RtpPacketizer::NextPacket fills RtpPacket instead of payload.

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(FrameType frame_type, size_t max_payload_len);
 
   virtual ~RtpPacketizerH264();
 
   void SetPayloadData(const uint8_t* payload_data,
                       size_t payload_size,
                       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;
 
   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 26 matching lines @@
     const Fragment source_fragment;
     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 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_;
   std::queue<PacketUnit> packets_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
 };
 
 // Depacketizer for H264.
 class RtpDepacketizerH264 : public RtpDepacketizer {
@@ skipping 10 matching lines @@
                     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_