Expose a function for setting bandwidth estimation parameters in ViERTP_RTCP.

Review URL: https://webrtc-codereview.appspot.com/678007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2591 4adac7df-926f-26a2-2b94-8c16560cd09d

src/video_engine/include/vie_rtp_rtcp.h

@@ -256,6 +256,16 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
       const int video_channel,
       unsigned int* estimated_bandwidth) const = 0;
 
+  // This function sets various options for the bandwidth estimator
+  // code.  The options are applied to new channels only.  For a given
+  // channel, the options that are active at the time when the channel
+  // is created are immutable for that channel.  See
+  // http://tools.ietf.org/html/draft-alvestrand-rtcweb-congestion-02
+  // (or later, updated documentation) and common_types.h to get a
+  // feel for what the options do.
+  virtual int SetOverUseDetectorOptions(
+      const OverUseDetectorOptions& options) const = 0;
+
   // This function enables capturing of RTP packets to a binary file on a
   // specific channel and for a given direction. The file can later be
   // replayed using e.g. RTP Tools rtpplay since the binary file format is

src/video_engine/vie_rtp_rtcp_impl.cc

@@ -761,6 +761,22 @@ int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
       static_cast<WebRtc_UWord32*>(estimated_bandwidth));
 }
 
+int ViERTP_RTCPImpl::SetOverUseDetectorOptions(
+    const OverUseDetectorOptions& options) const {
+  if (!shared_data_->Initialized()) {
+    shared_data_->SetLastError(kViENotInitialized);
+    WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
+                 "%s - ViE instance %d not initialized", __FUNCTION__,
+                 shared_data_->instance_id());
+    return -1;
+  }
+  // Lock the channel manager to avoid creating a channel with
+  // "undefined" bwe settings (atomic copy).
+  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
+  shared_data_->SetOverUseDetectorOptions(options);
+  return 0;
+}
+
 int ViERTP_RTCPImpl::StartRTPDump(const int video_channel,
                                   const char file_nameUTF8[1024],
                                   RTPDirections direction) {

src/video_engine/vie_rtp_rtcp_impl.h

@@ -95,6 +95,8 @@ class ViERTP_RTCPImpl
   virtual int GetEstimatedReceiveBandwidth(
       const int video_channel,
       unsigned int* estimated_bandwidth) const;
+  virtual int SetOverUseDetectorOptions(
+      const OverUseDetectorOptions& options) const;
   virtual int StartRTPDump(const int video_channel,
                            const char file_nameUTF8[1024],
                            RTPDirections direction);

src/video_engine/vie_shared_data.cc

@@ -75,6 +75,11 @@ int ViESharedData::LastErrorInternal() const {
   return error;
 }
 
+void ViESharedData::SetOverUseDetectorOptions(
+    const OverUseDetectorOptions& options) {
+  over_use_detector_options_ = options;
+}
+
 int ViESharedData::NumberOfCores() const {
   return number_cores_;
 }

src/video_engine/vie_shared_data.h

@@ -34,7 +34,7 @@ class ViESharedData {
   int SetUnInitialized();
   void SetLastError(const int error) const;
   int LastErrorInternal() const;
-
+  void SetOverUseDetectorOptions(const OverUseDetectorOptions& options);
   int NumberOfCores() const;
 
   int instance_id() { return instance_id_;}