H.264 packetization mode 0 (try 2)

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 9cf3150dfa25231fa80ec79456fe6e83ad4381df..527599ee397460b6edb50a64241c541c97e52cee 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
 
 #include <deque>
+#include <memory>
 #include <queue>
 #include <string>
 
@@ -25,7 +26,8 @@ class RtpPacketizerH264 : public RtpPacketizer {
  public:
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded H264 frame.
-  RtpPacketizerH264(FrameType frame_type, size_t max_payload_len);
+  RtpPacketizerH264(size_t max_payload_len,
+                    H264PacketizationMode packetization_mode);
 
   virtual ~RtpPacketizerH264();
 
@@ -89,10 +91,12 @@ class RtpPacketizerH264 : public RtpPacketizer {
   void GeneratePackets();
   void PacketizeFuA(size_t fragment_index);
   size_t PacketizeStapA(size_t fragment_index);
+  void PacketizeSingleNalu(size_t fragment_index);
   void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);
   void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);
 
   const size_t max_payload_len_;
+  const H264PacketizationMode packetization_mode_;
   std::deque<Fragment> input_fragments_;
   std::queue<PacketUnit> packets_;