Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.

webrtc/logging/rtc_event_log/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/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.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/clock.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:
-  explicit RtcEventLogImpl(const Clock* clock);
+  explicit RtcEventLogImpl();

[the sun, 2016/11/18 15:43:59]

no need for explicit

   ~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;
@@ skipping 15 matching lines @@
 
  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_;
 
-  const Clock* const clock_;
-
   RtcEventLogHelperThread helper_thread_;
   rtc::ThreadChecker thread_checker_;
-
-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtcEventLogImpl);

[the sun, 2016/11/18 15:43:59]

Why'd this go?

[nisse-webrtc, 2016/11/21 07:58:02]

Because otherwise it failed to compile... As far as I understood, the expansion
of this macro seemed to collide with the new default constructor
RtcEventLogImpl().

What should I do to be able to keep this?

[the sun, 2016/11/21 08:35:01]

Oh, sorry for being inattentive. You should use RTC_DISALLOW_COPY_AND_ASSIGN()
instead.

[nisse-webrtc, 2016/11/28 14:26:41]

Done.

 };
 
 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:
@@ skipping 23 matching lines @@
   return rtclog::ANY;
 }
 
 // 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(const Clock* clock)
+RtcEventLogImpl::RtcEventLogImpl()
     // Allocate buffers for roughly one second of history.
     : message_queue_(kControlMessagesPerSecond),
       event_queue_(kEventsPerSecond),
-      clock_(clock),
-      helper_thread_(&message_queue_, &event_queue_, clock),
+      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 = clock_->TimeInMicroseconds();
+  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 = clock_->TimeInMicroseconds();
+  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 = clock_->TimeInMicroseconds();
+  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(clock_->TimeInMicroseconds());
+  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);
 
@@ skipping 15 matching lines @@
     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(clock_->TimeInMicroseconds());
+  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(clock_->TimeInMicroseconds());
+  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(clock_->TimeInMicroseconds());
+  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);
@@ skipping 17 matching lines @@
 
   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(clock_->TimeInMicroseconds());
+  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(clock_->TimeInMicroseconds());
+  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;
@@ skipping 26 matching lines @@
     }
 
     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(clock_->TimeInMicroseconds());
+  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(clock_->TimeInMicroseconds());
+  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::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
   if (!event_queue_.Insert(event)) {
@@ skipping 23 matching lines @@
 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(const Clock* clock) {
+std::unique_ptr<RtcEventLog> RtcEventLog::Create() {
 #ifdef ENABLE_RTC_EVENT_LOG
-  return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl(clock));
+  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