Revert of H.264 packetization mode 0 (try 3)

webrtc/modules/rtp_rtcp/source/rtp_format_h264.h

Index: webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
index fe7b37832aa90f1208be3aa4c6dd48a5d6ed2c50..bc3240106d40918902ea97bb94320ff07f02f86a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -26,8 +26,7 @@
  public:
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded H264 frame.
-  RtpPacketizerH264(size_t max_payload_len,
-                    H264PacketizationMode packetization_mode);
+  RtpPacketizerH264(FrameType frame_type, size_t max_payload_len);
 
   virtual ~RtpPacketizerH264();
 
@@ -87,12 +86,10 @@
   void GeneratePackets();
   void PacketizeFuA(size_t fragment_index);
   size_t PacketizeStapA(size_t fragment_index);
-  void PacketizeSingleNalu(size_t fragment_index);
   void NextAggregatePacket(RtpPacketToSend* rtp_packet);
   void NextFragmentPacket(RtpPacketToSend* rtp_packet);
 
   const size_t max_payload_len_;
-  const H264PacketizationMode packetization_mode_;
   std::deque<Fragment> input_fragments_;
   std::queue<PacketUnit> packets_;