Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 <string.h>
 #include <algorithm>
 #include <map>
 #include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/optional.h"
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/audio/time_interval.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream_impl.h"
 #include "webrtc/call/rtp_stream_receiver_controller.h"
 #include "webrtc/call/rtp_transport_controller_send.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_rtcp_packet_incoming.h"
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.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/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/utility/include/process_thread.h"
@@ skipping 562 matching lines @@
 
 PacketReceiver* Call::Receiver() {
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
-  event_log_->LogAudioSendStreamConfig(*CreateRtcLogStreamConfig(config));
+  event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
+      CreateRtcLogStreamConfig(config)));
 
   rtc::Optional<RtpState> suspended_rtp_state;
   {
     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
     if (iter != suspended_audio_send_ssrcs_.end()) {
       suspended_rtp_state.emplace(iter->second);
     }
   }
 
   AudioSendStream* send_stream = new AudioSendStream(
@@ skipping 45 matching lines @@
   }
   UpdateAggregateNetworkState();
   sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
   delete send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
-  event_log_->LogAudioReceiveStreamConfig(*CreateRtcLogStreamConfig(config));
+  event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>(
+      CreateRtcLogStreamConfig(config)));
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       &audio_receiver_controller_, transport_send_->packet_router(), config,
       config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     receive_rtp_config_[config.rtp.remote_ssrc] =
         ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
     audio_receive_streams_.insert(receive_stream);
 
     ConfigureSync(config.sync_group);
@@ skipping 39 matching lines @@
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
     VideoEncoderConfig encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
 
   video_send_delay_stats_->AddSsrcs(config);
   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
        ++ssrc_index) {
-    event_log_->LogVideoSendStreamConfig(
-        *CreateRtcLogStreamConfig(config, ssrc_index));
+    event_log_->Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>(
+        CreateRtcLogStreamConfig(config, ssrc_index)));
   }
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   // Copy ssrcs from |config| since |config| is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
       call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
       video_send_delay_stats_.get(), event_log_, std::move(config),
@@ skipping 69 matching lines @@
       // type, we may get an incorrect value for the rtx stream, but
       // that is unlikely to matter in practice.
       receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
     }
     receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
     video_receive_streams_.insert(receive_stream);
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
-  event_log_->LogVideoReceiveStreamConfig(*CreateRtcLogStreamConfig(config));
+  event_log_->Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>(
+      CreateRtcLogStreamConfig(config)));
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl =
       static_cast<VideoReceiveStream*>(receive_stream);
@@ skipping 455 matching lines @@
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
     ReadLockScoped read_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
       if (kv.second->DeliverRtcp(packet, length))
         rtcp_delivered = true;
     }
   }
 
-  if (rtcp_delivered)
-    event_log_->LogIncomingRtcpPacket(rtc::MakeArrayView(packet, length));
+  if (rtcp_delivered) {
+    event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(
+        rtc::MakeArrayView(packet, length)));
+  }
 
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
 
@@ skipping 28 matching lines @@
     return DELIVERY_UNKNOWN_SSRC;
   }
   parsed_packet->IdentifyExtensions(it->second.extensions);
 
   NotifyBweOfReceivedPacket(*parsed_packet, media_type);
 
   if (media_type == MediaType::AUDIO) {
     if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
-      event_log_->LogIncomingRtpHeader(*parsed_packet);
+      event_log_->Log(
+          rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet));
       const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
       if (!first_received_rtp_audio_ms_) {
         first_received_rtp_audio_ms_.emplace(arrival_time_ms);
       }
       last_received_rtp_audio_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   } else if (media_type == MediaType::VIDEO) {
     if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
-      event_log_->LogIncomingRtpHeader(*parsed_packet);
+      event_log_->Log(
+          rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet));
       const int64_t arrival_time_ms = parsed_packet->arrival_time_ms();
       if (!first_received_rtp_video_ms_) {
         first_received_rtp_video_ms_.emplace(arrival_time_ms);
       }
       last_received_rtp_video_ms_.emplace(arrival_time_ms);
       return DELIVERY_OK;
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
@@ skipping 61 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc