diff --git a/content/html/content/src/HTMLMediaElement.cpp b/content/html/content/src/HTMLMediaElement.cpp
index ffd45fd77177..f978268dbc90 100644
--- a/content/html/content/src/HTMLMediaElement.cpp
+++ b/content/html/content/src/HTMLMediaElement.cpp
@@ -1763,7 +1763,29 @@ HTMLMediaElement::CaptureStreamInternal(bool aFinishWhenEnded)
   }
 
   OutputMediaStream* out = mOutputStreams.AppendElement();
+#ifdef DEBUG
+  // Estimate hints based on the type of the media element
+  // under the preference media.capturestream_hints for the
+  // debug builds only. This allows WebRTC Peer Connection
+  // to behave appropriately when media streams generated
+  // via mozCaptureStream*() are added to the Peer Connection.
+  // This functionality is planned to be used as part of Audio
+  // Quality Performance testing for WebRTC.
+  // Bug932845: Revisit this once hints mechanism is dealt with
+  // holistically.
+  uint8_t hints = 0;
+  if (Preferences::GetBool("media.capturestream_hints.enabled")) {
+    nsCOMPtr<nsIDOMHTMLVideoElement> video = do_QueryObject(this);
+    if (video && GetVideoFrameContainer()) {
+      hints = DOMMediaStream::HINT_CONTENTS_VIDEO | DOMMediaStream::HINT_CONTENTS_AUDIO;
+    } else {
+      hints = DOMMediaStream::HINT_CONTENTS_AUDIO;
+    }
+  }
+  out->mStream = DOMMediaStream::CreateTrackUnionStream(window, hints);
+#else
   out->mStream = DOMMediaStream::CreateTrackUnionStream(window);
+#endif
   nsRefPtr<nsIPrincipal> principal = GetCurrentPrincipal();
   out->mStream->CombineWithPrincipal(principal);
   out->mFinishWhenEnded = aFinishWhenEnded;
diff --git a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
index 5e9e93f8f087..400a77519e8c 100644
--- a/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
+++ b/media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp
@@ -758,9 +758,11 @@ void MediaPipelineTransmit::PipelineListener::ProcessAudioChunk(
   if (chunk.mBuffer) {
     switch (chunk.mBufferFormat) {
       case AUDIO_FORMAT_FLOAT32:
-        MOZ_MTLOG(ML_ERROR, "Can't process audio except in 16-bit PCM yet");
-        MOZ_ASSERT(PR_FALSE);
-        return;
+        {
+          const float* buf = static_cast<const float *>(chunk.mChannelData[0]);
+          ConvertAudioSamplesWithScale(buf, static_cast<int16_t*>(samples),
+                                       chunk.mDuration, chunk.mVolume);
+        }
         break;
       case AUDIO_FORMAT_S16:
         {