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

webrtc/modules/rtp_rtcp/source/rtp_format.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 class RtpPacketToSend;
 
 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;
 
   virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
 
   virtual std::string ToString() = 0;
 };
 
 // TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
 // of the parsed payload, rather than just a pointer into the incoming buffer.
@@ skipping 12 matching lines @@
 
   virtual ~RtpDepacketizer() {}
 
   // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
   virtual bool Parse(ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_