1 files changed, 71 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc
new file mode 100644
index 0000000000..bed9f875ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/timing/inter_frame_delay.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2011 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 "modules/video_coding/timing/inter_frame_delay.h"
+
+#include "absl/types/optional.h"
+#include "api/units/frequency.h"
+#include "api/units/time_delta.h"
+#include "modules/include/module_common_types_public.h"
+
+namespace webrtc {
+
+namespace {
+constexpr Frequency k90kHz = Frequency::KiloHertz(90);
+}
+
+InterFrameDelay::InterFrameDelay() {
+  Reset();
+}
+
+// Resets the delay estimate.
+void InterFrameDelay::Reset() {
+  prev_wall_clock_ = absl::nullopt;
+  prev_rtp_timestamp_unwrapped_ = 0;
+}
+
+// Calculates the delay of a frame with the given timestamp.
+// This method is called when the frame is complete.
+absl::optional<TimeDelta> InterFrameDelay::CalculateDelay(
+    uint32_t rtp_timestamp,
+    Timestamp now) {
+  int64_t rtp_timestamp_unwrapped = unwrapper_.Unwrap(rtp_timestamp);
+  if (!prev_wall_clock_) {
+    // First set of data, initialization, wait for next frame.
+    prev_wall_clock_ = now;
+    prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
+    return TimeDelta::Zero();
+  }
+
+  // Account for reordering in jitter variance estimate in the future?
+  // Note that this also captures incomplete frames which are grabbed for
+  // decoding after a later frame has been complete, i.e. real packet losses.
+  uint32_t cropped_last = static_cast<uint32_t>(prev_rtp_timestamp_unwrapped_);
+  if (rtp_timestamp_unwrapped < prev_rtp_timestamp_unwrapped_ ||
+      !IsNewerTimestamp(rtp_timestamp, cropped_last)) {
+    return absl::nullopt;
+  }
+
+  // Compute the compensated timestamp difference.
+  int64_t d_rtp_ticks = rtp_timestamp_unwrapped - prev_rtp_timestamp_unwrapped_;
+  TimeDelta dts = d_rtp_ticks / k90kHz;
+  TimeDelta dt = now - *prev_wall_clock_;
+
+  // frameDelay is the difference of dT and dTS -- i.e. the difference of the
+  // wall clock time difference and the timestamp difference between two
+  // following frames.
+  TimeDelta delay = dt - dts;
+
+  prev_rtp_timestamp_unwrapped_ = rtp_timestamp_unwrapped;
+  prev_wall_clock_ = now;
+  return delay;
+}
+
+}  // namespace webrtc