RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 
 #include <assert.h>  // assert
 #include <string.h>  // memcpy
 
 #include <vector>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
 namespace webrtc {
 namespace {
 int ParseVP8PictureID(RTPVideoHeaderVP8* vp8,
                       const uint8_t** data,
                       size_t* data_length,
                       size_t* parsed_bytes) {
   assert(vp8 != NULL);
   if (*data_length == 0)
     return -1;
@@ skipping 160 matching lines @@
     part_info_.CopyFrom(*fragmentation);
     num_partitions_ = fragmentation->fragmentationVectorSize;
   } else {
     part_info_.VerifyAndAllocateFragmentationHeader(1);
     part_info_.fragmentationLength[0] = payload_size;
     part_info_.fragmentationOffset[0] = 0;
     num_partitions_ = part_info_.fragmentationVectorSize;
   }
 }
 
-bool RtpPacketizerVp8::NextPacket(uint8_t* buffer,
-                                  size_t* bytes_to_send,
-                                  bool* last_packet) {
+bool RtpPacketizerVp8::NextPacket(RtpPacketToSend* packet, bool* last_packet) {
+  RTC_DCHECK(packet);
+  RTC_DCHECK(last_packet);
   if (!packets_calculated_) {
     int ret = 0;
     if (aggr_mode_ == kAggrPartitions && balance_) {
       ret = GeneratePacketsBalancedAggregates();
     } else {
       ret = GeneratePackets();
     }
     if (ret < 0) {
       return false;
     }
   }
   if (packets_.empty()) {
     return false;
   }
   InfoStruct packet_info = packets_.front();
   packets_.pop();
 
+  uint8_t* buffer = packet->AllocatePayload(max_payload_len_);
   int bytes = WriteHeaderAndPayload(packet_info, buffer, max_payload_len_);
   if (bytes < 0) {
     return false;
   }
-  *bytes_to_send = static_cast<size_t>(bytes);
-
+  packet->SetPayloadSize(bytes);
   *last_packet = packets_.empty();
+  packet->SetMarker(*last_packet);
   return true;
 }
 
 ProtectionType RtpPacketizerVp8::GetProtectionType() {
   bool protect =
       hdr_info_.temporalIdx == 0 || hdr_info_.temporalIdx == kNoTemporalIdx;
   return protect ? kProtectedPacket : kUnprotectedPacket;
 }
 
 StorageType RtpPacketizerVp8::GetStorageType(uint32_t retransmission_settings) {
@@ skipping 497 matching lines @@
   if (ParseVP8FrameSize(parsed_payload, payload_data, payload_data_length) !=
       0) {
     return false;
   }
 
   parsed_payload->payload = payload_data;
   parsed_payload->payload_length = payload_data_length;
   return true;
 }
 }  // namespace webrtc