Index: webrtc/api/peerconnectioninterface.h |
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h |
index 08a131920e3be649746ebbc083cdf8b4f64d7a69..9259275b86a99071131fc5beeb9b422c71920269 100644 |
--- a/webrtc/api/peerconnectioninterface.h |
+++ b/webrtc/api/peerconnectioninterface.h |
@@ -223,6 +223,36 @@ class PeerConnectionInterface : public rtc::RefCountInterface { |
// 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; |