Wire up bandwidth stats to the new API and webrtcvideoengine2.

Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video/call.cc

diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc
index b03b6c9..2b4f76f 100644
--- a/webrtc/video/call.cc
+++ b/webrtc/video/call.cc
@@ -114,8 +114,7 @@
   virtual void DestroyVideoReceiveStream(
       webrtc::VideoReceiveStream* receive_stream) OVERRIDE;
 
-  virtual uint32_t SendBitrateEstimate() OVERRIDE;
-  virtual uint32_t ReceiveBitrateEstimate() OVERRIDE;
+  virtual Stats GetStats() const OVERRIDE;
 
   virtual DeliveryStatus DeliverPacket(const uint8_t* packet,
                                        size_t length) OVERRIDE;
@@ -321,14 +320,26 @@
   delete receive_stream_impl;
 }
 
-uint32_t Call::SendBitrateEstimate() {
-  // TODO(pbos): Return send-bitrate estimate
-  return 0;
-}
-
-uint32_t Call::ReceiveBitrateEstimate() {
-  // TODO(pbos): Return receive-bitrate estimate
-  return 0;
+Call::Stats Call::GetStats() const {
+  Stats stats;
+  // Ignoring return values.
+  uint32_t send_bandwidth = 0;
+  rtp_rtcp_->GetEstimatedSendBandwidth(base_channel_id_, &send_bandwidth);
+  stats.send_bandwidth_bps = send_bandwidth;
+  uint32_t recv_bandwidth = 0;
+  rtp_rtcp_->GetEstimatedReceiveBandwidth(base_channel_id_, &recv_bandwidth);
+  stats.recv_bandwidth_bps = recv_bandwidth;
+  {
+    ReadLockScoped read_lock(*send_crit_);
+    for (std::map<uint32_t, VideoSendStream*>::const_iterator it =
+             send_ssrcs_.begin();
+         it != send_ssrcs_.end();
+         ++it) {
+      stats.pacer_delay_ms =
+          std::max(it->second->GetPacerQueuingDelayMs(), stats.pacer_delay_ms);
+    }
+  }
+  return stats;
 }
 
 void Call::SignalNetworkState(NetworkState state) {