Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 101 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/video/video_timing.cc b/third_party/libwebrtc/api/video/video_timing.cc
new file mode 100644
index 0000000000..0483c20e66
--- /dev/null
+++ b/third_party/libwebrtc/api/video/video_timing.cc
@@ -0,0 +1,101 @@
+/*
+ *  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.
+ */
+
+#include "api/video/video_timing.h"
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
+  if (time_ms < base_ms) {
+    RTC_DLOG(LS_ERROR) << "Delta " << (time_ms - base_ms)
+                       << "ms expected to be positive";
+  }
+  return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
+}
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(TimeDelta delta) {
+  if (delta < TimeDelta::Zero()) {
+    RTC_DLOG(LS_ERROR) << "Delta " << delta.ms()
+                       << "ms expected to be positive";
+  }
+  return rtc::saturated_cast<uint16_t>(delta.ms());
+}
+
+TimingFrameInfo::TimingFrameInfo()
+    : rtp_timestamp(0),
+      capture_time_ms(-1),
+      encode_start_ms(-1),
+      encode_finish_ms(-1),
+      packetization_finish_ms(-1),
+      pacer_exit_ms(-1),
+      network_timestamp_ms(-1),
+      network2_timestamp_ms(-1),
+      receive_start_ms(-1),
+      receive_finish_ms(-1),
+      decode_start_ms(-1),
+      decode_finish_ms(-1),
+      render_time_ms(-1),
+      flags(VideoSendTiming::kNotTriggered) {}
+
+int64_t TimingFrameInfo::EndToEndDelay() const {
+  return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
+}
+
+bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
+  int64_t other_delay = other.EndToEndDelay();
+  return other_delay == -1 || EndToEndDelay() > other_delay;
+}
+
+bool TimingFrameInfo::operator<(const TimingFrameInfo& other) const {
+  return other.IsLongerThan(*this);
+}
+
+bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
+  return !IsLongerThan(other);
+}
+
+bool TimingFrameInfo::IsOutlier() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
+}
+
+bool TimingFrameInfo::IsTimerTriggered() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
+}
+
+bool TimingFrameInfo::IsInvalid() const {
+  return flags == VideoSendTiming::kInvalid;
+}
+
+std::string TimingFrameInfo::ToString() const {
+  if (IsInvalid()) {
+    return "";
+  }
+
+  char buf[1024];
+  rtc::SimpleStringBuilder sb(buf);
+
+  sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
+     << encode_finish_ms << ',' << packetization_finish_ms << ','
+     << pacer_exit_ms << ',' << network_timestamp_ms << ','
+     << network2_timestamp_ms << ',' << receive_start_ms << ','
+     << receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
+     << ',' << render_time_ms << ',' << IsOutlier() << ','
+     << IsTimerTriggered();
+
+  return sb.str();
+}
+
+}  // namespace webrtc