Add RTCConfiguration getter and setter methods.

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 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.
  */
 
@@ skipping 205 matching lines @@
     kTcpCandidatePolicyDisabled
   };
 
   enum ContinualGatheringPolicy {
     GATHER_ONCE,
     GATHER_CONTINUALLY
   };
 
   // TODO(hbos): Change into class with private data and public getters.
   struct RTCConfiguration {
+    // This struct is subject to reorganization, both for naming
+    // consistency, and to group settings to match where they are used
+    // in the implementation. To do that, we need getter and setter
+    // methods for all settings which are of interest to applications,
+    // Chrome in particular.
+
+    bool dscp() { return enable_dscp.value_or(false); }
+    void set_dscp(bool enable) { enable_dscp = rtc::Optional<bool>(enable); }
+
+    // TODO(nisse): The corresponding flag in MediaConfig and
+    // elsewhere should be renamed enable_cpu_adaptation.
+    bool cpu_adaptation() { return cpu_overuse_detection.value_or(true); }
+    void set_cpu_adaptation(bool enable) {
+      cpu_overuse_detection = rtc::Optional<bool>(enable);
+    }
+
+    // TODO(nisse): Currently no getter method, since it collides with
+    // the flag itself. Add when the flag is moved to MediaConfig.
+    void set_suspend_below_min_bitrate(bool enable) {
+      suspend_below_min_bitrate = rtc::Optional<bool>(enable);
+    }
+
+    // TODO(nisse): The negation in the corresponding MediaConfig
+    // attribute is inconsistent, and it should be renamed at some
+    // point.
+    bool prerenderer_smoothing() { return !disable_prerenderer_smoothing; }
+    void set_prerenderer_smoothing(bool enable) {
+      disable_prerenderer_smoothing = !enable;
+    }
+
     static const int kUndefined = -1;
     // Default maximum number of packets in the audio jitter buffer.
     static const int kAudioJitterBufferMaxPackets = 50;
     // TODO(pthatcher): Rename this ice_transport_type, but update
     // Chromium at the same time.
     IceTransportsType type;
     // TODO(pthatcher): Rename this ice_servers, but update Chromium
     // at the same time.
     IceServers servers;
     BundlePolicy bundle_policy;
@@ skipping 421 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_