C++11 in-class member initialization in Call configs.

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_
@@ skipping 55 matching lines @@
 // 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:
   enum NetworkState {
     kNetworkUp,
     kNetworkDown,
   };
   struct Config {
+    Config() = delete;
     explicit Config(newapi::Transport* send_transport)
-        : send_transport(send_transport),
-          voice_engine(NULL),
-          overuse_callback(NULL) {}
+        : send_transport(send_transport) {}
 
     static const int kDefaultStartBitrateBps;
 
     // TODO(solenberg): Need to add media type to the interface for outgoing
     // packets too.
-    newapi::Transport* send_transport;
+    newapi::Transport* send_transport = nullptr;
 
     // VoiceEngine used for audio/video synchronization for this Call.
-    VoiceEngine* voice_engine;
+    VoiceEngine* voice_engine = nullptr;
 
     // Callback for overuse and normal usage based on the jitter of incoming
-    // captured frames. 'NULL' disables the callback.
-    LoadObserver* overuse_callback;
+    // captured frames. 'nullptr' disables the callback.
+    LoadObserver* overuse_callback = nullptr;
 
     // Bitrate config used until valid bitrate estimates are calculated. Also
     // used to cap total bitrate used.
     struct BitrateConfig {
-      BitrateConfig()
-          : min_bitrate_bps(0),
-            start_bitrate_bps(kDefaultStartBitrateBps),
-            max_bitrate_bps(-1) {}
-      int min_bitrate_bps;
-      int start_bitrate_bps;
-      int max_bitrate_bps;
+      int min_bitrate_bps = 0;
+      int start_bitrate_bps = kDefaultStartBitrateBps;
+      int max_bitrate_bps = -1;
     } bitrate_config;
 
     struct AudioConfig {
-      AudioDeviceModule* audio_device_manager;
-      AudioProcessing* audio_processing;
-      VoiceEngineObserver* voice_engine_observer;
+      AudioDeviceModule* audio_device_manager = nullptr;
+      AudioProcessing* audio_processing = nullptr;
+      VoiceEngineObserver* voice_engine_observer = nullptr;
     } audio_config;
   };
 
   struct Stats {
-    Stats()
-        : send_bandwidth_bps(0),
-          recv_bandwidth_bps(0),
-          pacer_delay_ms(0),
-          rtt_ms(-1) {}
-
-    int send_bandwidth_bps;
-    int recv_bandwidth_bps;
-    int64_t pacer_delay_ms;
-    int64_t rtt_ms;
+    int send_bandwidth_bps = 0;
+    int recv_bandwidth_bps = 0;
+    int64_t pacer_delay_ms = 0;
+    int64_t rtt_ms = -1;
   };
 
   static Call* Create(const Call::Config& config);
 
   virtual AudioSendStream* CreateAudioSendStream(
       const AudioSendStream::Config& config) = 0;
   virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
 
   virtual AudioReceiveStream* CreateAudioReceiveStream(
       const AudioReceiveStream::Config& config) = 0;
@@ skipping 26 matching lines @@
   // implemented.
   virtual void SetBitrateConfig(
       const Config::BitrateConfig& bitrate_config) = 0;
   virtual void SignalNetworkState(NetworkState state) = 0;
 
   virtual ~Call() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_H_