1 files changed, 147 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
new file mode 100644
index 0000000000..438ca354ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -0,0 +1,147 @@
+/*
+ *  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 <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_timing.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+
+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:
+  // RtpPacketToSend::Type is deprecated. Use RtpPacketMediaType directly.
+  using Type = RtpPacketMediaType;
+
+  explicit RtpPacketToSend(const ExtensionManager* extensions);
+  RtpPacketToSend(const ExtensionManager* extensions, 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(RtpPacketMediaType type) { packet_type_ = type; }
+  absl::optional<RtpPacketMediaType> packet_type() const {
+    return 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;
+  }
+  absl::optional<uint16_t> retransmitted_sequence_number() const {
+    return retransmitted_sequence_number_;
+  }
+
+  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;
+  }
+  absl::optional<TimeDelta> time_in_send_queue() const {
+    return time_in_send_queue_;
+  }
+
+ private:
+  webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
+  absl::optional<RtpPacketMediaType> packet_type_;
+  bool allow_retransmission_ = false;
+  absl::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;
+  absl::optional<TimeDelta> time_in_send_queue_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_