Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/channel.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/call/rtc_event_log_proxy.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
@@ skipping 143 matching lines @@
 {
  public:
   friend class VoERtcpObserver;
 
   enum { KNumSocketThreads = 1 };
   enum { KNumberOfSocketBuffers = 8 };
   virtual ~Channel();
   static int32_t CreateChannel(Channel*& channel,
                                int32_t channelId,
                                uint32_t instanceId,
-                               RtcEventLog* const event_log,
                                const Config& config);
   Channel(int32_t channelId,
           uint32_t instanceId,
-          RtcEventLog* const event_log,
           const Config& config);
   int32_t Init();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
                                TransmitMixer& transmitMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect);
   int32_t UpdateLocalTimeStamp();
@@ skipping 252 matching lines @@
   // Used for obtaining RTT for a receive-only channel.
   void set_associate_send_channel(const ChannelOwner& channel) {
     assert(_channelId != channel.channel()->ChannelId());
     rtc::CritScope lock(&assoc_send_channel_lock_);
     associate_send_channel_ = channel;
   }
 
   // Disassociate a send channel if it was associated.
   void DisassociateSendChannel(int channel_id);
 
+  // Set a RtcEventLog logging object.
+  void SetRtcEventLog(RtcEventLog* event_log) {
+    event_log_.SetEventLog(event_log);
+  }
+
  protected:
   void OnIncomingFractionLoss(int fraction_lost);
 
  private:
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   bool HandleRtxPacket(const uint8_t* packet,
                        size_t packet_length,
@@ skipping 17 matching lines @@
   int64_t GetRTT(bool allow_associate_channel) const;
 
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   int32_t _channelId;
 
   ChannelState channel_state_;
 
-  RtcEventLog* const event_log_;
+  RtcEventLogProxy event_log_;
 
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<StatisticsProxy> statistics_proxy_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
   TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   std::unique_ptr<AudioSinkInterface> audio_sink_;
@@ skipping 89 matching lines @@
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_