Update to current webrtc library

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

webrtc/api/video/video_timing.h

@@ -0,0 +1,129 @@
+/*
+ *  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_VIDEO_VIDEO_TIMING_H_
+#define API_VIDEO_VIDEO_TIMING_H_
+
+#include <stdint.h>
+
+#include <limits>
+#include <string>
+
+namespace webrtc {
+
+// Video timing timestamps in ms counted from capture_time_ms of a frame.
+// This structure represents data sent in video-timing RTP header extension.
+struct VideoSendTiming {
+  enum TimingFrameFlags : uint8_t {
+    kNotTriggered = 0,  // Timing info valid, but not to be transmitted.
+                        // Used on send-side only.
+    kTriggeredByTimer = 1 << 0,  // Frame marked for tracing by periodic timer.
+    kTriggeredBySize = 1 << 1,   // Frame marked for tracing due to size.
+    kInvalid = std::numeric_limits<uint8_t>::max()  // Invalid, ignore!
+  };
+
+  // Returns |time_ms - base_ms| capped at max 16-bit value.
+  // Used to fill this data structure as per
+  // https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
+  // 16-bit deltas of timestamps from packet capture time.
+  static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
+
+  uint16_t encode_start_delta_ms;
+  uint16_t encode_finish_delta_ms;
+  uint16_t packetization_finish_delta_ms;
+  uint16_t pacer_exit_delta_ms;
+  uint16_t network_timestamp_delta_ms;
+  uint16_t network2_timestamp_delta_ms;
+  uint8_t flags;
+};
+
+// Used to report precise timings of a 'timing frames'. Contains all important
+// timestamps for a lifetime of that specific frame. Reported as a string via
+// GetStats(). Only frame which took the longest between two GetStats calls is
+// reported.
+struct TimingFrameInfo {
+  TimingFrameInfo();
+
+  // Returns end-to-end delay of a frame, if sender and receiver timestamps are
+  // synchronized, -1 otherwise.
+  int64_t EndToEndDelay() const;
+
+  // Returns true if current frame took longer to process than |other| frame.
+  // If other frame's clocks are not synchronized, current frame is always
+  // preferred.
+  bool IsLongerThan(const TimingFrameInfo& other) const;
+
+  // Returns true if flags are set to indicate this frame was marked for tracing
+  // due to the size being outside some limit.
+  bool IsOutlier() const;
+
+  // Returns true if flags are set to indicate this frame was marked fro tracing
+  // due to cyclic timer.
+  bool IsTimerTriggered() const;
+
+  // Returns true if the timing data is marked as invalid, in which case it
+  // should be ignored.
+  bool IsInvalid() const;
+
+  std::string ToString() const;
+
+  bool operator<(const TimingFrameInfo& other) const;
+
+  bool operator<=(const TimingFrameInfo& other) const;
+
+  uint32_t rtp_timestamp;  // Identifier of a frame.
+  // All timestamps below are in local monotonous clock of a receiver.
+  // If sender clock is not yet estimated, sender timestamps
+  // (capture_time_ms ... pacer_exit_ms) are negative values, still
+  // relatively correct.
+  int64_t capture_time_ms;          // Captrue time of a frame.
+  int64_t encode_start_ms;          // Encode start time.
+  int64_t encode_finish_ms;         // Encode completion time.
+  int64_t packetization_finish_ms;  // Time when frame was passed to pacer.
+  int64_t pacer_exit_ms;  // Time when last packet was pushed out of pacer.
+  // Two in-network RTP processor timestamps: meaning is application specific.
+  int64_t network_timestamp_ms;
+  int64_t network2_timestamp_ms;
+  int64_t receive_start_ms;   // First received packet time.
+  int64_t receive_finish_ms;  // Last received packet time.
+  int64_t decode_start_ms;    // Decode start time.
+  int64_t decode_finish_ms;   // Decode completion time.
+  int64_t render_time_ms;     // Proposed render time to insure smooth playback.
+
+  uint8_t flags;  // Flags indicating validity and/or why tracing was triggered.
+};
+
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// A value < 0 indicates no change from previous valid value.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+struct VideoPlayoutDelay {
+  VideoPlayoutDelay() = default;
+  VideoPlayoutDelay(int min_ms, int max_ms) : min_ms(min_ms), max_ms(max_ms) {}
+  int min_ms = -1;
+  int max_ms = -1;
+
+  bool operator==(const VideoPlayoutDelay& rhs) const {
+    return min_ms == rhs.min_ms && max_ms == rhs.max_ms;
+  }
+};
+
+// TODO(bugs.webrtc.org/7660): Old name, delete after downstream use is updated.
+using PlayoutDelay = VideoPlayoutDelay;
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_TIMING_H_