diff --git a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
index c4bec98ed8b6..8bf6410d262e 100644
--- a/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
+++ b/media/webrtc/signaling/src/media-conduit/AudioConduit.cpp
@@ -226,12 +226,9 @@ bool WebrtcAudioConduit::GetAVStats(int32_t* jitterBufferDelayMs,
 bool WebrtcAudioConduit::GetRTPStats(unsigned int* jitterMs,
                                      unsigned int* cumulativeLost) {
   ASSERT_ON_THREAD(mStsThread);
-  unsigned int maxJitterMs = 0;
-  unsigned int discardedPackets;
   *jitterMs = 0;
   *cumulativeLost = 0;
-  return !mSendChannelProxy->GetRTPStatistics(*jitterMs, maxJitterMs,
-                                              discardedPackets, *cumulativeLost);
+  return !mSendChannelProxy->GetRTPStatistics(*jitterMs, *cumulativeLost);
 }
 
 DOMHighResTimeStamp
diff --git a/media/webrtc/trunk/webrtc/voice_engine/channel.h b/media/webrtc/trunk/webrtc/voice_engine/channel.h
index cd00e7a701bf..157d729b4b92 100644
--- a/media/webrtc/trunk/webrtc/voice_engine/channel.h
+++ b/media/webrtc/trunk/webrtc/voice_engine/channel.h
@@ -357,6 +357,13 @@ class Channel
     return rtp_receiver_->GetSources();
   }
 
+  int GetPlayoutFrequency() const {
+    if (audio_coding_) {
+      return audio_coding_->PlayoutFrequency();
+    }
+    return 0;
+  }
+
  private:
   class ProcessAndEncodeAudioTask;