Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead

webrtc/modules/rtp_rtcp/source/rtp_sender.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/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <algorithm>
 #include <utility>
 
+#include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/rtc_base/arraysize.h"
 #include "webrtc/rtc_base/checks.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/safe_minmax.h"
 #include "webrtc/rtc_base/timeutils.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
@@ skipping 596 matching lines @@
 bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
                                     const PacketOptions& options,
                                     const PacedPacketInfo& pacing_info) {
   int bytes_sent = -1;
   if (transport_) {
     UpdateRtpOverhead(packet);
     bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
                      ? static_cast<int>(packet.size())
                      : -1;
     if (event_log_ && bytes_sent > 0) {
-      event_log_->LogOutgoingRtpHeader(packet, pacing_info.probe_cluster_id);
+      event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(
+          packet, pacing_info.probe_cluster_id));
     }
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", packet.size(),
                        "sent", bytes_sent);
   // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
   if (bytes_sent <= 0) {
     LOG(LS_WARNING) << "Transport failed to send packet.";
     return false;
   }
@@ skipping 649 matching lines @@
     rtc::CritScope lock(&send_critsect_);
     packet->SetTimestamp(last_rtp_timestamp_);
     packet->set_capture_time_ms(capture_time_ms_);
   }
   AssignSequenceNumber(packet.get());
   SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
                 RtpPacketSender::Priority::kLowPriority);
 }
 
 }  // namespace webrtc