Log audio network adapter decisions in event log.

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/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.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"
@@ skipping 37 matching lines @@
                     const uint8_t* header,
                     size_t packet_length) override;
   void LogRtcpPacket(PacketDirection direction,
                      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;
+  void LogAudioEncoderConfig(
+      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
 
  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_;
 
@@ skipping 340 matching lines @@
   std::unique_ptr<rtclog::Event> event(new rtclog::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);
   StoreEvent(&event);
 }
 
+void RtcEventLogImpl::LogAudioEncoderConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  std::unique_ptr<rtclog::Event> event(new rtclog::Event());
+  event->set_timestamp_us(clock_->TimeInMicroseconds());
+  event->set_type(rtclog::Event::AUDIO_ENCODER_CONFIG_EVENT);
+  DebugDumpWriter::ConvertEncoderConfigToDumpEntry(
+      config, event->mutable_encoder_config());
+  StoreEvent(&event);
+}
+
 void RtcEventLogImpl::StoreEvent(std::unique_ptr<rtclog::Event>* event) {
   if (!event_queue_.Insert(event)) {
     LOG(LS_ERROR) << "WebRTC event log queue full. Dropping event.";
   }
   helper_thread_.SignalNewEvent();
 }
 
 bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
                                    rtclog::EventStream* result) {
   char tmp_buffer[1024];
@@ skipping 28 matching lines @@
 #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