Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/test/cmd_test/voe_cmd_test.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.
  */
 
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
 #ifndef _WIN32
 #include <unistd.h>
 #endif
 
 #include <memory>
 #include <vector>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_dtmf.h"
 #include "webrtc/voice_engine/include/voe_errors.h"
@@ skipping 412 matching lines @@
              option_index++);
       printf("%i. Add an additional file-playing channel \n", option_index++);
       printf("%i. Remove a file-playing channel \n", option_index++);
       printf("%i. Toggle Opus stereo (Opus must be selected again to apply "
              "the setting) \n", option_index++);
       printf("%i. Set Opus maximum playback rate \n", option_index++);
       printf("%i. Toggle Opus DTX \n", option_index++);
       printf("%i. Set bit rate (only take effect on codecs that allow the "
              "change) \n", option_index++);
       printf("%i. Toggle AECdump recording \n", option_index++);
-      printf("%i. Record RtcEventLog file of 30 seconds \n", option_index++);
 
       printf("Select action or %i to stop the call: ", option_index);
       int option_selection;
       ASSERT_EQ(1, scanf("%i", &option_selection));
 
       option_index = num_codecs;
       if (option_selection < option_index) {
         res = codec->GetCodec(option_selection, cinst);
         VALIDATE;
         if (strcmp(cinst.plname, "red") == 0) {
@@ skipping 326 matching lines @@
       } else if (option_selection == option_index++) {
         const char* kDebugFileName = "audio.aecdump";
         if (debug_recording_started) {
           apm->StopDebugRecording();
           printf("Debug recording named %s stopped\n", kDebugFileName);
         } else {
           apm->StartDebugRecording(kDebugFileName);
           printf("Debug recording named %s started\n", kDebugFileName);
         }
         debug_recording_started = !debug_recording_started;
-      } else if (option_selection == option_index++) {
-        const char* kDebugFileName = "eventlog.rel";
-        codec->GetEventLog()->StartLogging(kDebugFileName, 30000);

[the sun, 2016/03/03 09:25:13]

Add a test at the PeerConnection level which exercises the whole stack.

[ivoc, 2016/03/10 13:15:36]

This is a little complicated to do, because there is currently no way to open an
rtc::PlatformFile in WebRTC, which is used on the PC interface to pass in the
file. How do you think I should handle this?

       } else {
         break;
       }
     }
 
     if (debug_recording_started) {
       apm->StopDebugRecording();
     }
 
     if (send) {
@@ skipping 44 matching lines @@
 
   printf("Delete channels \n");
   res = base1->DeleteChannel(chan);
   VALIDATE;
 
   for (int i = 0; i < kMaxNumChannels; ++i) {
     res = base1->DeleteChannel(channels[i]);
     VALIDATE;
   }
 }