| Index: webrtc/video/rtc_event_log.cc
|
| diff --git a/webrtc/video/rtc_event_log.cc b/webrtc/video/rtc_event_log.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..476ee2afd79c82b9be6b41feb7424939842d942b
|
| --- /dev/null
|
| +++ b/webrtc/video/rtc_event_log.cc
|
| @@ -0,0 +1,406 @@
|
| +/*
|
| + * 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/video/rtc_event_log.h"
|
| +
|
| +#include <deque>
|
| +
|
| +#include "webrtc/base/checks.h"
|
| +#include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/base/thread_annotations.h"
|
| +#include "webrtc/call.h"
|
| +#include "webrtc/system_wrappers/interface/clock.h"
|
| +#include "webrtc/system_wrappers/interface/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/video/rtc_event_log.pb.h"
|
| +#else
|
| +#include "webrtc/video/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 StartLogging(const std::string& file_name, int duration_ms) override {}
|
| + 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 header_length,
|
| + size_t total_length) override {}
|
| + void LogRtcpPacket(bool incoming,
|
| + MediaType media_type,
|
| + const uint8_t* packet,
|
| + size_t length) override {}
|
| + void LogDebugEvent(DebugEvent event_type) override {}
|
| +};
|
| +
|
| +#else // ENABLE_RTC_EVENT_LOG is defined
|
| +
|
| +class RtcEventLogImpl final : public RtcEventLog {
|
| + public:
|
| + RtcEventLogImpl();
|
| +
|
| + void StartLogging(const std::string& file_name, int duration_ms) 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 header_length,
|
| + size_t total_length) override;
|
| + void LogRtcpPacket(bool incoming,
|
| + MediaType media_type,
|
| + const uint8_t* packet,
|
| + size_t length) override;
|
| + void LogDebugEvent(DebugEvent event_type) override;
|
| +
|
| + private:
|
| + // 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_);
|
| +
|
| + // Amount of time in microseconds to record log events, before starting the
|
| + // actual log.
|
| + const int recent_log_duration_us = 10000000;
|
| +
|
| + rtc::CriticalSection crit_;
|
| + rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_);
|
| + rtclog::EventStream stream_ GUARDED_BY(crit_);
|
| + std::deque<rtclog::Event> recent_log_events_ 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::DebugEvent_EventType ConvertDebugEvent(
|
| + RtcEventLog::DebugEvent event_type) {
|
| + switch (event_type) {
|
| + case RtcEventLog::DebugEvent::kLogStart:
|
| + return rtclog::DebugEvent::LOG_START;
|
| + case RtcEventLog::DebugEvent::kLogEnd:
|
| + return rtclog::DebugEvent::LOG_END;
|
| + case RtcEventLog::DebugEvent::kAudioPlayout:
|
| + return rtclog::DebugEvent::AUDIO_PLAYOUT;
|
| + }
|
| + RTC_NOTREACHED();
|
| + return rtclog::DebugEvent::UNKNOWN_EVENT;
|
| +}
|
| +
|
| +rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(
|
| + newapi::RtcpMode rtcp_mode) {
|
| + switch (rtcp_mode) {
|
| + case newapi::kRtcpCompound:
|
| + return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
|
| + case newapi::kRtcpReducedSize:
|
| + return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
|
| + }
|
| + 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
|
| +
|
| +// RtcEventLogImpl member functions.
|
| +RtcEventLogImpl::RtcEventLogImpl()
|
| + : file_(FileWrapper::Create()),
|
| + stream_(),
|
| + currently_logging_(false),
|
| + start_time_us_(0),
|
| + duration_us_(0),
|
| + clock_(Clock::GetRealTimeClock()) {
|
| +}
|
| +
|
| +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;
|
| + }
|
| + currently_logging_ = true;
|
| + start_time_us_ = clock_->TimeInMicroseconds();
|
| + duration_us_ = static_cast<int64_t>(duration_ms) * 1000;
|
| + // Write all the recent events to the log file, ignoring any old events.
|
| + for (auto& event : recent_log_events_) {
|
| + if (event.timestamp_us() >= start_time_us_ - recent_log_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::DEBUG_EVENT);
|
| + auto debug_event = start_event.mutable_debug_event();
|
| + debug_event->set_type(ConvertDebugEvent(DebugEvent::kLogStart));
|
| + 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;
|
| + const int64_t timestamp = clock_->TimeInMicroseconds();
|
| + event.set_timestamp_us(timestamp);
|
| + 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_receiver_reference_time_report(
|
| + config.rtp.rtcp_xr.receiver_reference_time_report);
|
| + 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);
|
| + }
|
| + // TODO(terelius): We should use a separate event queue for config events.
|
| + // The current approach of storing the configuration together with the
|
| + // RTP events causes the configuration information to be removed 10s
|
| + // after the ReceiveStream is created.
|
| + HandleEvent(&event);
|
| +}
|
| +
|
| +void RtcEventLogImpl::LogVideoSendStreamConfig(
|
| + const VideoSendStream::Config& config) {
|
| + rtc::CritScope lock(&crit_);
|
| +
|
| + rtclog::Event event;
|
| + const int64_t timestamp = clock_->TimeInMicroseconds();
|
| + event.set_timestamp_us(timestamp);
|
| + 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);
|
| +
|
| + sender_config->set_c_name(config.rtp.c_name);
|
| +
|
| + rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
|
| + encoder->set_name(config.encoder_settings.payload_name);
|
| + encoder->set_payload_type(config.encoder_settings.payload_type);
|
| +
|
| + // TODO(terelius): We should use a separate event queue for config events.
|
| + // The current approach of storing the configuration together with the
|
| + // RTP events causes the configuration information to be removed 10s
|
| + // after the ReceiveStream is created.
|
| + HandleEvent(&event);
|
| +}
|
| +
|
| +// TODO(terelius): It is more convenient and less error prone to parse the
|
| +// header length from the packet instead of relying on the caller to provide it.
|
| +void RtcEventLogImpl::LogRtpHeader(bool incoming,
|
| + MediaType media_type,
|
| + const uint8_t* header,
|
| + size_t header_length,
|
| + size_t total_length) {
|
| + rtc::CritScope lock(&crit_);
|
| + rtclog::Event rtp_event;
|
| + const int64_t timestamp = clock_->TimeInMicroseconds();
|
| + rtp_event.set_timestamp_us(timestamp);
|
| + 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(total_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;
|
| + const int64_t timestamp = clock_->TimeInMicroseconds();
|
| + rtcp_event.set_timestamp_us(timestamp);
|
| + 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));
|
| + rtcp_event.mutable_rtcp_packet()->set_packet_data(packet, length);
|
| + HandleEvent(&rtcp_event);
|
| +}
|
| +
|
| +void RtcEventLogImpl::LogDebugEvent(DebugEvent event_type) {
|
| + rtc::CritScope lock(&crit_);
|
| + rtclog::Event event;
|
| + const int64_t timestamp = clock_->TimeInMicroseconds();
|
| + event.set_timestamp_us(timestamp);
|
| + event.set_type(rtclog::Event::DEBUG_EVENT);
|
| + auto debug_event = event.mutable_debug_event();
|
| + debug_event->set_type(ConvertDebugEvent(event_type));
|
| + HandleEvent(&event);
|
| +}
|
| +
|
| +void RtcEventLogImpl::StopLoggingLocked() {
|
| + if (currently_logging_) {
|
| + currently_logging_ = false;
|
| + // Create a LogEnd debug event
|
| + rtclog::Event event;
|
| + int64_t timestamp = clock_->TimeInMicroseconds();
|
| + event.set_timestamp_us(timestamp);
|
| + event.set_type(rtclog::Event::DEBUG_EVENT);
|
| + auto debug_event = event.mutable_debug_event();
|
| + debug_event->set_type(ConvertDebugEvent(DebugEvent::kLogEnd));
|
| + // Store the event and close the file
|
| + DCHECK(file_->Open());
|
| + StoreToFile(&event);
|
| + file_->CloseFile();
|
| + }
|
| + 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();
|
| + }
|
| + 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() - recent_log_duration_us) {
|
| + 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
|
|
|