| /* |
| * 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. |
| */ |
| |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| |
| #include <limits> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/event.h" |
| #include "webrtc/base/swap_queue.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/call/call.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_packet/app.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| #include "webrtc/system_wrappers/include/file_wrapper.h" |
| #include "webrtc/system_wrappers/include/logging.h" |
| |
| #ifdef ENABLE_RTC_EVENT_LOG |
| // Files generated at build-time by the protobuf compiler. |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| #else |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| #endif |
| #endif |
| |
| namespace webrtc { |
| |
| #ifdef ENABLE_RTC_EVENT_LOG |
| |
| class RtcEventLogImpl final : public RtcEventLog { |
| public: |
| RtcEventLogImpl(); |
| ~RtcEventLogImpl() override; |
| |
| bool StartLogging(const std::string& file_name, |
| int64_t max_size_bytes) override; |
| bool StartLogging(rtc::PlatformFile platform_file, |
| int64_t max_size_bytes) override; |
| void StopLogging() override; |
| void LogVideoReceiveStreamConfig( |
| const VideoReceiveStream::Config& config) override; |
| void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override; |
| void LogAudioReceiveStreamConfig( |
| const AudioReceiveStream::Config& config) override; |
| void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override; |
| void LogRtpHeader(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) override; |
| void LogRtcpPacket(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* packet, |
| size_t length) override; |
| void LogAudioPlayout(uint32_t ssrc) override; |
| void LogBwePacketLossEvent(int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets) override; |
| void LogBwePacketDelayEvent(int32_t bitrate, |
| BandwidthUsage detector_state) override; |
| void LogAudioNetworkAdaptation( |
| const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override; |
| |
| private: |
| void StoreEvent(std::unique_ptr<rtclog::Event>* event); |
| |
| // Message queue for passing control messages to the logging thread. |
| SwapQueue<RtcEventLogHelperThread::ControlMessage> message_queue_; |
| |
| // Message queue for passing events to the logging thread. |
| SwapQueue<std::unique_ptr<rtclog::Event> > event_queue_; |
| |
| RtcEventLogHelperThread helper_thread_; |
| rtc::ThreadChecker thread_checker_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogImpl); |
| }; |
| |
| namespace { |
| // The functions in this namespace convert enums from the runtime format |
| // that the rest of the WebRtc project can use, to the corresponding |
| // serialized enum which is defined by the protobuf. |
| |
| rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) { |
| switch (rtcp_mode) { |
| case RtcpMode::kCompound: |
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| case RtcpMode::kReducedSize: |
| return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE; |
| case RtcpMode::kOff: |
| RTC_NOTREACHED(); |
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| } |
| RTC_NOTREACHED(); |
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| } |
| |
| rtclog::MediaType ConvertMediaType(MediaType media_type) { |
| switch (media_type) { |
| case MediaType::ANY: |
| return rtclog::MediaType::ANY; |
| case MediaType::AUDIO: |
| return rtclog::MediaType::AUDIO; |
| case MediaType::VIDEO: |
| return rtclog::MediaType::VIDEO; |
| case MediaType::DATA: |
| return rtclog::MediaType::DATA; |
| } |
| RTC_NOTREACHED(); |
| return rtclog::ANY; |
| } |
| |
| rtclog::BwePacketDelayEvent::DetectorState ConvertDetectorState( |
| BandwidthUsage state) { |
| switch (state) { |
| case BandwidthUsage::kBwNormal: |
| return rtclog::BwePacketDelayEvent::BWE_NORMAL; |
| case BandwidthUsage::kBwUnderusing: |
| return rtclog::BwePacketDelayEvent::BWE_UNDERUSING; |
| case BandwidthUsage::kBwOverusing: |
| return rtclog::BwePacketDelayEvent::BWE_OVERUSING; |
| } |
| RTC_NOTREACHED(); |
| return rtclog::BwePacketDelayEvent::BWE_NORMAL; |
| } |
| |
| // The RTP and RTCP buffers reserve space for twice the expected number of |
| // sent packets because they also contain received packets. |
| static const int kEventsPerSecond = 1000; |
| static const int kControlMessagesPerSecond = 10; |
| } // namespace |
| |
| // RtcEventLogImpl member functions. |
| RtcEventLogImpl::RtcEventLogImpl() |
| // Allocate buffers for roughly one second of history. |
| : message_queue_(kControlMessagesPerSecond), |
| event_queue_(kEventsPerSecond), |
| helper_thread_(&message_queue_, &event_queue_), |
| thread_checker_() { |
| thread_checker_.DetachFromThread(); |
| } |
| |
| RtcEventLogImpl::~RtcEventLogImpl() { |
| // The RtcEventLogHelperThread destructor closes the file |
| // and waits for the thread to terminate. |
| } |
| |
| bool RtcEventLogImpl::StartLogging(const std::string& file_name, |
| int64_t max_size_bytes) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| RtcEventLogHelperThread::ControlMessage message; |
| message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE; |
| message.max_size_bytes = max_size_bytes <= 0 |
| ? std::numeric_limits<int64_t>::max() |
| : max_size_bytes; |
| message.start_time = rtc::TimeMicros(); |
| message.stop_time = std::numeric_limits<int64_t>::max(); |
| message.file.reset(FileWrapper::Create()); |
| if (!message.file->OpenFile(file_name.c_str(), false)) { |
| LOG(LS_ERROR) << "Can't open file. WebRTC event log not started."; |
| return false; |
| } |
| if (!message_queue_.Insert(&message)) { |
| LOG(LS_ERROR) << "Message queue full. Can't start logging."; |
| return false; |
| } |
| helper_thread_.SignalNewEvent(); |
| LOG(LS_INFO) << "Starting WebRTC event log."; |
| return true; |
| } |
| |
| bool RtcEventLogImpl::StartLogging(rtc::PlatformFile platform_file, |
| int64_t max_size_bytes) { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| RtcEventLogHelperThread::ControlMessage message; |
| message.message_type = RtcEventLogHelperThread::ControlMessage::START_FILE; |
| message.max_size_bytes = max_size_bytes <= 0 |
| ? std::numeric_limits<int64_t>::max() |
| : max_size_bytes; |
| message.start_time = rtc::TimeMicros(); |
| message.stop_time = std::numeric_limits<int64_t>::max(); |
| message.file.reset(FileWrapper::Create()); |
| FILE* file_handle = rtc::FdopenPlatformFileForWriting(platform_file); |
| if (!file_handle) { |
| LOG(LS_ERROR) << "Can't open file. WebRTC event log not started."; |
| // Even though we failed to open a FILE*, the platform_file is still open |
| // and needs to be closed. |
| if (!rtc::ClosePlatformFile(platform_file)) { |
| LOG(LS_ERROR) << "Can't close file."; |
| } |
| return false; |
| } |
| if (!message.file->OpenFromFileHandle(file_handle)) { |
| LOG(LS_ERROR) << "Can't open file. WebRTC event log not started."; |
| return false; |
| } |
| if (!message_queue_.Insert(&message)) { |
| LOG(LS_ERROR) << "Message queue full. Can't start logging."; |
| return false; |
| } |
| helper_thread_.SignalNewEvent(); |
| LOG(LS_INFO) << "Starting WebRTC event log."; |
| return true; |
| } |
| |
| void RtcEventLogImpl::StopLogging() { |
| RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
| RtcEventLogHelperThread::ControlMessage message; |
| message.message_type = RtcEventLogHelperThread::ControlMessage::STOP_FILE; |
| message.stop_time = rtc::TimeMicros(); |
| while (!message_queue_.Insert(&message)) { |
| // TODO(terelius): We would like to have a blocking Insert function in the |
| // SwapQueue, but for the time being we will just clear any previous |
| // messages. |
| // Since StopLogging waits for the thread, it is essential that we don't |
| // clear any STOP_FILE messages. To ensure that there is only one call at a |
| // time, we require that all calls to StopLogging are made on the same |
| // thread. |
| LOG(LS_ERROR) << "Message queue full. Clearing queue to stop logging."; |
| message_queue_.Clear(); |
| } |
| LOG(LS_INFO) << "Stopping WebRTC event log."; |
| helper_thread_.WaitForFileFinished(); |
| } |
| |
| void RtcEventLogImpl::LogVideoReceiveStreamConfig( |
| const VideoReceiveStream::Config& config) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT); |
| |
| rtclog::VideoReceiveConfig* receiver_config = |
| event->mutable_video_receiver_config(); |
| receiver_config->set_remote_ssrc(config.rtp.remote_ssrc); |
| receiver_config->set_local_ssrc(config.rtp.local_ssrc); |
| |
| receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode)); |
| receiver_config->set_remb(config.rtp.remb); |
| |
| for (const auto& kv : config.rtp.rtx_payload_types) { |
| rtclog::RtxMap* rtx = receiver_config->add_rtx_map(); |
| rtx->set_payload_type(kv.first); |
| rtx->mutable_config()->set_rtx_ssrc(config.rtp.rtx_ssrc); |
| rtx->mutable_config()->set_rtx_payload_type(kv.second); |
| } |
| |
| for (const auto& e : config.rtp.extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| receiver_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| for (const auto& d : config.decoders) { |
| rtclog::DecoderConfig* decoder = receiver_config->add_decoders(); |
| decoder->set_name(d.payload_name); |
| decoder->set_payload_type(d.payload_type); |
| } |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogVideoSendStreamConfig( |
| const VideoSendStream::Config& config) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT); |
| |
| rtclog::VideoSendConfig* sender_config = event->mutable_video_sender_config(); |
| |
| for (const auto& ssrc : config.rtp.ssrcs) { |
| sender_config->add_ssrcs(ssrc); |
| } |
| |
| for (const auto& e : config.rtp.extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| sender_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| for (const auto& rtx_ssrc : config.rtp.rtx.ssrcs) { |
| sender_config->add_rtx_ssrcs(rtx_ssrc); |
| } |
| sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type); |
| |
| rtclog::EncoderConfig* encoder = sender_config->mutable_encoder(); |
| encoder->set_name(config.encoder_settings.payload_name); |
| encoder->set_payload_type(config.encoder_settings.payload_type); |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogAudioReceiveStreamConfig( |
| const AudioReceiveStream::Config& config) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT); |
| |
| rtclog::AudioReceiveConfig* receiver_config = |
| event->mutable_audio_receiver_config(); |
| receiver_config->set_remote_ssrc(config.rtp.remote_ssrc); |
| receiver_config->set_local_ssrc(config.rtp.local_ssrc); |
| |
| for (const auto& e : config.rtp.extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| receiver_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogAudioSendStreamConfig( |
| const AudioSendStream::Config& config) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT); |
| |
| rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config(); |
| |
| sender_config->set_ssrc(config.rtp.ssrc); |
| |
| for (const auto& e : config.rtp.extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| sender_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogRtpHeader(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) { |
| // Read header length (in bytes) from packet data. |
| if (packet_length < 12u) { |
| return; // Don't read outside the packet. |
| } |
| const bool x = (header[0] & 0x10) != 0; |
| const uint8_t cc = header[0] & 0x0f; |
| size_t header_length = 12u + cc * 4u; |
| |
| if (x) { |
| if (packet_length < 12u + cc * 4u + 4u) { |
| return; // Don't read outside the packet. |
| } |
| size_t x_len = ByteReader<uint16_t>::ReadBigEndian(header + 14 + cc * 4); |
| header_length += (x_len + 1) * 4; |
| } |
| |
| std::unique_ptr<rtclog::Event> rtp_event(new rtclog::Event()); |
| rtp_event->set_timestamp_us(rtc::TimeMicros()); |
| rtp_event->set_type(rtclog::Event::RTP_EVENT); |
| rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket); |
| rtp_event->mutable_rtp_packet()->set_type(ConvertMediaType(media_type)); |
| rtp_event->mutable_rtp_packet()->set_packet_length(packet_length); |
| rtp_event->mutable_rtp_packet()->set_header(header, header_length); |
| StoreEvent(&rtp_event); |
| } |
| |
| void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction, |
| MediaType media_type, |
| const uint8_t* packet, |
| size_t length) { |
| std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event()); |
| rtcp_event->set_timestamp_us(rtc::TimeMicros()); |
| rtcp_event->set_type(rtclog::Event::RTCP_EVENT); |
| rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket); |
| rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type)); |
| |
| rtcp::CommonHeader header; |
| const uint8_t* block_begin = packet; |
| const uint8_t* packet_end = packet + length; |
| RTC_DCHECK(length <= IP_PACKET_SIZE); |
| uint8_t buffer[IP_PACKET_SIZE]; |
| uint32_t buffer_length = 0; |
| while (block_begin < packet_end) { |
| if (!header.Parse(block_begin, packet_end - block_begin)) { |
| break; // Incorrect message header. |
| } |
| const uint8_t* next_block = header.NextPacket(); |
| uint32_t block_size = next_block - block_begin; |
| switch (header.type()) { |
| case rtcp::SenderReport::kPacketType: |
| case rtcp::ReceiverReport::kPacketType: |
| case rtcp::Bye::kPacketType: |
| case rtcp::ExtendedJitterReport::kPacketType: |
| case rtcp::Rtpfb::kPacketType: |
| case rtcp::Psfb::kPacketType: |
| case rtcp::ExtendedReports::kPacketType: |
| // We log sender reports, receiver reports, bye messages |
| // inter-arrival jitter, third-party loss reports, payload-specific |
| // feedback and extended reports. |
| memcpy(buffer + buffer_length, block_begin, block_size); |
| buffer_length += block_size; |
| break; |
| case rtcp::Sdes::kPacketType: |
| case rtcp::App::kPacketType: |
| default: |
| // We don't log sender descriptions, application defined messages |
| // or message blocks of unknown type. |
| break; |
| } |
| |
| block_begin += block_size; |
| } |
| rtcp_event->mutable_rtcp_packet()->set_packet_data(buffer, buffer_length); |
| StoreEvent(&rtcp_event); |
| } |
| |
| void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT); |
| auto playout_event = event->mutable_audio_playout_event(); |
| playout_event->set_local_ssrc(ssrc); |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT); |
| auto bwe_event = event->mutable_bwe_packet_loss_event(); |
| bwe_event->set_bitrate(bitrate); |
| bwe_event->set_fraction_loss(fraction_loss); |
| bwe_event->set_total_packets(total_packets); |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogBwePacketDelayEvent(int32_t bitrate, |
| BandwidthUsage detector_state) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::BWE_PACKET_DELAY_EVENT); |
| auto bwe_event = event->mutable_bwe_packet_delay_event(); |
| bwe_event->set_bitrate(bitrate); |
| bwe_event->set_detector_state(ConvertDetectorState(detector_state)); |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogAudioNetworkAdaptation( |
| const AudioNetworkAdaptor::EncoderRuntimeConfig& config) { |
| std::unique_ptr<rtclog::Event> event(new rtclog::Event()); |
| event->set_timestamp_us(rtc::TimeMicros()); |
| event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
| auto audio_network_adaptation = event->mutable_audio_network_adaptation(); |
| if (config.bitrate_bps) |
| audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps); |
| if (config.frame_length_ms) |
| audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms); |
| if (config.uplink_packet_loss_fraction) { |
| audio_network_adaptation->set_uplink_packet_loss_fraction( |
| *config.uplink_packet_loss_fraction); |
| } |
| if (config.enable_fec) |
| audio_network_adaptation->set_enable_fec(*config.enable_fec); |
| if (config.enable_dtx) |
| audio_network_adaptation->set_enable_dtx(*config.enable_dtx); |
| if (config.num_channels) |
| audio_network_adaptation->set_num_channels(*config.num_channels); |
| StoreEvent(&event); |
| } |
| |
| void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) { |
| if (!event_queue_.Insert(event)) { |
| LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event."; |
| } |
| helper_thread_.SignalNewEvent(); |
| } |
| |
| bool RtcEventLog::ParseRtcEventLog(const std::string& file_name, |
| rtclog::EventStream* result) { |
| char tmp_buffer[1024]; |
| int bytes_read = 0; |
| std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create()); |
| if (!dump_file->OpenFile(file_name.c_str(), true)) { |
| return false; |
| } |
| std::string dump_buffer; |
| while ((bytes_read = dump_file->Read(tmp_buffer, sizeof(tmp_buffer))) > 0) { |
| dump_buffer.append(tmp_buffer, bytes_read); |
| } |
| dump_file->CloseFile(); |
| return result->ParseFromString(dump_buffer); |
| } |
| |
| #endif // ENABLE_RTC_EVENT_LOG |
| |
| bool RtcEventLogNullImpl::StartLogging(rtc::PlatformFile platform_file, |
| int64_t max_size_bytes) { |
| // The platform_file is open and needs to be closed. |
| if (!rtc::ClosePlatformFile(platform_file)) { |
| LOG(LS_ERROR) << "Can't close file."; |
| } |
| return false; |
| } |
| |
| // RtcEventLog member functions. |
| std::unique_ptr<RtcEventLog> RtcEventLog::Create() { |
| #ifdef ENABLE_RTC_EVENT_LOG |
| return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl()); |
| #else |
| return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
| #endif // ENABLE_RTC_EVENT_LOG |
| } |
| |
| std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() { |
| return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl()); |
| } |
| |
| } // namespace webrtc |