Adds logging of configuration information for VideoReceiveStream

webrtc/video/rtc_event_log.cc

+/*
+ *  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 {
+
+// No-op implementation if flag is not set.
+#ifndef ENABLE_RTC_EVENT_LOG
+class RtcEventLogImpl final : public RtcEventLog {
+ public:
+  void StartLogging(const std::string& file_name, int duration_ms) override{};

[stefan-webrtc, 2015/07/28 08:15:34]

Space after override. Same below. Also no ; after {}

[terelius, 2015/07/28 11:52:08]

Done.

+  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

[stefan-webrtc, 2015/07/28 08:15:34]

new line above, possibly

[terelius, 2015/07/28 11:52:09]

Done.

+
+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. TODO

[stefan-webrtc, 2015/07/28 08:15:34]

Owner for TODO

[terelius, 2015/07/28 11:52:08]

Done. Removed empty TODO.

+  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;
+}
+
+}  // Anonymous namespace.

[stefan-webrtc, 2015/07/28 08:15:34]

Remove Anonymous (at least that's how I think we typically write).

[terelius, 2015/07/28 11:52:09]

Done.

+
+// 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