Update to current webrtc library

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

webrtc/api/rtp_headers.h

@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2017 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_HEADERS_H_
+#define API_RTP_HEADERS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/units/timestamp.h"
+#include "api/video/color_space.h"
+#include "api/video/video_content_type.h"
+#include "api/video/video_rotation.h"
+#include "api/video/video_timing.h"
+
+namespace webrtc {
+
+struct FeedbackRequest {
+  // Determines whether the recv delta as specified in
+  // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
+  // should be included.
+  bool include_timestamps;
+  // Include feedback of received packets in the range [sequence_number -
+  // sequence_count + 1, sequence_number]. That is, no feedback will be sent if
+  // sequence_count is zero.
+  int sequence_count;
+};
+
+// The Absolute Capture Time extension is used to stamp RTP packets with a NTP
+// timestamp showing when the first audio or video frame in a packet was
+// originally captured. The intent of this extension is to provide a way to
+// accomplish audio-to-video synchronization when RTCP-terminating intermediate
+// systems (e.g. mixers) are involved. See:
+// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time
+struct AbsoluteCaptureTime {
+  // Absolute capture timestamp is the NTP timestamp of when the first frame in
+  // a packet was originally captured. This timestamp MUST be based on the same
+  // clock as the clock used to generate NTP timestamps for RTCP sender reports
+  // on the capture system.
+  //
+  // It’s not always possible to do an NTP clock readout at the exact moment of
+  // when a media frame is captured. A capture system MAY postpone the readout
+  // until a more convenient time. A capture system SHOULD have known delays
+  // (e.g. from hardware buffers) subtracted from the readout to make the final
+  // timestamp as close to the actual capture time as possible.
+  //
+  // This field is encoded as a 64-bit unsigned fixed-point number with the high
+  // 32 bits for the timestamp in seconds and low 32 bits for the fractional
+  // part. This is also known as the UQ32.32 format and is what the RTP
+  // specification defines as the canonical format to represent NTP timestamps.
+  uint64_t absolute_capture_timestamp;
+
+  // Estimated capture clock offset is the sender’s estimate of the offset
+  // between its own NTP clock and the capture system’s NTP clock. The sender is
+  // here defined as the system that owns the NTP clock used to generate the NTP
+  // timestamps for the RTCP sender reports on this stream. The sender system is
+  // typically either the capture system or a mixer.
+  //
+  // This field is encoded as a 64-bit two’s complement signed fixed-point
+  // number with the high 32 bits for the seconds and low 32 bits for the
+  // fractional part. It’s intended to make it easy for a receiver, that knows
+  // how to estimate the sender system’s NTP clock, to also estimate the capture
+  // system’s NTP clock:
+  //
+  //   Capture NTP Clock = Sender NTP Clock + Capture Clock Offset
+  absl::optional<int64_t> estimated_capture_clock_offset;
+};
+
+inline bool operator==(const AbsoluteCaptureTime& lhs,
+                       const AbsoluteCaptureTime& rhs) {
+  return (lhs.absolute_capture_timestamp == rhs.absolute_capture_timestamp) &&
+         (lhs.estimated_capture_clock_offset ==
+          rhs.estimated_capture_clock_offset);
+}
+
+inline bool operator!=(const AbsoluteCaptureTime& lhs,
+                       const AbsoluteCaptureTime& rhs) {
+  return !(lhs == rhs);
+}
+
+struct RTPHeaderExtension {
+  RTPHeaderExtension();
+  RTPHeaderExtension(const RTPHeaderExtension& other);
+  RTPHeaderExtension& operator=(const RTPHeaderExtension& other);
+
+  static constexpr int kAbsSendTimeFraction = 18;
+
+  Timestamp GetAbsoluteSendTimestamp() const {
+    RTC_DCHECK(hasAbsoluteSendTime);
+    RTC_DCHECK(absoluteSendTime < (1ul << 24));
+    return Timestamp::Micros((absoluteSendTime * 1000000ll) /
+                             (1 << kAbsSendTimeFraction));
+  }
+
+  TimeDelta GetAbsoluteSendTimeDelta(uint32_t previous_sendtime) const {
+    RTC_DCHECK(hasAbsoluteSendTime);
+    RTC_DCHECK(absoluteSendTime < (1ul << 24));
+    RTC_DCHECK(previous_sendtime < (1ul << 24));
+    int32_t delta =
+        static_cast<int32_t>((absoluteSendTime - previous_sendtime) << 8) >> 8;
+    return TimeDelta::Micros((delta * 1000000ll) / (1 << kAbsSendTimeFraction));
+  }
+
+  bool hasTransmissionTimeOffset;
+  int32_t transmissionTimeOffset;
+  bool hasAbsoluteSendTime;
+  uint32_t absoluteSendTime;
+  absl::optional<AbsoluteCaptureTime> absolute_capture_time;
+  bool hasTransportSequenceNumber;
+  uint16_t transportSequenceNumber;
+  absl::optional<FeedbackRequest> feedback_request;
+
+  // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
+  // https://tools.ietf.org/html/rfc6464#section-3
+  bool hasAudioLevel;
+  bool voiceActivity;
+  uint8_t audioLevel;
+
+  // For Coordination of Video Orientation. See
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
+  // ts_126114v120700p.pdf
+  bool hasVideoRotation;
+  VideoRotation videoRotation;
+
+  // TODO(ilnik): Refactor this and one above to be absl::optional() and remove
+  // a corresponding bool flag.
+  bool hasVideoContentType;
+  VideoContentType videoContentType;
+
+  bool has_video_timing;
+  VideoSendTiming video_timing;
+
+  VideoPlayoutDelay playout_delay;
+
+  // For identification of a stream when ssrc is not signaled. See
+  // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
+  // TODO(danilchap): Update url from draft to release version.
+  std::string stream_id;
+  std::string repaired_stream_id;
+
+  // For identifying the media section used to interpret this RTP packet. See
+  // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
+  std::string mid;
+
+  absl::optional<ColorSpace> color_space;
+};
+
+enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13
+
+struct RTPHeader {
+  RTPHeader();
+  RTPHeader(const RTPHeader& other);
+  RTPHeader& operator=(const RTPHeader& other);
+
+  bool markerBit;
+  uint8_t payloadType;
+  uint16_t sequenceNumber;
+  uint32_t timestamp;
+  uint32_t ssrc;
+  uint8_t numCSRCs;
+  uint32_t arrOfCSRCs[kRtpCsrcSize];
+  size_t paddingLength;
+  size_t headerLength;
+  int payload_type_frequency;
+  RTPHeaderExtension extension;
+};
+
+// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
+// RTCP mode is described by RFC 5506.
+enum class RtcpMode { kOff, kCompound, kReducedSize };
+
+enum NetworkState {
+  kNetworkUp,
+  kNetworkDown,
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_HEADERS_H_