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Unified Diff: webrtc/video/rtc_event_log.cc

Issue 1230973005: Adds logging of configuration information for VideoReceiveStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments from stefan. Created 5 years, 5 months ago
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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
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