diff --git a/dom/media/webrtc/MediaEngineWebRTC.h b/dom/media/webrtc/MediaEngineWebRTC.h
index 719e540c725f..56c0fd53eda3 100644
--- a/dom/media/webrtc/MediaEngineWebRTC.h
+++ b/dom/media/webrtc/MediaEngineWebRTC.h
@@ -434,7 +434,7 @@ public:
     , mSampleFrequency(MediaEngine::DEFAULT_SAMPLE_RATE)
     , mPlayoutDelay(0)
     , mNullTransport(nullptr)
-    , mInputBufferLen(0) {
+  {
     MOZ_ASSERT(aVoiceEnginePtr);
     MOZ_ASSERT(aAudioInput);
     mDeviceName.Assign(NS_ConvertUTF8toUTF16(name));
@@ -554,9 +554,7 @@ private:
 
   NullTransport *mNullTransport;
 
-  // For full_duplex packetizer output
-  size_t mInputBufferLen;
-  UniquePtr<int16_t[]> mInputBuffer;
+  nsTArray<int16_t> mInputBuffer;
 };
 
 class MediaEngineWebRTC : public MediaEngine
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
index 4ca80b4468a9..f7953607de36 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -479,10 +479,10 @@ MediaEngineWebRTCMicrophoneSource::NotifyInputData(MediaStreamGraph* aGraph,
   while (mPacketizer->PacketsAvailable()) {
     uint32_t samplesPerPacket = mPacketizer->PacketSize() *
                                 mPacketizer->Channels();
-    if (mInputBufferLen < samplesPerPacket) {
-      mInputBuffer = MakeUnique<int16_t[]>(samplesPerPacket);
+    if (mInputBuffer.Length() < samplesPerPacket) {
+      mInputBuffer.SetLength(samplesPerPacket);
     }
-    int16_t *packet = mInputBuffer.get();
+    int16_t* packet = mInputBuffer.Elements();
     mPacketizer->Output(packet);
 
     mVoERender->ExternalRecordingInsertData(packet, samplesPerPacket, aRate, 0);
@@ -516,10 +516,6 @@ do {                                                                \
   }                                                                 \
 }  while(0)
 
-
-
-
-
 void
 MediaEngineWebRTCMicrophoneSource::DeviceChanged() {
   // Reset some processing