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.cc

@@ -0,0 +1,60 @@
+/*
+ *  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.
+ */
+
+#include "api/rtp_packet_info.h"
+
+#include <algorithm>
+#include <utility>
+
+namespace webrtc {
+
+RtpPacketInfo::RtpPacketInfo()
+    : ssrc_(0), rtp_timestamp_(0), receive_time_ms_(-1) {}
+
+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)
+    : ssrc_(ssrc),
+      csrcs_(std::move(csrcs)),
+      rtp_timestamp_(rtp_timestamp),
+      audio_level_(audio_level),
+      absolute_capture_time_(absolute_capture_time),
+      receive_time_ms_(receive_time_ms) {}
+
+RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header,
+                             int64_t receive_time_ms)
+    : ssrc_(rtp_header.ssrc),
+      rtp_timestamp_(rtp_header.timestamp),
+      receive_time_ms_(receive_time_ms) {
+  const auto& extension = rtp_header.extension;
+  const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize);
+
+  csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]);
+
+  if (extension.hasAudioLevel) {
+    audio_level_ = extension.audioLevel;
+  }
+
+  absolute_capture_time_ = extension.absolute_capture_time;
+}
+
+bool operator==(const RtpPacketInfo& lhs, const RtpPacketInfo& rhs) {
+  return (lhs.ssrc() == rhs.ssrc()) && (lhs.csrcs() == rhs.csrcs()) &&
+         (lhs.rtp_timestamp() == rhs.rtp_timestamp()) &&
+         (lhs.audio_level() == rhs.audio_level()) &&
+         (lhs.absolute_capture_time() == rhs.absolute_capture_time()) &&
+         (lhs.receive_time_ms() == rhs.receive_time_ms());
+}
+
+}  // namespace webrtc