Refactoring: move ownership of RtcEventLog from Call to PeerConnection

webrtc/call.h

 /*
  *  Copyright (c) 2013 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_CALL_H_
 #define WEBRTC_CALL_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/api/call/audio_receive_stream.h"
 #include "webrtc/api/call/audio_send_stream.h"
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/common_types.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 class AudioProcessing;
+class RtcEventLog;
 
 const char* Version();
 
 enum class MediaType {
   ANY,
   AUDIO,
   VIDEO,
   DATA
 };
 
@@ skipping 26 matching lines @@
  protected:
   virtual ~LoadObserver() {}
 };
 
 // A Call instance can contain several send and/or receive streams. All streams
 // are assumed to have the same remote endpoint and will share bitrate estimates
 // etc.
 class Call {
  public:
   struct Config {
+    explicit Config(RtcEventLog* event_log) : event_log(event_log) {
+      RTC_DCHECK(event_log);
+    }
+
     static const int kDefaultStartBitrateBps;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
       int min_bitrate_bps = 0;
       int start_bitrate_bps = kDefaultStartBitrateBps;
       int max_bitrate_bps = -1;
     } bitrate_config;
 
     // AudioState which is possibly shared between multiple calls.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     rtc::scoped_refptr<AudioState> audio_state;
 
     // Audio Processing Module to be used in this call.
     // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
     AudioProcessing* audio_processing = nullptr;
+
+    // RtcEventLog to use for this call. Required.
+    // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+    RtcEventLog* event_log = nullptr;
   };
 
   struct Stats {
     std::string ToString(int64_t time_ms) const;
 
     int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
     int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
     int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
     int64_t pacer_delay_ms = 0;
     int64_t rtt_ms = -1;
@@ skipping 42 matching lines @@
   // for each stream separately. Right now it's global per media type.
   virtual void SignalChannelNetworkState(MediaType media,
                                          NetworkState state) = 0;
 
   virtual void OnNetworkRouteChanged(
       const std::string& transport_name,
       const rtc::NetworkRoute& network_route) = 0;
 
   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
 
-  virtual bool StartEventLog(rtc::PlatformFile log_file,
-                             int64_t max_size_bytes) = 0;
-  virtual void StopEventLog() = 0;
-
   virtual ~Call() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_