RtpPacketizer::NextPacket fills RtpPacket instead of payload.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.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 "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 
 #include <string.h>
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/common_video/h264/sps_vui_rewriter.h"
 #include "webrtc/common_video/h264/h264_common.h"
 #include "webrtc/common_video/h264/pps_parser.h"
 #include "webrtc/common_video/h264/sps_parser.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
 
 static const size_t kNalHeaderSize = 1;
@@ skipping 190 matching lines @@
     // we need to add the STAP-A NALU header and a length field for the first
     // NALU of this packet.
     if (aggregated_fragments == 0)
       fragment_headers_length += kNalHeaderSize + kLengthFieldSize;
     ++aggregated_fragments;
   }
   packets_.back().last_fragment = true;
   return fragment_index;
 }
 
-bool RtpPacketizerH264::NextPacket(uint8_t* buffer,
-                                   size_t* bytes_to_send,
+bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet,
                                    bool* last_packet) {
-  *bytes_to_send = 0;
+  RTC_DCHECK(rtp_packet);
+  RTC_DCHECK(last_packet);
   if (packets_.empty()) {
-    *bytes_to_send = 0;
     *last_packet = true;
     return false;
   }
 
   PacketUnit packet = packets_.front();
-
   if (packet.first_fragment && packet.last_fragment) {
     // Single NAL unit packet.
-    *bytes_to_send = packet.source_fragment.length;
-    memcpy(buffer, packet.source_fragment.buffer, *bytes_to_send);
+    size_t bytes_to_send = packet.source_fragment.length;
+    uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send);
+    memcpy(buffer, packet.source_fragment.buffer, bytes_to_send);
     packets_.pop();
     input_fragments_.pop_front();
-    RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
   } else if (packet.aggregated) {
-    NextAggregatePacket(buffer, bytes_to_send);
-    RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
+    NextAggregatePacket(rtp_packet);
   } else {
-    NextFragmentPacket(buffer, bytes_to_send);
-    RTC_CHECK_LE(*bytes_to_send, max_payload_len_);
+    NextFragmentPacket(rtp_packet);
   }
+  RTC_DCHECK_LE(rtp_packet->payload_size(), max_payload_len_);
   *last_packet = packets_.empty();
+  rtp_packet->SetMarker(*last_packet);
   return true;
 }
 
-void RtpPacketizerH264::NextAggregatePacket(uint8_t* buffer,
-                                            size_t* bytes_to_send) {
+void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) {
+  uint8_t* buffer = rtp_packet->AllocatePayload(max_payload_len_);
+  RTC_DCHECK(buffer);
   PacketUnit* packet = &packets_.front();
   RTC_CHECK(packet->first_fragment);
   // STAP-A NALU header.
   buffer[0] = (packet->header & (kFBit | kNriMask)) | H264::NaluType::kStapA;
-  int index = kNalHeaderSize;
-  *bytes_to_send += kNalHeaderSize;
+  size_t index = kNalHeaderSize;
   while (packet->aggregated) {
     const Fragment& fragment = packet->source_fragment;
     // Add NAL unit length field.
     ByteWriter<uint16_t>::WriteBigEndian(&buffer[index], fragment.length);
     index += kLengthFieldSize;
-    *bytes_to_send += kLengthFieldSize;
     // Add NAL unit.
     memcpy(&buffer[index], fragment.buffer, fragment.length);
     index += fragment.length;
-    *bytes_to_send += fragment.length;
     packets_.pop();
     input_fragments_.pop_front();
     if (packet->last_fragment)
       break;
     packet = &packets_.front();
   }
   RTC_CHECK(packet->last_fragment);
+  rtp_packet->SetPayloadSize(index);
 }
 
-void RtpPacketizerH264::NextFragmentPacket(uint8_t* buffer,
-                                           size_t* bytes_to_send) {
+void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) {
   PacketUnit* packet = &packets_.front();
   // NAL unit fragmented over multiple packets (FU-A).
   // We do not send original NALU header, so it will be replaced by the
   // FU indicator header of the first packet.
   uint8_t fu_indicator =
       (packet->header & (kFBit | kNriMask)) | H264::NaluType::kFuA;
   uint8_t fu_header = 0;
 
   // S | E | R | 5 bit type.
   fu_header |= (packet->first_fragment ? kSBit : 0);
   fu_header |= (packet->last_fragment ? kEBit : 0);
   uint8_t type = packet->header & kTypeMask;
   fu_header |= type;
+  const Fragment& fragment = packet->source_fragment;
+  uint8_t* buffer =
+      rtp_packet->AllocatePayload(kFuAHeaderSize + fragment.length);
   buffer[0] = fu_indicator;
   buffer[1] = fu_header;
-
-  const Fragment& fragment = packet->source_fragment;
-  *bytes_to_send = fragment.length + kFuAHeaderSize;
   memcpy(buffer + kFuAHeaderSize, fragment.buffer, fragment.length);
   if (packet->last_fragment)
     input_fragments_.pop_front();
   packets_.pop();
 }
 
 ProtectionType RtpPacketizerH264::GetProtectionType() {
   return kProtectedPacket;
 }
 
@@ skipping 276 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc