Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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/voice_engine/channel.h"
 
 #include <algorithm>
 #include <map>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/array_view.h"
 #include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+// TODO(eladalon): Remove events/* after removing the deprecated functions.
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_logging_started.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_logging_stopped.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/format_macros.h"
 #include "webrtc/rtc_base/location.h"
 #include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/rate_limiter.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/output_mixer.h"
 #include "webrtc/voice_engine/statistics.h"
 #include "webrtc/voice_engine/utility.h"
@@ skipping 22 matching lines @@
   }
 
   bool StartLogging(rtc::PlatformFile log_file,
                     int64_t max_size_bytes) override {
     RTC_NOTREACHED();
     return false;
   }
 
   void StopLogging() override { RTC_NOTREACHED(); }
 
+  void Log(std::unique_ptr<RtcEvent> event) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->Log(std::move(event));
+    }
+  }
+
   void LogVideoReceiveStreamConfig(
       const webrtc::rtclog::StreamConfig&) override {
     RTC_NOTREACHED();
   }
 
   void LogVideoSendStreamConfig(const webrtc::rtclog::StreamConfig&) override {
     RTC_NOTREACHED();
   }
 
   void LogAudioReceiveStreamConfig(
       const webrtc::rtclog::StreamConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogAudioReceiveStreamConfig(config);
+      event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>(
+          rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config)));
     }
   }
 
   void LogAudioSendStreamConfig(
       const webrtc::rtclog::StreamConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogAudioSendStreamConfig(config);
+      event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
+          rtc::MakeUnique<webrtc::rtclog::StreamConfig>(config)));
     }
   }
 
   void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogIncomingRtpHeader(packet);
+      event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketIncoming>(packet));
     }
   }
 
   void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
                             int probe_cluster_id) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogOutgoingRtpHeader(packet, probe_cluster_id);
+      event_log_->Log(
+          rtc::MakeUnique<RtcEventRtpPacketOutgoing>(packet, probe_cluster_id));
     }
   }
 
   void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogIncomingRtcpPacket(packet);
+      event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(packet));
     }
   }
 
   void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogOutgoingRtcpPacket(packet);
+      event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
     }
   }
 
   void LogAudioPlayout(uint32_t ssrc) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogAudioPlayout(ssrc);
+      event_log_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc));
     }
   }
 
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogLossBasedBweUpdate(bitrate_bps, fraction_loss,
-                                        total_packets);
+      event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(
+          bitrate_bps, fraction_loss, total_packets));
     }
   }
 
   void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogDelayBasedBweUpdate(bitrate_bps, detector_state);
+      event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
+          bitrate_bps, detector_state));
     }
   }
 
   void LogAudioNetworkAdaptation(
       const AudioEncoderRuntimeConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogAudioNetworkAdaptation(config);
+      event_log_->Log(rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(
+          rtc::MakeUnique<AudioEncoderRuntimeConfig>(config)));
     }
   }
 
   void LogProbeClusterCreated(int id,
                               int bitrate_bps,
                               int min_probes,
                               int min_bytes) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogProbeClusterCreated(id, bitrate_bps, min_probes,
-                                         min_bytes);
+      event_log_->Log(rtc::MakeUnique<RtcEventProbeClusterCreated>(
+          id, bitrate_bps, min_probes, min_bytes));
     }
   };
 
   void LogProbeResultSuccess(int id, int bitrate_bps) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogProbeResultSuccess(id, bitrate_bps);
+      event_log_->Log(
+          rtc::MakeUnique<RtcEventProbeResultSuccess>(id, bitrate_bps));
     }
   };
 
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogProbeResultFailure(id, failure_reason);
+      event_log_->Log(
+          rtc::MakeUnique<RtcEventProbeResultFailure>(id, failure_reason));
     }
   };
 
   void SetEventLog(RtcEventLog* event_log) {
     rtc::CritScope lock(&crit_);
     event_log_ = event_log;
   }
 
  private:
   rtc::CriticalSection crit_;
@@ skipping 2937 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc