RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h

 /*
  *  Copyright (c) 2011 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.
  */
 
 // This file contains the class RtpFormatVp8TestHelper. The class is
 // responsible for setting up a fake VP8 bitstream according to the
 // RTPVideoHeaderVP8 header, and partition information. After initialization,
 // an RTPFragmentationHeader is provided so that the tester can create a
 // packetizer. The packetizer can then be provided to this helper class, which
 // will then extract all packets and compare to the expected outcome.
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 namespace test {
 
 class RtpFormatVp8TestHelper {
  public:
   explicit RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr);
   ~RtpFormatVp8TestHelper();
   bool Init(const size_t* partition_sizes, size_t num_partitions);
   void GetAllPacketsAndCheck(RtpPacketizerVp8* packetizer,
                              const size_t* expected_sizes,
                              const int* expected_part,
                              const bool* expected_frag_start,
                              size_t expected_num_packets);
 
   uint8_t* payload_data() const { return payload_data_; }
   size_t payload_size() const { return payload_size_; }
   RTPFragmentationHeader* fragmentation() const { return fragmentation_; }
-  size_t buffer_size() const { return buffer_size_; }
+  size_t buffer_size() const {
+    static constexpr size_t kVp8PayloadDescriptorMaxSize = 6;
+    return payload_size_ + kVp8PayloadDescriptorMaxSize;
+  }
   void set_sloppy_partitioning(bool value) { sloppy_partitioning_ = value; }
 
  private:
   void CheckHeader(bool frag_start);
   void CheckPictureID();
   void CheckTl0PicIdx();
   void CheckTIDAndKeyIdx();
-  void CheckPayload(size_t payload_end);
+  void CheckPayload();
   void CheckLast(bool last) const;
-  void CheckPacket(size_t send_bytes, size_t expect_bytes, bool last,
-                   bool frag_start);
+  void CheckPacket(size_t expect_bytes, bool last, bool frag_start);
 
+  RtpPacketToSend packet_;
   uint8_t* payload_data_;
-  uint8_t* buffer_;
   uint8_t* data_ptr_;
   RTPFragmentationHeader* fragmentation_;
   const RTPVideoHeaderVP8* hdr_info_;
   int payload_start_;
   size_t payload_size_;
-  size_t buffer_size_;
   bool sloppy_partitioning_;
   bool inited_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper);
 };
 
 }  // namespace test
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_