Move RtcEventLog object from inside VoiceEngine to Call.

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.
  */
 
@@ skipping 96 matching lines @@
 RTPSender::RTPSender(
     bool audio,
     Clock* clock,
     Transport* transport,
     RtpAudioFeedback* audio_feedback,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
-    SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log)
+    SendSideDelayObserver* send_side_delay_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       random_(clock_->TimeInMicroseconds()),
       bitrates_(bitrate_callback),
       total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
@@ skipping 17 matching lines @@
       // NACK.
       nack_byte_count_times_(),
       nack_byte_count_(),
       nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()),
       packet_history_(clock),
       // Statistics
       statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
-      event_log_(event_log),
+      event_log_crit_(CriticalSectionWrapper::CreateCriticalSection()),
+      event_log_(nullptr),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
       ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
       remote_ssrc_(0),
       sequence_number_forced_(false),
       ssrc_forced_(false),
       timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
@@ skipping 583 matching lines @@
 }
 
 bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
                                     size_t size,
                                     const PacketOptions& options) {
   int bytes_sent = -1;
   if (transport_) {
     bytes_sent = transport_->SendRtp(packet, size, options)
                      ? static_cast<int>(size)
                      : -1;
+    CriticalSectionScoped lock(event_log_crit_.get());
     if (event_log_ && bytes_sent > 0) {
       event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet, size);
     }
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", 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";
@@ skipping 1140 matching lines @@
 RtpState RTPSender::GetRtxRtpState() const {
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
+void RTPSender::SetRtcEventLog(webrtc::RtcEventLog* event_log) {
+  CriticalSectionScoped lock(event_log_crit_.get());
+  event_log_ = event_log;
+}
 }  // namespace webrtc