RtpPacketizer::NextPacket fills RtpPacket instead of payload.

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..bc3240106d40918902ea97bb94320ff07f02f86a 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>
 
@@ -34,15 +35,11 @@ class RtpPacketizerH264 : public RtpPacketizer {
                       const RTPFragmentationHeader* fragmentation) override;
 
   // Get the next payload with H264 payload header.
-  // buffer is a pointer to where the output will be written.
-  // bytes_to_send is an output variable that will contain number of bytes
-  // written to buffer. The parameter last_packet is true for the last packet of
-  // the frame, false otherwise (i.e., call the function again to get the
-  // next packet).
-  // Returns true on success or false if there was no payload to packetize.
-  bool NextPacket(uint8_t* buffer,
-                  size_t* bytes_to_send,
-                  bool* last_packet) override;
+  // Write payload and set marker bit of the |packet|.
+  // The parameter |last_packet| is true for the last packet of the frame, false
+  // otherwise (i.e., call the function again to get the next packet).
+  // Returns true on success, false otherwise.
+  bool NextPacket(RtpPacketToSend* rtp_packet, bool* last_packet) override;
 
   ProtectionType GetProtectionType() override;
 
@@ -89,8 +86,8 @@ class RtpPacketizerH264 : public RtpPacketizer {
   void GeneratePackets();
   void PacketizeFuA(size_t fragment_index);
   size_t PacketizeStapA(size_t fragment_index);
-  void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);
-  void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);
+  void NextAggregatePacket(RtpPacketToSend* rtp_packet);
+  void NextFragmentPacket(RtpPacketToSend* rtp_packet);
 
   const size_t max_payload_len_;
   std::deque<Fragment> input_fragments_;