Fix packetization logic to leave space for extensions in the last packet

webrtc/modules/rtp_rtcp/source/rtp_format.h

Index: webrtc/modules/rtp_rtcp/source/rtp_format.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 3b6004b9a13f84e30a5e1ba1611a1c3c68e8696c..9fa3df5b9489234f7320e06dac71d7d6e7022db4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -24,21 +24,22 @@ class RtpPacketizer {
  public:
   static RtpPacketizer* Create(RtpVideoCodecTypes type,
                                size_t max_payload_len,
+                               size_t last_packet_reduction_len,
                                const RTPVideoTypeHeader* rtp_type_header,
                                FrameType frame_type);
 
   virtual ~RtpPacketizer() {}
 
-  virtual void SetPayloadData(const uint8_t* payload_data,
-                              size_t payload_size,
-                              const RTPFragmentationHeader* fragmentation) = 0;
+  // Returns total number of packets which would be produced by the packetizer.
+  virtual size_t SetPayloadData(
+      const uint8_t* payload_data,
+      size_t payload_size,
+      const RTPFragmentationHeader* fragmentation) = 0;
 
   // Get the next payload with payload header.
   // 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.
-  virtual bool NextPacket(RtpPacketToSend* packet, bool* last_packet) = 0;
+  virtual bool NextPacket(RtpPacketToSend* packet) = 0;
 
   virtual ProtectionType GetProtectionType() = 0;