RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 344 matching lines @@
   // issue is fixed.
   const RTPFragmentationHeader* frag =
       (video_type == kRtpVideoVp8) ? nullptr : fragmentation;
   packetizer->SetPayloadData(payload_data, payload_size, frag);
 
   bool first_frame = first_frame_sent_();
   bool first = true;
   bool last = false;
   while (!last) {
     std::unique_ptr<RtpPacketToSend> packet(new RtpPacketToSend(*rtp_header));
-    uint8_t* payload = packet->AllocatePayload(max_data_payload_length);
-    RTC_DCHECK(payload);
 
-    size_t payload_bytes_in_packet = 0;
-    if (!packetizer->NextPacket(payload, &payload_bytes_in_packet, &last))
+    if (!packetizer->NextPacket(packet.get(), &last))
       return false;
+    RTC_DCHECK_LE(packet->payload_size(), max_data_payload_length);
 
-    packet->SetPayloadSize(payload_bytes_in_packet);
-    packet->SetMarker(last);
     if (!rtp_sender_->AssignSequenceNumber(packet.get()))
       return false;
 
     const bool protect_packet =
         (packetizer->GetProtectionType() == kProtectedPacket);
     if (flexfec_enabled()) {
       // TODO(brandtr): Remove the FlexFEC code path when FlexfecSender
       // is wired up to PacedSender instead.
       SendVideoPacketWithFlexfec(std::move(packet), storage, protect_packet);
     } else if (red_enabled) {
@@ skipping 35 matching lines @@
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc