Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/api/rtp_packet_info.h

@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2019 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 API_RTP_PACKET_INFO_H_
+#define API_RTP_PACKET_INFO_H_
+
+#include <cstdint>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+//
+// Structure to hold information about a received |RtpPacket|. It is primarily
+// used to carry per-packet information from when a packet is received until
+// the information is passed to |SourceTracker|.
+//
+class RTC_EXPORT RtpPacketInfo {
+ public:
+  RtpPacketInfo();
+
+  RtpPacketInfo(uint32_t ssrc,
+                std::vector<uint32_t> csrcs,
+                uint32_t rtp_timestamp,
+                absl::optional<uint8_t> audio_level,
+                absl::optional<AbsoluteCaptureTime> absolute_capture_time,
+                int64_t receive_time_ms);
+
+  RtpPacketInfo(const RTPHeader& rtp_header, int64_t receive_time_ms);
+
+  RtpPacketInfo(const RtpPacketInfo& other) = default;
+  RtpPacketInfo(RtpPacketInfo&& other) = default;
+  RtpPacketInfo& operator=(const RtpPacketInfo& other) = default;
+  RtpPacketInfo& operator=(RtpPacketInfo&& other) = default;
+
+  uint32_t ssrc() const { return ssrc_; }
+  void set_ssrc(uint32_t value) { ssrc_ = value; }
+
+  const std::vector<uint32_t>& csrcs() const { return csrcs_; }
+  void set_csrcs(std::vector<uint32_t> value) { csrcs_ = std::move(value); }
+
+  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
+  void set_rtp_timestamp(uint32_t value) { rtp_timestamp_ = value; }
+
+  absl::optional<uint8_t> audio_level() const { return audio_level_; }
+  void set_audio_level(absl::optional<uint8_t> value) { audio_level_ = value; }
+
+  const absl::optional<AbsoluteCaptureTime>& absolute_capture_time() const {
+    return absolute_capture_time_;
+  }
+  void set_absolute_capture_time(
+      const absl::optional<AbsoluteCaptureTime>& value) {
+    absolute_capture_time_ = value;
+  }
+
+  int64_t receive_time_ms() const { return receive_time_ms_; }
+  void set_receive_time_ms(int64_t value) { receive_time_ms_ = value; }
+
+ private:
+  // Fields from the RTP header:
+  // https://tools.ietf.org/html/rfc3550#section-5.1
+  uint32_t ssrc_;
+  std::vector<uint32_t> csrcs_;
+  uint32_t rtp_timestamp_;
+
+  // Fields from the Audio Level header extension:
+  // https://tools.ietf.org/html/rfc6464#section-3
+  absl::optional<uint8_t> audio_level_;
+
+  // Fields from the Absolute Capture Time header extension:
+  // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time_;
+
+  // Local |webrtc::Clock|-based timestamp of when the packet was received.
+  int64_t receive_time_ms_;
+};
+
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs);
+
+inline bool operator!=(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
+  return !(lhs == rhs);
+}
+
+}  // namespace webrtc
+
+#endif  // API_RTP_PACKET_INFO_H_