Probing EndToEndTests.

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 401 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(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
 }
 
 BaseTest::~BaseTest() {
 }
 
 Call::Config BaseTest::GetSenderCallConfig() {
   return Call::Config(&event_log_);
 }
 
@@ skipping 57 matching lines @@
     FrameGeneratorCapturer* frame_generator_capturer) {
 }
 
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool SendTest::ShouldCreateReceivers() const {
   return false;
 }
 
+EndToEndTest::EndToEndTest() {}
+
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc