the file has been closed, after + // which it would be permissible to read and/or modify it. + virtual void StopLogging() = 0; + + // Stops logging to file and calls `callback` when the file has been closed. + // Note that it is not safe to call any other members, including the + // destructor, until the callback has been called. + // TODO(srte): Remove default implementation when it's safe to do so. + virtual void StopLogging(std::function<void()> callback) { + StopLogging(); + callback(); + } + + // Log an RTC event (the type of event is determined by the subclass). + virtual void Log(std::unique_ptr<RtcEvent> event) = 0; +}; + +// No-op implementation is used if flag is not set, or in tests. +class RtcEventLogNull final : public RtcEventLog { + public: + bool StartLogging(std::unique_ptr<RtcEventLogOutput> output, + int64_t output_period_ms) override; + void StopLogging() override {} + void Log(std::unique_ptr<RtcEvent> event) override {} +}; + +} // namespace webrtc + +#endif // API_RTC_EVENT_LOG_RTC_EVENT_LOG_H_ diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.cc b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.cc new file mode 100644 index 0000000000..a3cb68cf54 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.cc @@ -0,0 +1,48 @@ +/* + * Copyright 2017 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. + */ + +#include "api/rtc_event_log/rtc_event_log_factory.h" + +#include <memory> +#include <utility> + +#include "rtc_base/checks.h" +#include "system_wrappers/include/field_trial.h" + +#ifdef WEBRTC_ENABLE_RTC_EVENT_LOG +#include "logging/rtc_event_log/rtc_event_log_impl.h" +#endif + +namespace webrtc { + +RtcEventLogFactory::RtcEventLogFactory(TaskQueueFactory* task_queue_factory) + : task_queue_factory_(task_queue_factory) { + RTC_DCHECK(task_queue_factory_); +} + +std::unique_ptr<RtcEventLog> RtcEventLogFactory::Create( + RtcEventLog::EncodingType encoding_type) const { +#ifdef WEBRTC_ENABLE_RTC_EVENT_LOG + if (field_trial::IsEnabled("WebRTC-RtcEventLogKillSwitch")) { + return std::make_unique<RtcEventLogNull>(); + } + return std::make_unique<RtcEventLogImpl>( + RtcEventLogImpl::CreateEncoder(encoding_type), task_queue_factory_); +#else + return std::make_unique<RtcEventLogNull>(); +#endif +} + +std::unique_ptr<RtcEventLog> RtcEventLogFactory::CreateRtcEventLog( + RtcEventLog::EncodingType encoding_type) { + return Create(encoding_type); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.h b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.h new file mode 100644 index 0000000000..fd1db3c728 --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory.h @@ -0,0 +1,39 @@ +/* + * Copyright 2017 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. + */ + +#ifndef API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_H_ +#define API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_H_ + +#include <memory> + +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/rtc_event_log/rtc_event_log_factory_interface.h" +#include "api/task_queue/task_queue_factory.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +class RTC_EXPORT RtcEventLogFactory : public RtcEventLogFactoryInterface { + public: + explicit RtcEventLogFactory(TaskQueueFactory* task_queue_factory); + ~RtcEventLogFactory() override {} + + std::unique_ptr<RtcEventLog> Create( + RtcEventLog::EncodingType encoding_type) const override; + std::unique_ptr<RtcEventLog> CreateRtcEventLog( + RtcEventLog::EncodingType encoding_type) override; + + private: + TaskQueueFactory* const task_queue_factory_; +}; + +} // namespace webrtc + +#endif // API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_H_ diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory_interface.h b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory_interface.h new file mode 100644 index 0000000000..a6f4dee92f --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_factory_interface.h @@ -0,0 +1,35 @@ +/* + * Copyright 2017 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. + */ + +#ifndef API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_INTERFACE_H_ +#define API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_INTERFACE_H_ + +#include <memory> + +#include "api/rtc_event_log/rtc_event_log.h" + +namespace webrtc { + +// This interface exists to allow webrtc to be optionally built without +// RtcEventLog support. A PeerConnectionFactory is constructed with an +// RtcEventLogFactoryInterface, which may or may not be null. +class RtcEventLogFactoryInterface { + public: + virtual ~RtcEventLogFactoryInterface() = default; + + virtual std::unique_ptr<RtcEventLog> Create( + RtcEventLog::EncodingType encoding_type) const = 0; + [[deprecated]] virtual std::unique_ptr<RtcEventLog> CreateRtcEventLog( + RtcEventLog::EncodingType encoding_type) = 0; +}; + +} // namespace webrtc + +#endif // API_RTC_EVENT_LOG_RTC_EVENT_LOG_FACTORY_INTERFACE_H_ diff --git a/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_gn/moz.build b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_gn/moz.build new file mode 100644 index 0000000000..a96be6791f --- /dev/null +++ b/third_party/libwebrtc/api/rtc_event_log/rtc_event_log_gn/moz.build @@ -0,0 +1,226 @@ +# This Source Code Form is subject to the terms of the Mozilla Public +# License, v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. + + + ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ### + ### DO NOT edit it by hand. ### + +COMPILE_FLAGS["OS_INCLUDES"] = [] +AllowCompilerWarnings() + +DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1" +DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True +DEFINES["RTC_ENABLE_VP9"] = True +DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0" +DEFINES["WEBRTC_LIBRARY_IMPL"] = True +DEFINES["WEBRTC_MOZILLA_BUILD"] = True +DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0" +DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0" + +FINAL_LIBRARY = "webrtc" + + +LOCAL_INCLUDES += [ + "!/ipc/ipdl/_ipdlheaders", + "!/third_party/libwebrtc/gen", + "/ipc/chromium/src", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/rtc_event_log/rtc_event.cc", + "/third_party/libwebrtc/api/rtc_event_log/rtc_event_log.cc" +] + +if not CONFIG["MOZ_DEBUG"]: + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0" + DEFINES["NDEBUG"] = True + DEFINES["NVALGRIND"] = True + +if CONFIG["MOZ_DEBUG"] == "1": + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1" + +if CONFIG["OS_TARGET"] == "Android": + + DEFINES["ANDROID"] = True + DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1" + DEFINES["HAVE_SYS_UIO_H"] = True + DEFINES["WEBRTC_ANDROID"] = True + DEFINES["WEBRTC_ANDROID_OPENSLES"] = True + DEFINES["WEBRTC_LINUX"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_GNU_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + + OS_LIBS += [ + "log" + ] + +if CONFIG["OS_TARGET"] == "Darwin": + + DEFINES["WEBRTC_MAC"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True + DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0" + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "Linux": + + DEFINES["USE_AURA"] = "1" + DEFINES["USE_GLIB"] = "1" + DEFINES["USE_NSS_CERTS"] = "1" + DEFINES["USE_OZONE"] = "1" + DEFINES["USE_UDEV"] = True + DEFINES["WEBRTC_LINUX"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_FILE_OFFSET_BITS"] = "64" + DEFINES["_LARGEFILE64_SOURCE"] = True + DEFINES["_LARGEFILE_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "OpenBSD": + + DEFINES["USE_GLIB"] = "1" + DEFINES["USE_OZONE"] = "1" + DEFINES["USE_X11"] = "1" + DEFINES["WEBRTC_BSD"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_FILE_OFFSET_BITS"] = "64" + DEFINES["_LARGEFILE64_SOURCE"] = True + DEFINES["_LARGEFILE_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "WINNT": + + DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True + DEFINES["NOMINMAX"] = True + DEFINES["NTDDI_VERSION"] = "0x0A000000" + DEFINES["PSAPI_VERSION"] = "2" + DEFINES["UNICODE"] = True + DEFINES["USE_AURA"] = "1" + DEFINES["WEBRTC_WIN"] = True + DEFINES["WIN32"] = True + DEFINES["WIN32_LEAN_AND_MEAN"] = True + DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP" + DEFINES["WINVER"] = "0x0A00" + DEFINES["_ATL_NO_OPENGL"] = True + DEFINES["_CRT_RAND_S"] = True + DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True + DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True + DEFINES["_HAS_EXCEPTIONS"] = "0" + DEFINES["_HAS_NODISCARD"] = True + DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True + DEFINES["_SECURE_ATL"] = True + DEFINES["_UNICODE"] = True + DEFINES["_WIN32_WINNT"] = "0x0A00" + DEFINES["_WINDOWS"] = True + DEFINES["__STD_C"] = True + + OS_LIBS += [ + "winmm" + ] + +if CONFIG["CPU_ARCH"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "arm": + + CXXFLAGS += [ + "-mfpu=neon" + ] + + DEFINES["WEBRTC_ARCH_ARM"] = True + DEFINES["WEBRTC_ARCH_ARM_V7"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "mips32": + + DEFINES["MIPS32_LE"] = True + DEFINES["MIPS_FPU_LE"] = True + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "mips64": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["CPU_ARCH"] == "x86_64": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT": + + DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0" + +if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["USE_X11"] = "1" + +if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android": + + OS_LIBS += [ + "android_support", + "unwind" + ] + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android": + + CXXFLAGS += [ + "-msse2" + ] + + OS_LIBS += [ + "android_support" + ] + +if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux": + + CXXFLAGS += [ + "-msse2" + ] + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +Library("rtc_event_log_gn") |