[feat] add rtp packet history module

src/rtp/rtp_packet/rtp_packet_history.cpp

@@ -0,0 +1,39 @@
+#include "rtp_packet_history.h"
+
+#include "sequence_number_compare.h"
+
+RtpPacketHistory::RtpPacketHistory() {}
+
+RtpPacketHistory::~RtpPacketHistory() {}
+
+void RtpPacketHistory::AddPacket(std::shared_ptr<RtpPacketToSend> rtp_packet,
+                                 Timestamp send_time) {
+  rtp_packet_history_.push_back(
+      {rtp_packet, send_time, GetPacketIndex(rtp_packet->SequenceNumber())});
+}
+
+int RtpPacketHistory::GetPacketIndex(uint16_t sequence_number) const {
+  if (packet_history_.empty()) {
+    return 0;
+  }
+
+  int first_seq = packet_history_.front().packet_->SequenceNumber();
+  if (first_seq == sequence_number) {
+    return 0;
+  }
+
+  int packet_index = sequence_number - first_seq;
+  constexpr int kSeqNumSpan = std::numeric_limits<uint16_t>::max() + 1;
+
+  if (IsNewerSequenceNumber(sequence_number, first_seq)) {
+    if (sequence_number < first_seq) {
+      // Forward wrap.
+      packet_index += kSeqNumSpan;
+    }
+  } else if (sequence_number > first_seq) {
+    // Backwards wrap.
+    packet_index -= kSeqNumSpan;
+  }
+
+  return packet_index;
+}

src/rtp/rtp_packet/rtp_packet_history.h

@@ -0,0 +1,38 @@
+/*
+ * @Author: DI JUNKUN
+ * @Date: 2025-02-14
+ * Copyright (c) 2025 by DI JUNKUN, All Rights Reserved.
+ */
+
+#ifndef _RTP_PACKET_HISTORY_H_
+#define _RTP_PACKET_HISTORY_H_
+
+#include <deque>
+
+#include "rtp_packet_to_send.h"
+
+class RtpPacketHistory {
+ public:
+  RtpPacketHistory();
+  ~RtpPacketHistory();
+
+  void AddPacket(std::shared_ptr<RtpPacketToSend> rtp_packet,
+                 Timestamp send_time);
+
+ private:
+  int GetPacketIndex(uint16_t sequence_number) const;
+
+  return packet_index;
+}
+
+private : struct RtpPacketToSendInfo {
+  std::shared_ptr<RtpPacketToSend> rtp_packet;
+  Timestamp send_time;
+  uint64_t index;
+};
+
+private:
+std::deque<std::shared_ptr<RtpPacketToSend>> rtp_packet_history_;
+}
+
+#endif

src/rtp/rtp_packet/rtp_packet_to_send.cpp

@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2018 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 "rtp_packet_to_send.h"
+
+#include <cstdint>
+
+namespace webrtc {
+
+RtpPacketToSend::RtpPacketToSend() {}
+RtpPacketToSend::RtpPacketToSend(size_t capacity) : RtpPacket(capacity) {}
+RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
+RtpPacketToSend::RtpPacketToSend(RtpPacketToSend&& packet) = default;
+
+RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
+    default;
+RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
+
+RtpPacketToSend::~RtpPacketToSend() = default;
+
+void RtpPacketToSend::set_packet_type(RtpPacketMediaType type) {
+  if (packet_type_ == RtpPacketMediaType::kAudio) {
+    original_packet_type_ = OriginalType::kAudio;
+  } else if (packet_type_ == RtpPacketMediaType::kVideo) {
+    original_packet_type_ = OriginalType::kVideo;
+  }
+  packet_type_ = type;
+}
+
+}  // namespace webrtc

src/rtp/rtp_packet/rtp_packet_to_send.h

@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2016 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.
+ */
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <optional>
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/rtp_rtcp/rtp_rtcp_typedef.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_timing.h"
+#include "rtp_packet.h"
+
+// Forward declare the RtpPacket class since it is not in the webrtc namespace.
+class RtpPacket;
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for sender side.
+// The metadata is not send over the wire, but packet sender may use it to
+// create rtp header extensions or other data that is sent over the wire.
+class RtpPacketToSend : public ::RtpPacket {
+ public:
+  explicit RtpPacketToSend();
+  RtpPacketToSend(size_t capacity);
+  RtpPacketToSend(const RtpPacketToSend& packet);
+  RtpPacketToSend(RtpPacketToSend&& packet);
+
+  RtpPacketToSend& operator=(const RtpPacketToSend& packet);
+  RtpPacketToSend& operator=(RtpPacketToSend&& packet);
+
+  ~RtpPacketToSend();
+
+  // Time in local time base as close as it can to frame capture time.
+  webrtc::Timestamp capture_time() const { return capture_time_; }
+  void set_capture_time(webrtc::Timestamp time) { capture_time_ = time; }
+
+  void set_packet_type(webrtc::RtpPacketMediaType type);
+
+  std::optional<webrtc::RtpPacketMediaType> packet_type() const {
+    return packet_type_;
+  }
+
+  enum class OriginalType { kAudio, kVideo };
+  // Original type does not change if packet type is changed to kRetransmission.
+  std::optional<OriginalType> original_packet_type() const {
+    return original_packet_type_;
+  }
+
+  // If this is a retransmission, indicates the sequence number of the original
+  // media packet that this packet represents. If RTX is used this will likely
+  // be different from SequenceNumber().
+  void set_retransmitted_sequence_number(uint16_t sequence_number) {
+    retransmitted_sequence_number_ = sequence_number;
+  }
+  std::optional<uint16_t> retransmitted_sequence_number() const {
+    return retransmitted_sequence_number_;
+  }
+
+  // If this is a retransmission, indicates the SSRC of the original
+  // media packet that this packet represents.
+  void set_original_ssrc(uint32_t ssrc) { original_ssrc_ = ssrc; }
+  std::optional<uint32_t> original_ssrc() const { return original_ssrc_; }
+
+  void set_allow_retransmission(bool allow_retransmission) {
+    allow_retransmission_ = allow_retransmission;
+  }
+  bool allow_retransmission() const { return allow_retransmission_; }
+
+  // An application can attach arbitrary data to an RTP packet using
+  // `additional_data`. The additional data does not affect WebRTC processing.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
+    return additional_data_;
+  }
+  void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
+    additional_data_ = std::move(data);
+  }
+
+  void set_packetization_finish_time(webrtc::Timestamp time) {
+    // SetExtension<VideoTimingExtension>(
+    //     VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+    //     VideoTimingExtension::kPacketizationFinishDeltaOffset);
+  }
+
+  void set_pacer_exit_time(webrtc::Timestamp time) {
+    // SetExtension<VideoTimingExtension>(
+    //     VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+    //     VideoTimingExtension::kPacerExitDeltaOffset);
+  }
+
+  void set_network_time(webrtc::Timestamp time) {
+    // SetExtension<VideoTimingExtension>(
+    //     VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+    //     VideoTimingExtension::kNetworkTimestampDeltaOffset);
+  }
+
+  void set_network2_time(webrtc::Timestamp time) {
+    // SetExtension<VideoTimingExtension>(
+    //     VideoSendTiming::GetDeltaCappedMs(time - capture_time_),
+    //     VideoTimingExtension::kNetwork2TimestampDeltaOffset);
+  }
+
+  // Indicates if packet is the first packet of a video frame.
+  void set_first_packet_of_frame(bool is_first_packet) {
+    is_first_packet_of_frame_ = is_first_packet;
+  }
+  bool is_first_packet_of_frame() const { return is_first_packet_of_frame_; }
+
+  // Indicates if packet contains payload for a video key-frame.
+  void set_is_key_frame(bool is_key_frame) { is_key_frame_ = is_key_frame; }
+  bool is_key_frame() const { return is_key_frame_; }
+
+  // Indicates if packets should be protected by FEC (Forward Error Correction).
+  void set_fec_protect_packet(bool protect) { fec_protect_packet_ = protect; }
+  bool fec_protect_packet() const { return fec_protect_packet_; }
+
+  // Indicates if packet is using RED encapsulation, in accordance with
+  // https://tools.ietf.org/html/rfc2198
+  void set_is_red(bool is_red) { is_red_ = is_red; }
+  bool is_red() const { return is_red_; }
+
+  // The amount of time spent in the send queue, used for totalPacketSendDelay.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+  void set_time_in_send_queue(TimeDelta time_in_send_queue) {
+    time_in_send_queue_ = time_in_send_queue;
+  }
+  std::optional<TimeDelta> time_in_send_queue() const {
+    return time_in_send_queue_;
+  }
+  // A sequence number guaranteed to be monotically increasing by one for all
+  // packets where transport feedback is expected.
+  std::optional<int64_t> transport_sequence_number() const {
+    return transport_sequence_number_;
+  }
+  void set_transport_sequence_number(int64_t transport_sequence_number) {
+    transport_sequence_number_ = transport_sequence_number;
+  }
+  // Transport is capable of handling explicit congestion notification and the
+  // RTP packet should be sent as ect(1)
+  // https://www.rfc-editor.org/rfc/rfc9331.html
+  bool send_as_ect1() const { return send_as_ect1_; }
+  void set_send_as_ect1() { send_as_ect1_ = true; }
+
+ private:
+  webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
+  std::optional<webrtc::RtpPacketMediaType> packet_type_;
+  std::optional<OriginalType> original_packet_type_;
+  std::optional<uint32_t> original_ssrc_;
+  std::optional<int64_t> transport_sequence_number_;
+  bool allow_retransmission_ = false;
+  std::optional<uint16_t> retransmitted_sequence_number_;
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
+  bool is_first_packet_of_frame_ = false;
+  bool is_key_frame_ = false;
+  bool fec_protect_packet_ = false;
+  bool is_red_ = false;
+  bool send_as_ect1_ = false;
+  std::optional<TimeDelta> time_in_send_queue_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_

src/rtp/rtp_packetizer/rtp_packetizer.h

@@ -12,6 +12,7 @@
 #include <memory>
 
 #include "rtp_packet.h"
+#include "rtp_packet_to_send.h"
 
 class RtpPacketizer {
  public:
@@ -20,8 +21,8 @@ class RtpPacketizer {
 
   virtual ~RtpPacketizer() = default;
 
-  virtual std::vector<RtpPacket> Build(uint8_t* payload,
-                                       uint32_t payload_size) = 0;
+  virtual std::vector<std::shared_ptr<RtpPacket>> Build(
+      uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) = 0;
 };
 
 #endif

src/rtp/rtp_packetizer/rtp_packetizer_av1.cpp

@@ -4,8 +4,9 @@ RtpPacketizerAv1::RtpPacketizerAv1(uint32_t ssrc) {}
 
 RtpPacketizerAv1::~RtpPacketizerAv1() {}
 
-std::vector<RtpPacket> RtpPacketizerAv1::Build(uint8_t* payload,
-                                               uint32_t payload_size) {
-  std::vector<RtpPacket> rtp_packets;
+std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerAv1::Build(
+    uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
+  std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
+
   return rtp_packets;
-}
+}

src/rtp/rtp_packetizer/rtp_packetizer_av1.h

@@ -15,8 +15,9 @@ class RtpPacketizerAv1 : public RtpPacketizer {
 
   virtual ~RtpPacketizerAv1();
 
-  std::vector<RtpPacket> Build(uint8_t* payload,
-                               uint32_t payload_size) override;
+  std::vector<std::shared_ptr<RtpPacket>> Build(
+      uint8_t* payload, uint32_t payload_size,
+      bool use_rtp_packet_to_send) override;
 
  private:
   uint8_t version_;

src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp

@@ -17,8 +17,37 @@ RtpPacketizerGeneric::RtpPacketizerGeneric(uint32_t ssrc)
 
 RtpPacketizerGeneric::~RtpPacketizerGeneric() {}
 
-std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
-                                                   uint32_t payload_size) {
+void RtpPacketizerGeneric::AddAbsSendTimeExtension(
+    std::vector<uint8_t>& rtp_packet_frame) {
+  uint16_t extension_profile = 0xBEDE;  // One-byte header extension
+  uint8_t sub_extension_id = 3;         // ID for Absolute Send Time
+  uint8_t sub_extension_length =
+      2;  // Length of the extension data in bytes minus 1
+
+  uint32_t abs_send_time =
+      std::chrono::duration_cast<std::chrono::microseconds>(
+          std::chrono::system_clock::now().time_since_epoch())
+          .count();
+  abs_send_time &= 0x00FFFFFF;  // Absolute Send Time is 24 bits
+
+  // Add extension profile
+  rtp_packet_frame.push_back((extension_profile >> 8) & 0xFF);
+  rtp_packet_frame.push_back(extension_profile & 0xFF);
+
+  // Add extension length (in 32-bit words, minus one)
+  rtp_packet_frame.push_back(
+      0x00);  // Placeholder for length, will be updated later
+  rtp_packet_frame.push_back(0x01);  // One 32-bit word
+
+  // Add Absolute Send Time extension
+  rtp_packet_frame.push_back((sub_extension_id << 4) | sub_extension_length);
+  rtp_packet_frame.push_back((abs_send_time >> 16) & 0xFF);
+  rtp_packet_frame.push_back((abs_send_time >> 8) & 0xFF);
+  rtp_packet_frame.push_back(abs_send_time & 0xFF);
+}
+
+std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerGeneric::Build(
+    uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
   uint32_t last_packet_size = payload_size % MAX_NALU_LEN;
   uint32_t packet_num =
       payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
@@ -28,7 +57,8 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
                            std::chrono::system_clock::now().time_since_epoch())
                            .count();
 
-  std::vector<RtpPacket> rtp_packets;
+  std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
+
   for (uint32_t index = 0; index < packet_num; index++) {
     version_ = kRtpVersion;
     has_padding_ = false;
@@ -77,40 +107,17 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
                                payload + MAX_NALU_LEN);
     }
 
-    RtpPacket rtp_packet;
-    rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
-
-    rtp_packets.emplace_back(rtp_packet);
+    if (use_rtp_packet_to_send) {
+      std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet =
+          std::make_unique<webrtc::RtpPacketToSend>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    } else {
+      std::shared_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    }
   }
 
   return rtp_packets;
 }
-
-void RtpPacketizerGeneric::AddAbsSendTimeExtension(
-    std::vector<uint8_t>& rtp_packet_frame) {
-  uint16_t extension_profile = 0xBEDE;  // One-byte header extension
-  uint8_t sub_extension_id = 3;         // ID for Absolute Send Time
-  uint8_t sub_extension_length =
-      2;  // Length of the extension data in bytes minus 1
-
-  uint32_t abs_send_time =
-      std::chrono::duration_cast<std::chrono::microseconds>(
-          std::chrono::system_clock::now().time_since_epoch())
-          .count();
-  abs_send_time &= 0x00FFFFFF;  // Absolute Send Time is 24 bits
-
-  // Add extension profile
-  rtp_packet_frame.push_back((extension_profile >> 8) & 0xFF);
-  rtp_packet_frame.push_back(extension_profile & 0xFF);
-
-  // Add extension length (in 32-bit words, minus one)
-  rtp_packet_frame.push_back(
-      0x00);  // Placeholder for length, will be updated later
-  rtp_packet_frame.push_back(0x01);  // One 32-bit word
-
-  // Add Absolute Send Time extension
-  rtp_packet_frame.push_back((sub_extension_id << 4) | sub_extension_length);
-  rtp_packet_frame.push_back((abs_send_time >> 16) & 0xFF);
-  rtp_packet_frame.push_back((abs_send_time >> 8) & 0xFF);
-  rtp_packet_frame.push_back(abs_send_time & 0xFF);
-}

src/rtp/rtp_packetizer/rtp_packetizer_generic.h

@@ -15,8 +15,9 @@ class RtpPacketizerGeneric : public RtpPacketizer {
 
   virtual ~RtpPacketizerGeneric();
 
-  std::vector<RtpPacket> Build(uint8_t* payload,
-                               uint32_t payload_size) override;
+  std::vector<std::shared_ptr<RtpPacket>> Build(
+      uint8_t* payload, uint32_t payload_size,
+      bool use_rtp_packet_to_send) override;
 
  private:
   void AddAbsSendTimeExtension(std::vector<uint8_t>& rtp_packet_frame);

src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp

@@ -17,15 +17,6 @@ RtpPacketizerH264::RtpPacketizerH264(uint32_t ssrc)
 
 RtpPacketizerH264::~RtpPacketizerH264() {}
 
-std::vector<RtpPacket> RtpPacketizerH264::Build(uint8_t* payload,
-                                                uint32_t payload_size) {
-  if (payload_size <= MAX_NALU_LEN) {
-    return BuildNalu(payload, payload_size);
-  } else {
-    return BuildFua(payload, payload_size);
-  }
-}
-
 //  0                   1                   2                   3
 //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
@@ -67,9 +58,18 @@ void RtpPacketizerH264::AddAbsSendTimeExtension(
   rtp_packet_frame.push_back(abs_send_time & 0xFF);
 }
 
-std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
-                                                    uint32_t payload_size) {
-  std::vector<RtpPacket> rtp_packets;
+std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::Build(
+    uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
+  if (payload_size <= MAX_NALU_LEN) {
+    return BuildNalu(payload, payload_size, use_rtp_packet_to_send);
+  } else {
+    return BuildFua(payload, payload_size, use_rtp_packet_to_send);
+  }
+}
+
+std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
+    uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
+  std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
 
   version_ = kRtpVersion;
   has_padding_ = false;
@@ -123,16 +123,23 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
   rtp_packet_frame_.insert(rtp_packet_frame_.end(), payload,
                            payload + payload_size);
 
-  RtpPacket rtp_packet;
-  rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
-  rtp_packets.emplace_back(rtp_packet);
+  if (use_rtp_packet_to_send) {
+    std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet =
+        std::make_unique<webrtc::RtpPacketToSend>();
+    rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+    rtp_packets.emplace_back(std::move(rtp_packet));
+  } else {
+    std::shared_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
+    rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+    rtp_packets.emplace_back(std::move(rtp_packet));
+  }
 
   return rtp_packets;
 }
 
-std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
-                                                   uint32_t payload_size) {
-  std::vector<RtpPacket> rtp_packets;
+std::vector<std::shared_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
+    uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send) {
+  std::vector<std::shared_ptr<RtpPacket>> rtp_packets;
 
   uint32_t last_packet_size = payload_size % MAX_NALU_LEN;
   uint32_t packet_num =
@@ -214,10 +221,16 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
                                payload + index * MAX_NALU_LEN + MAX_NALU_LEN);
     }
 
-    RtpPacket rtp_packet;
-    rtp_packet.Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
-
-    rtp_packets.emplace_back(rtp_packet);
+    if (use_rtp_packet_to_send) {
+      std::shared_ptr<webrtc::RtpPacketToSend> rtp_packet =
+          std::make_unique<webrtc::RtpPacketToSend>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    } else {
+      std::shared_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packets.emplace_back(std::move(rtp_packet));
+    }
   }
 
   return rtp_packets;

src/rtp/rtp_packetizer/rtp_packetizer_h264.h

@@ -15,12 +15,16 @@ class RtpPacketizerH264 : public RtpPacketizer {
 
   virtual ~RtpPacketizerH264();
 
-  std::vector<RtpPacket> Build(uint8_t* payload,
-                               uint32_t payload_size) override;
+  std::vector<std::shared_ptr<RtpPacket>> Build(
+      uint8_t* payload, uint32_t payload_size,
+      bool use_rtp_packet_to_send) override;
 
-  std::vector<RtpPacket> BuildNalu(uint8_t* payload, uint32_t payload_size);
+  std::vector<std::shared_ptr<RtpPacket>> BuildNalu(
+      uint8_t* payload, uint32_t payload_size, bool use_rtp_packet_to_send);
 
-  std::vector<RtpPacket> BuildFua(uint8_t* payload, uint32_t payload_size);
+  std::vector<std::shared_ptr<RtpPacket>> BuildFua(uint8_t* payload,
+                                                   uint32_t payload_size,
+                                                   bool use_rtp_packet_to_send);
 
  private:
   bool EncodeH264Fua(RtpPacket& rtp_packet, uint8_t* payload,