| /* |
| * 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/call/rtc_event_log.h" |
| |
| #include <deque> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/call.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" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/system_wrappers/include/file_wrapper.h" |
| |
| #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" |
| #else |
| #include "webrtc/call/rtc_event_log.pb.h" |
| #endif |
| #endif |
| |
| namespace webrtc { |
| |
| #ifndef ENABLE_RTC_EVENT_LOG |
| |
| // No-op implementation if flag is not set. |
| class RtcEventLogImpl final : public RtcEventLog { |
| public: |
| void SetBufferDuration(int64_t buffer_duration_us) override {} |
| void StartLogging(const std::string& file_name, int duration_ms) override {} |
| bool StartLogging(rtc::PlatformFile log_file) override { return false; } |
| void StopLogging(void) override {} |
| void LogVideoReceiveStreamConfig( |
| const VideoReceiveStream::Config& config) override {} |
| void LogVideoSendStreamConfig( |
| const VideoSendStream::Config& config) override {} |
| void LogRtpHeader(bool incoming, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) override {} |
| void LogRtcpPacket(bool incoming, |
| 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 {} |
| }; |
| |
| #else // ENABLE_RTC_EVENT_LOG is defined |
| |
| class RtcEventLogImpl final : public RtcEventLog { |
| public: |
| RtcEventLogImpl(); |
| |
| void SetBufferDuration(int64_t buffer_duration_us) override; |
| void StartLogging(const std::string& file_name, int duration_ms) override; |
| bool StartLogging(rtc::PlatformFile log_file) override; |
| void StopLogging() override; |
| void LogVideoReceiveStreamConfig( |
| const VideoReceiveStream::Config& config) override; |
| void LogVideoSendStreamConfig(const VideoSendStream::Config& config) override; |
| void LogRtpHeader(bool incoming, |
| MediaType media_type, |
| const uint8_t* header, |
| size_t packet_length) override; |
| void LogRtcpPacket(bool incoming, |
| 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; |
| |
| private: |
| // Starts logging. This function assumes the file_ has been opened succesfully |
| // and that the start_time_us_ and _duration_us_ have been set. |
| void StartLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| // Stops logging and clears the stored data and buffers. |
| void StopLoggingLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| // Adds a new event to the logfile if logging is active, or adds it to the |
| // list of recent log events otherwise. |
| void HandleEvent(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| // Writes the event to the file. Note that this will destroy the state of the |
| // input argument. |
| void StoreToFile(rtclog::Event* event) EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| // Adds the event to the list of recent events, and removes any events that |
| // are too old and no longer fall in the time window. |
| void AddRecentEvent(const rtclog::Event& event) |
| EXCLUSIVE_LOCKS_REQUIRED(crit_); |
| |
| rtc::CriticalSection crit_; |
| rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_) = |
| rtc::scoped_ptr<FileWrapper>(FileWrapper::Create()); |
| rtc::PlatformFile platform_file_ GUARDED_BY(crit_) = |
| rtc::kInvalidPlatformFileValue; |
| rtclog::EventStream stream_ GUARDED_BY(crit_); |
| std::deque<rtclog::Event> recent_log_events_ GUARDED_BY(crit_); |
| std::vector<rtclog::Event> config_events_ GUARDED_BY(crit_); |
| |
| // Microseconds to record log events, before starting the actual log. |
| int64_t buffer_duration_us_ GUARDED_BY(crit_); |
| bool currently_logging_ GUARDED_BY(crit_); |
| int64_t start_time_us_ GUARDED_BY(crit_); |
| int64_t duration_us_ GUARDED_BY(crit_); |
| const Clock* const clock_; |
| }; |
| |
| 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. |
| |
| // Do not add default return values to the conversion functions in this |
| // unnamed namespace. The intention is to make the compiler warn if anyone |
| // adds unhandled new events/modes/etc. |
| |
| 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; |
| } |
| |
| } // namespace |
| |
| namespace { |
| bool IsConfigEvent(const rtclog::Event& event) { |
| rtclog::Event_EventType event_type = event.type(); |
| return event_type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT || |
| event_type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT || |
| event_type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT || |
| event_type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT; |
| } |
| } // namespace |
| |
| // RtcEventLogImpl member functions. |
| RtcEventLogImpl::RtcEventLogImpl() |
| : file_(FileWrapper::Create()), |
| stream_(), |
| buffer_duration_us_(10000000), |
| currently_logging_(false), |
| start_time_us_(0), |
| duration_us_(0), |
| clock_(Clock::GetRealTimeClock()) { |
| } |
| |
| void RtcEventLogImpl::SetBufferDuration(int64_t buffer_duration_us) { |
| rtc::CritScope lock(&crit_); |
| buffer_duration_us_ = buffer_duration_us; |
| } |
| |
| void RtcEventLogImpl::StartLogging(const std::string& file_name, |
| int duration_ms) { |
| rtc::CritScope lock(&crit_); |
| if (currently_logging_) { |
| StopLoggingLocked(); |
| } |
| if (file_->OpenFile(file_name.c_str(), false) != 0) { |
| return; |
| } |
| start_time_us_ = clock_->TimeInMicroseconds(); |
| duration_us_ = static_cast<int64_t>(duration_ms) * 1000; |
| StartLoggingLocked(); |
| } |
| |
| bool RtcEventLogImpl::StartLogging(rtc::PlatformFile log_file) { |
| rtc::CritScope lock(&crit_); |
| |
| if (currently_logging_) { |
| StopLoggingLocked(); |
| } |
| RTC_DCHECK(platform_file_ == rtc::kInvalidPlatformFileValue); |
| |
| FILE* file_stream = rtc::FdopenPlatformFileForWriting(log_file); |
| if (!file_stream) { |
| rtc::ClosePlatformFile(log_file); |
| return false; |
| } |
| |
| if (file_->OpenFromFileHandle(file_stream, true, false) != 0) { |
| rtc::ClosePlatformFile(log_file); |
| return false; |
| } |
| platform_file_ = log_file; |
| // Set the start time and duration to keep logging for 10 minutes. |
| start_time_us_ = clock_->TimeInMicroseconds(); |
| duration_us_ = 10 * 60 * 1000000; |
| StartLoggingLocked(); |
| return true; |
| } |
| |
| void RtcEventLogImpl::StartLoggingLocked() { |
| currently_logging_ = true; |
| |
| // Write all old configuration events to the log file. |
| for (auto& event : config_events_) { |
| StoreToFile(&event); |
| } |
| // Write all recent configuration events to the log file, and |
| // write all other recent events to the log file, ignoring any old events. |
| for (auto& event : recent_log_events_) { |
| if (IsConfigEvent(event)) { |
| StoreToFile(&event); |
| config_events_.push_back(event); |
| } else if (event.timestamp_us() >= start_time_us_ - buffer_duration_us_) { |
| StoreToFile(&event); |
| } |
| } |
| recent_log_events_.clear(); |
| // Write a LOG_START event to the file. |
| rtclog::Event start_event; |
| start_event.set_timestamp_us(start_time_us_); |
| start_event.set_type(rtclog::Event::LOG_START); |
| StoreToFile(&start_event); |
| } |
| |
| void RtcEventLogImpl::StopLogging() { |
| rtc::CritScope lock(&crit_); |
| StopLoggingLocked(); |
| } |
| |
| void RtcEventLogImpl::LogVideoReceiveStreamConfig( |
| const VideoReceiveStream::Config& config) { |
| rtc::CritScope lock(&crit_); |
| |
| rtclog::Event event; |
| event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| 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) { |
| rtclog::RtxMap* rtx = receiver_config->add_rtx_map(); |
| rtx->set_payload_type(kv.first); |
| rtx->mutable_config()->set_rtx_ssrc(kv.second.ssrc); |
| rtx->mutable_config()->set_rtx_payload_type(kv.second.payload_type); |
| } |
| |
| for (const auto& e : config.rtp.extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| receiver_config->add_header_extensions(); |
| extension->set_name(e.name); |
| 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); |
| } |
| HandleEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogVideoSendStreamConfig( |
| const VideoSendStream::Config& config) { |
| rtc::CritScope lock(&crit_); |
| |
| rtclog::Event event; |
| event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| 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.name); |
| 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); |
| HandleEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogRtpHeader(bool incoming, |
| 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; |
| } |
| |
| rtc::CritScope lock(&crit_); |
| rtclog::Event rtp_event; |
| rtp_event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| rtp_event.set_type(rtclog::Event::RTP_EVENT); |
| rtp_event.mutable_rtp_packet()->set_incoming(incoming); |
| 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); |
| HandleEvent(&rtp_event); |
| } |
| |
| void RtcEventLogImpl::LogRtcpPacket(bool incoming, |
| MediaType media_type, |
| const uint8_t* packet, |
| size_t length) { |
| rtc::CritScope lock(&crit_); |
| rtclog::Event rtcp_event; |
| rtcp_event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| rtcp_event.set_type(rtclog::Event::RTCP_EVENT); |
| rtcp_event.mutable_rtcp_packet()->set_incoming(incoming); |
| rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type)); |
| |
| RTCPUtility::RtcpCommonHeader 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 (!RtcpParseCommonHeader(block_begin, packet_end - block_begin, |
| &header)) { |
| break; // Incorrect message header. |
| } |
| uint32_t block_size = header.BlockSize(); |
| switch (header.packet_type) { |
| case RTCPUtility::PT_SR: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_RR: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_BYE: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_IJ: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_RTPFB: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_PSFB: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_XR: |
| // 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 RTCPUtility::PT_SDES: |
| FALLTHROUGH(); |
| case RTCPUtility::PT_APP: |
| FALLTHROUGH(); |
| 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); |
| HandleEvent(&rtcp_event); |
| } |
| |
| void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) { |
| rtc::CritScope lock(&crit_); |
| rtclog::Event event; |
| event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT); |
| auto playout_event = event.mutable_audio_playout_event(); |
| playout_event->set_local_ssrc(ssrc); |
| HandleEvent(&event); |
| } |
| |
| void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate, |
| uint8_t fraction_loss, |
| int32_t total_packets) { |
| rtc::CritScope lock(&crit_); |
| rtclog::Event event; |
| event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| 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); |
| HandleEvent(&event); |
| } |
| |
| void RtcEventLogImpl::StopLoggingLocked() { |
| if (currently_logging_) { |
| currently_logging_ = false; |
| // Create a LogEnd event |
| rtclog::Event event; |
| event.set_timestamp_us(clock_->TimeInMicroseconds()); |
| event.set_type(rtclog::Event::LOG_END); |
| // Store the event and close the file |
| RTC_DCHECK(file_->Open()); |
| StoreToFile(&event); |
| file_->CloseFile(); |
| if (platform_file_ != rtc::kInvalidPlatformFileValue) { |
| rtc::ClosePlatformFile(platform_file_); |
| platform_file_ = rtc::kInvalidPlatformFileValue; |
| } |
| } |
| RTC_DCHECK(!file_->Open()); |
| stream_.Clear(); |
| } |
| |
| void RtcEventLogImpl::HandleEvent(rtclog::Event* event) { |
| if (currently_logging_) { |
| if (clock_->TimeInMicroseconds() < start_time_us_ + duration_us_) { |
| StoreToFile(event); |
| return; |
| } |
| StopLoggingLocked(); |
| } |
| AddRecentEvent(*event); |
| } |
| |
| void RtcEventLogImpl::StoreToFile(rtclog::Event* event) { |
| // Reuse the same object at every log event. |
| if (stream_.stream_size() < 1) { |
| stream_.add_stream(); |
| } |
| RTC_DCHECK_EQ(stream_.stream_size(), 1); |
| stream_.mutable_stream(0)->Swap(event); |
| // TODO(terelius): Doesn't this create a new EventStream per event? |
| // Is this guaranteed to work e.g. in future versions of protobuf? |
| std::string dump_buffer; |
| stream_.SerializeToString(&dump_buffer); |
| file_->Write(dump_buffer.data(), dump_buffer.size()); |
| } |
| |
| void RtcEventLogImpl::AddRecentEvent(const rtclog::Event& event) { |
| recent_log_events_.push_back(event); |
| while (recent_log_events_.front().timestamp_us() < |
| event.timestamp_us() - buffer_duration_us_) { |
| if (IsConfigEvent(recent_log_events_.front())) { |
| config_events_.push_back(recent_log_events_.front()); |
| } |
| recent_log_events_.pop_front(); |
| } |
| } |
| |
| bool RtcEventLog::ParseRtcEventLog(const std::string& file_name, |
| rtclog::EventStream* result) { |
| char tmp_buffer[1024]; |
| int bytes_read = 0; |
| rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create()); |
| if (dump_file->OpenFile(file_name.c_str(), true) != 0) { |
| 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 |
| |
| // RtcEventLog member functions. |
| rtc::scoped_ptr<RtcEventLog> RtcEventLog::Create() { |
| return rtc::scoped_ptr<RtcEventLog>(new RtcEventLogImpl()); |
| } |
| |
| } // namespace webrtc |