Drop calls to Trace::CreateTrace and Trace::ReturnTrace from Call.

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.
  */
 
@@ skipping 34 matching lines @@
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/task_queue.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/rtc_base/thread_checker.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 #include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/send_delay_stats.h"
 #include "webrtc/video/stats_counter.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/video/video_send_stream.h"
 
 namespace webrtc {
 
 namespace {
 
@@ skipping 351 matching lines @@
       worker_queue_("call_worker_queue"),
       base_bitrate_config_(config.bitrate_config) {
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
-  Trace::CreateTrace();
   transport_send->send_side_cc()->RegisterNetworkObserver(this);
   transport_send_ = std::move(transport_send);
   transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
   transport_send_->send_side_cc()->SetBweBitrates(
       config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
   call_stats_->RegisterStatsObserver(&receive_side_cc_);
   call_stats_->RegisterStatsObserver(transport_send_->send_side_cc());
 
@@ skipping 40 matching lines @@
   int64_t first_sent_packet_ms =
       transport_send_->send_side_cc()->GetFirstPacketTimeMs();
   // Only update histograms after process threads have been shut down, so that
   // they won't try to concurrently update stats.
   {
     rtc::CritScope lock(&bitrate_crit_);
     UpdateSendHistograms(first_sent_packet_ms);
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
-
-  Trace::ReturnTrace();
 }
 
 rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket(
     const uint8_t* packet,
     size_t length,
     const PacketTime* packet_time) const {
   RtpPacketReceived parsed_packet;
   if (!parsed_packet.Parse(packet, length))
     return rtc::Optional<RtpPacketReceived>();
 
@@ skipping 922 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