RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h

 /*
  *  Copyright (c) 2013 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_VIDEO_GENERIC_H_
@@ skipping 18 matching lines @@
   // The payload_data must be exactly one encoded generic frame.
   RtpPacketizerGeneric(FrameType frametype, size_t max_payload_len);
 
   virtual ~RtpPacketizerGeneric();
 
   void SetPayloadData(const uint8_t* payload_data,
                       size_t payload_size,
                       const RTPFragmentationHeader* fragmentation) override;
 
   // Get the next payload with generic 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* packet, bool* last_packet) override;
 
   ProtectionType GetProtectionType() override;
 
   StorageType GetStorageType(uint32_t retransmission_settings) override;
 
   std::string ToString() override;
 
  private:
   const uint8_t* payload_data_;
   size_t payload_size_;
   const size_t max_payload_len_;
   FrameType frame_type_;
   size_t payload_length_;
   uint8_t generic_header_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
 };
 
 // Depacketizer for generic codec.
 class RtpDepacketizerGeneric : public RtpDepacketizer {
  public:
   virtual ~RtpDepacketizerGeneric() {}
 
   bool Parse(ParsedPayload* parsed_payload,
              const uint8_t* payload_data,
              size_t payload_data_length) override;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_