RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc

 /*
  *  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.
  */
 
 #include <string>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 
 namespace webrtc {
 
 static const size_t kGenericHeaderLength = 1;
 
 RtpPacketizerGeneric::RtpPacketizerGeneric(FrameType frame_type,
                                            size_t max_payload_len)
     : payload_data_(NULL),
       payload_size_(0),
       max_payload_len_(max_payload_len - kGenericHeaderLength),
@@ skipping 12 matching lines @@
 
   // Fragment packets more evenly by splitting the payload up evenly.
   size_t num_packets =
       (payload_size_ + max_payload_len_ - 1) / max_payload_len_;
   payload_length_ = (payload_size_ + num_packets - 1) / num_packets;
   assert(payload_length_ <= max_payload_len_);
 
   generic_header_ = RtpFormatVideoGeneric::kFirstPacketBit;
 }
 
-bool RtpPacketizerGeneric::NextPacket(uint8_t* buffer,
-                                      size_t* bytes_to_send,
+bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet,
                                       bool* last_packet) {
+  RTC_DCHECK(packet);
+  RTC_DCHECK(last_packet);
   if (payload_size_ < payload_length_) {
     payload_length_ = payload_size_;
   }
 
   payload_size_ -= payload_length_;
-  *bytes_to_send = payload_length_ + kGenericHeaderLength;
-  assert(payload_length_ <= max_payload_len_);
+  RTC_DCHECK_LE(payload_length_, max_payload_len_);
 
-  uint8_t* out_ptr = buffer;
+  uint8_t* out_ptr =
+      packet->AllocatePayload(kGenericHeaderLength + payload_length_);
   // Put generic header in packet
   if (frame_type_ == kVideoFrameKey) {
     generic_header_ |= RtpFormatVideoGeneric::kKeyFrameBit;
   }
-  *out_ptr++ = generic_header_;
+  out_ptr[0] = generic_header_;
   // Remove first-packet bit, following packets are intermediate
   generic_header_ &= ~RtpFormatVideoGeneric::kFirstPacketBit;
 
   // Put payload in packet
-  memcpy(out_ptr, payload_data_, payload_length_);
+  memcpy(out_ptr + kGenericHeaderLength, payload_data_, payload_length_);
   payload_data_ += payload_length_;
 
   *last_packet = payload_size_ <= 0;
-
+  packet->SetMarker(*last_packet);
   return true;
 }
 
 ProtectionType RtpPacketizerGeneric::GetProtectionType() {
   return kProtectedPacket;
 }
 
 StorageType RtpPacketizerGeneric::GetStorageType(
     uint32_t retransmission_settings) {
   return kAllowRetransmission;
@@ skipping 23 matching lines @@
       (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
   parsed_payload->type.Video.codec = kRtpVideoGeneric;
   parsed_payload->type.Video.width = 0;
   parsed_payload->type.Video.height = 0;
 
   parsed_payload->payload = payload_data;
   parsed_payload->payload_length = payload_data_length;
   return true;
 }
 }  // namespace webrtc