Event log cleanup in tests.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 33 matching lines @@
       media_type = MediaType::VIDEO;
     } else {
       media_type = MediaType::AUDIO;
     }
   }
   return receiver_->DeliverPacket(media_type, packet, length, packet_time);
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
+      event_log_(RtcEventLog::CreateNull()),
       video_send_config_(nullptr),
       video_send_stream_(nullptr),
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(1),
       num_audio_streams_(0),
       num_flexfec_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
       fake_send_audio_device_(nullptr),
@@ skipping 390 matching lines @@
 const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
                                                      0xBADCAFF};
 const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
                                                        0xC0FFEF};
 const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
 const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
 const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
 const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
 const int CallTest::kNackRtpHistoryMs = 1000;
 
-BaseTest::BaseTest() {}
+BaseTest::BaseTest() : event_log_(RtcEventLog::CreateNull()) {}
 
-BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
-}
+BaseTest::BaseTest(unsigned int timeout_ms)
+    : RtpRtcpObserver(timeout_ms), event_log_(RtcEventLog::CreateNull()) {}
 
 BaseTest::~BaseTest() {
 }
 
 std::unique_ptr<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() {
   return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000);
 }
 
 std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() {
   return FakeAudioDevice::CreateDiscardRenderer(48000);
 }
 
 void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
                                          FakeAudioDevice* recv_audio_device) {
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
-  return Call::Config(&event_log_);
+  return Call::Config(event_log_.get());
 }
 
 Call::Config BaseTest::GetReceiverCallConfig() {
-  return Call::Config(&event_log_);
+  return Call::Config(event_log_.get());
 }
 
 void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
 }
 
 MediaType BaseTest::SelectMediaType() {
   if (GetNumVideoStreams() > 0) {
     if (GetNumAudioStreams() > 0) {
       // Relies on PayloadDemuxer to set media type from payload type.
       return MediaType::ANY;
@@ skipping 74 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc