Add task queue to Call.

webrtc/config.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.
  */
 
@@ skipping 106 matching lines @@
   // The VideoEncoder may redistribute bitrates over the temporal layers so a
   // bitrate threshold of 100k and an estimate of 105k does not imply that we
   // get 100k in one temporal layer and 5k in the other, just that the bitrate
   // in the first temporal layer should not exceed 100k.
   // TODO(pbos): Apart from a special case for two-layer screencast these
   // thresholds are not propagated to the VideoEncoder. To be implemented.
   std::vector<int> temporal_layer_thresholds_bps;
 };
 
 struct VideoEncoderConfig {
+ public:
   enum class ContentType {
     kRealtimeVideo,
     kScreen,
   };
 
+  VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
+  VideoEncoderConfig& operator=(const VideoEncoderConfig&) = delete;
+
+  // Mostly used by tests.  Avoid creating copies if you can.
+  VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
+
   VideoEncoderConfig();
+  VideoEncoderConfig(VideoEncoderConfig&&) = default;
   ~VideoEncoderConfig();
   std::string ToString() const;
 
   std::vector<VideoStream> streams;
   std::vector<SpatialLayer> spatial_layers;
   ContentType content_type;
   void* encoder_specific_settings;
 
   // Padding will be used up to this bitrate regardless of the bitrate produced
   // by the encoder. Padding above what's actually produced by the encoder helps
   // maintaining a higher bitrate estimate. Padding will however not be sent
   // unless the estimated bandwidth indicates that the link can handle it.
   int min_transmit_bitrate_bps;
   bool expect_encode_from_texture;
+
+ private:
+  // Access to the copy constructor is private to force use of the Copy()
+  // method for those exceptional cases where we do use it.
+  VideoEncoderConfig(const VideoEncoderConfig&) = default;
 };
 
 // Controls the capacity of the packet buffer in NetEq. The capacity is the
 // maximum number of packets that the buffer can contain. If the limit is
 // exceeded, the buffer will be flushed. The capacity does not affect the actual
 // audio delay in the general case, since this is governed by the target buffer
 // level (calculated from the jitter profile). It is only in the rare case of
 // severe network freezes that a higher capacity will lead to a (transient)
 // increase in audio delay.
 struct NetEqCapacityConfig {
@@ skipping 14 matching lines @@
 struct VoicePacing {
   VoicePacing() : enabled(false) {}
   explicit VoicePacing(bool value) : enabled(value) {}
   static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
   bool enabled;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CONFIG_H_