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

webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h

 /*
  *  Copyright (c) 2015 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.
  */
 
@@ skipping 15 matching lines @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class RtpPacketizerVp9 : public RtpPacketizer {
  public:
-  RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, size_t max_payload_length);
+  RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr,
+                   size_t max_payload_length,
+                   size_t last_packet_reduction_len);
 
   virtual ~RtpPacketizerVp9();
 
   ProtectionType GetProtectionType() override;
 
   StorageType GetStorageType(uint32_t retransmission_settings) override;
 
   std::string ToString() override;
 
-  // The payload data must be one encoded VP9 frame.
-  void SetPayloadData(const uint8_t* payload,
-                      size_t payload_size,
-                      const RTPFragmentationHeader* fragmentation) override;
+  // The payload data must be one encoded VP9 layer frame.
+  size_t SetPayloadData(const uint8_t* payload,
+                        size_t payload_size,
+                        const RTPFragmentationHeader* fragmentation) override;
 
   // Gets the next payload with VP9 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.
-  bool NextPacket(RtpPacketToSend* packet, bool* last_packet) override;
+  bool NextPacket(RtpPacketToSend* packet) override;
 
   typedef struct {
     size_t payload_start_pos;
     size_t size;
     bool layer_begin;
     bool layer_end;
   } PacketInfo;
   typedef std::queue<PacketInfo> PacketInfoQueue;
 
  private:
   // Calculates all packet sizes and loads info to packet queue.
   void GeneratePackets();
 
   // Writes the payload descriptor header and copies payload to the |buffer|.
   // |packet_info| determines which part of the payload to write.
-  // |bytes_to_send| contains the number of written bytes to the buffer.
+  // |last| indicates if the packet is last.

[danilchap, 2017/05/23 15:55:03]

probably drop the: 'if packet is last'
may be add '...is last in the frame.'
or 'is last to generate.'

[ilnik, 2017/05/23 16:14:30]

Done.

   // Returns true on success, false otherwise.
   bool WriteHeaderAndPayload(const PacketInfo& packet_info,
-                             RtpPacketToSend* packet) const;
+                             RtpPacketToSend* packet,
+                             bool last) const;
 
   // Writes payload descriptor header to |buffer|.
   // Returns true on success, false otherwise.
   bool WriteHeader(const PacketInfo& packet_info,
                    uint8_t* buffer,
                    size_t* header_length) const;
 
   const RTPVideoHeaderVP9 hdr_;
   const size_t max_payload_length_;  // The max length in bytes of one packet.
   const uint8_t* payload_;           // The payload data to be packetized.
   size_t payload_size_;              // The size in bytes of the payload data.
+  const size_t last_packet_reduction_len_;
   PacketInfoQueue packets_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
 };
 
 
 class RtpDepacketizerVp9 : public RtpDepacketizer {
  public:
   virtual ~RtpDepacketizerVp9() {}
 
   bool Parse(ParsedPayload* parsed_payload,
              const uint8_t* payload,
              size_t payload_length) override;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_