Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/producer_fec.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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 
 #include <memory>
 #include <utility>
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 
 constexpr size_t kRedForFecHeaderLength = 1;
 
 // This controls the maximum amount of excess overhead (actual - target)
 // allowed in order to trigger GenerateFec(), before |params_.max_fec_frames|
 // is reached. Overhead here is defined as relative to number of media packets.
 constexpr int kMaxExcessOverhead = 50;  // Q8.
@@ skipping 107 matching lines @@
   // produced.
   new_params_ = *params;
   num_important_packets_ = num_important_packets;
   if (params->fec_rate > kHighProtectionThreshold) {
     min_num_media_packets_ = kMinMediaPackets;
   } else {
     min_num_media_packets_ = 1;
   }
 }
 
+void ProducerFec::BuildRedPacket(int red_pl_type,
+                                 const RtpPacketToSend& media_packet,
+                                 RtpPacketToSend* red_packet) {
+  RTC_DCHECK_GE(red_packet->capacity(),
+                media_packet.size() + kRedForFecHeaderLength);
+  red_packet->CopyHeaderFrom(media_packet);
+  red_packet->SetPayloadType(red_pl_type);
+  uint8_t* red_payload = red_packet->AllocatePayload(
+      kRedForFecHeaderLength + media_packet.payload_size());
+  red_payload[0] = media_packet.PayloadType();
+  memcpy(&red_payload[kRedForFecHeaderLength], media_packet.payload(),
+         media_packet.payload_size());
+}
+
 int ProducerFec::AddRtpPacketAndGenerateFec(const uint8_t* data_buffer,
                                             size_t payload_length,
                                             size_t rtp_header_length) {
   RTC_DCHECK(generated_fec_packets_.empty());
   if (media_packets_.empty()) {
     params_ = new_params_;
   }
   bool complete_frame = false;
   const bool marker_bit = (data_buffer[1] & kRtpMarkerBitMask) ? true : false;
   if (media_packets_.size() < ForwardErrorCorrection::kMaxMediaPackets) {
@@ skipping 104 matching lines @@
       fec_->GetNumberOfFecPackets(media_packets_.size(), params_.fec_rate);
   // Return the overhead in Q8.
   return (num_fec_packets << 8) / media_packets_.size();
 }
 
 void ProducerFec::DeleteMediaPackets() {
   media_packets_.clear();
 }
 
 }  // namespace webrtc