Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/test/auto_test/standard/codec_test.cc

Index: webrtc/voice_engine/test/auto_test/standard/codec_test.cc
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
index 3a3d83031d3e932661ecef1b4fb15c060c757cae..5a500af2df4f256b46e2d3dc31fec8d7150ffd87 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_test.cc
@@ -176,30 +176,6 @@ TEST_F(CodecTest, OpusDtxCannotBeSetForNonOpus) {
   }
 }
 
-#ifdef ENABLE_RTC_EVENT_LOG
-TEST_F(CodecTest, RtcEventLogIntegrationTest) {

[terelius, 2016/03/21 20:47:46]

Should there be an integration test for the new API too, or is it so simple that
isn't needed?

[ivoc, 2016/03/22 13:44:55]

I would like to add one, but there is currently no support for opening platform
files in WebRTC. It would be a good idea to try to add this in a follow-up CL.

-  webrtc::RtcEventLog* event_log = voe_codec_->GetEventLog();
-  ASSERT_TRUE(event_log);
-
-  // Find the name of the current test, in order to use it as a temporary
-  // filename.
-  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
-  const std::string temp_filename = webrtc::test::OutputPath() +
-                                    test_info->test_case_name() +
-                                    test_info->name();
-  // Create a log file.
-  event_log->StartLogging(temp_filename, 1000);
-  event_log->StopLogging();
-
-  // Check if the file has been created.
-  FILE* event_file = fopen(temp_filename.c_str(), "r");
-  ASSERT_TRUE(event_file);
-  fclose(event_file);
-  // Remove the temporary file.
-  remove(temp_filename.c_str());
-}
-#endif  // ENABLE_RTC_EVENT_LOG
-
 // TODO(xians, phoglund): Re-enable when issue 372 is resolved.
 TEST_F(CodecTest, DISABLED_ManualVerifySendCodecsForAllPacketSizes) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {