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

webrtc/modules/rtp_rtcp/source/rtp_format_h264.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_H264_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
 
 #include <deque>
 #include <memory>
 #include <queue>
 #include <string>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
 namespace webrtc {
 
 class RtpPacketizerH264 : public RtpPacketizer {
  public:
   // Initialize with payload from encoder.
   // The payload_data must be exactly one encoded H264 frame.
   RtpPacketizerH264(size_t max_payload_len,
+                    size_t last_packet_reduction_len,
                     H264PacketizationMode packetization_mode);
 
   virtual ~RtpPacketizerH264();
 
-  void SetPayloadData(const uint8_t* payload_data,
-                      size_t payload_size,
-                      const RTPFragmentationHeader* fragmentation) override;
+  size_t SetPayloadData(const uint8_t* payload_data,
+                        size_t payload_size,
+                        const RTPFragmentationHeader* fragmentation) override;
 
   // Get the next payload with H264 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* rtp_packet, bool* last_packet) override;
+  bool NextPacket(RtpPacketToSend* rtp_packet) override;
 
   ProtectionType GetProtectionType() override;
 
   StorageType GetStorageType(uint32_t retransmission_settings) override;
 
   std::string ToString() override;
 
  private:
   // Input fragments (NAL units), with an optionally owned temporary buffer,
   // used in case the fragment gets modified.
@@ skipping 27 matching lines @@
     bool first_fragment;
     bool last_fragment;
     bool aggregated;
     uint8_t header;
   };
 
   void GeneratePackets();
   void PacketizeFuA(size_t fragment_index);
   size_t PacketizeStapA(size_t fragment_index);
   void PacketizeSingleNalu(size_t fragment_index);
-  void NextAggregatePacket(RtpPacketToSend* rtp_packet);
+  void NextAggregatePacket(RtpPacketToSend* rtp_packet, bool last);
   void NextFragmentPacket(RtpPacketToSend* rtp_packet);
 
   const size_t max_payload_len_;
+  const size_t last_packet_reduction_len_;
+  size_t total_packets_;

[danilchap, 2017/05/23 12:17:27]

may be num_packets_left_
or is it needed? (can it be different than packets_.size()?)

[ilnik, 2017/05/23 12:37:59]

Yes, num_packets_left_ is better.
Still, It's different from packets_.size(), because in this implementation some
items in packets_ are parts of an 'aggregate' packet. 

   const H264PacketizationMode packetization_mode_;
   std::deque<Fragment> input_fragments_;
   std::queue<PacketUnit> packets_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
 };
 
 // Depacketizer for H264.
 class RtpDepacketizerH264 : public RtpDepacketizer {
  public:
   RtpDepacketizerH264();
   virtual ~RtpDepacketizerH264();
 
   bool Parse(ParsedPayload* parsed_payload,
              const uint8_t* payload_data,
              size_t payload_data_length) override;
 
  private:
   bool ParseFuaNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                     const uint8_t* payload_data);
   bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data);
 
   size_t offset_;
   size_t length_;
   std::unique_ptr<rtc::Buffer> modified_buffer_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_