Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

1 files changed, 69 insertions, 0 deletions

diff --git a/third_party/libwebrtc/rtc_base/rate_limiter.cc b/third_party/libwebrtc/rtc_base/rate_limiter.cc
new file mode 100644
index 0000000000..0f3f343aed
--- /dev/null
+++ b/third_party/libwebrtc/rtc_base/rate_limiter.cc
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2016 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 "rtc_base/rate_limiter.h"
+
+#include <limits>
+
+#include "absl/types/optional.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+RateLimiter::RateLimiter(Clock* clock, int64_t max_window_ms)
+    : clock_(clock),
+      current_rate_(max_window_ms, RateStatistics::kBpsScale),
+      window_size_ms_(max_window_ms),
+      max_rate_bps_(std::numeric_limits<uint32_t>::max()) {}
+
+RateLimiter::~RateLimiter() {}
+
+// Usage note: This class is intended be usable in a scenario where different
+// threads may call each of the the different method. For instance, a network
+// thread trying to send data calling TryUseRate(), the bandwidth estimator
+// calling SetMaxRate() and a timed maintenance thread periodically updating
+// the RTT.
+bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
+  MutexLock lock(&lock_);
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  absl::optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
+  if (current_rate) {
+    // If there is a current rate, check if adding bytes would cause maximum
+    // bitrate target to be exceeded. If there is NOT a valid current rate,
+    // allow allocating rate even if target is exceeded. This prevents
+    // problems
+    // at very low rates, where for instance retransmissions would never be
+    // allowed due to too high bitrate caused by a single packet.
+
+    size_t bitrate_addition_bps =
+        (packet_size_bytes * 8 * 1000) / window_size_ms_;
+    if (*current_rate + bitrate_addition_bps > max_rate_bps_)
+      return false;
+  }
+
+  current_rate_.Update(packet_size_bytes, now_ms);
+  return true;
+}
+
+void RateLimiter::SetMaxRate(uint32_t max_rate_bps) {
+  MutexLock lock(&lock_);
+  max_rate_bps_ = max_rate_bps;
+}
+
+// Set the window size over which to measure the current bitrate.
+// For retransmissions, this is typically the RTT.
+bool RateLimiter::SetWindowSize(int64_t window_size_ms) {
+  MutexLock lock(&lock_);
+  window_size_ms_ = window_size_ms;
+  return current_rate_.SetWindowSize(window_size_ms,
+                                     clock_->TimeInMilliseconds());
+}
+
+}  // namespace webrtc