From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 .../source/absolute_capture_time_interpolator.cc   | 110 +++++++++++++++++++++
 1 file changed, 110 insertions(+)
 create mode 100644 third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc

(limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc')

diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
new file mode 100644
index 0000000000..f151084c7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.cc
@@ -0,0 +1,110 @@
+/*
+ *  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 "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+
+#include <limits>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AbsoluteCaptureTimeInterpolator::AbsoluteCaptureTimeInterpolator(Clock* clock)
+    : clock_(clock) {}
+
+uint32_t AbsoluteCaptureTimeInterpolator::GetSource(
+    uint32_t ssrc,
+    rtc::ArrayView<const uint32_t> csrcs) {
+  if (csrcs.empty()) {
+    return ssrc;
+  }
+
+  return csrcs[0];
+}
+
+absl::optional<AbsoluteCaptureTime>
+AbsoluteCaptureTimeInterpolator::OnReceivePacket(
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    const absl::optional<AbsoluteCaptureTime>& received_extension) {
+  const Timestamp receive_time = clock_->CurrentTime();
+
+  MutexLock lock(&mutex_);
+
+  if (received_extension == absl::nullopt) {
+    if (!ShouldInterpolateExtension(receive_time, source, rtp_timestamp,
+                                    rtp_clock_frequency_hz)) {
+      last_receive_time_ = Timestamp::MinusInfinity();
+      return absl::nullopt;
+    }
+
+    return AbsoluteCaptureTime{
+        .absolute_capture_timestamp = InterpolateAbsoluteCaptureTimestamp(
+            rtp_timestamp, rtp_clock_frequency_hz, last_rtp_timestamp_,
+            last_received_extension_.absolute_capture_timestamp),
+        .estimated_capture_clock_offset =
+            last_received_extension_.estimated_capture_clock_offset,
+    };
+  } else {
+    last_source_ = source;
+    last_rtp_timestamp_ = rtp_timestamp;
+    last_rtp_clock_frequency_hz_ = rtp_clock_frequency_hz;
+    last_received_extension_ = *received_extension;
+
+    last_receive_time_ = receive_time;
+
+    return received_extension;
+  }
+}
+
+uint64_t AbsoluteCaptureTimeInterpolator::InterpolateAbsoluteCaptureTimestamp(
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz,
+    uint32_t last_rtp_timestamp,
+    uint64_t last_absolute_capture_timestamp) {
+  RTC_DCHECK_GT(rtp_clock_frequency_hz, 0);
+
+  return last_absolute_capture_timestamp +
+         static_cast<int64_t>(uint64_t{rtp_timestamp - last_rtp_timestamp}
+                              << 32) /
+             rtp_clock_frequency_hz;
+}
+
+bool AbsoluteCaptureTimeInterpolator::ShouldInterpolateExtension(
+    Timestamp receive_time,
+    uint32_t source,
+    uint32_t rtp_timestamp,
+    int rtp_clock_frequency_hz) const {
+  // Shouldn't if the last received extension is not eligible for interpolation,
+  // in particular if we don't have a previously received extension stored.
+  if (receive_time - last_receive_time_ > kInterpolationMaxInterval) {
+    return false;
+  }
+
+  // Shouldn't if the source has changed.
+  if (last_source_ != source) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency has changed.
+  if (last_rtp_clock_frequency_hz_ != rtp_clock_frequency_hz) {
+    return false;
+  }
+
+  // Shouldn't if the RTP clock frequency is invalid.
+  if (rtp_clock_frequency_hz <= 0) {
+    return false;
+  }
+
+  return true;
+}
+
+}  // namespace webrtc
-- 
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