Move webrtc/{base => rtc_base}

webrtc/base/rate_limiter.cc

Index: webrtc/base/rate_limiter.cc
diff --git a/webrtc/base/rate_limiter.cc b/webrtc/base/rate_limiter.cc
deleted file mode 100644
index 9215fa0c7fe456fda20ace948d0010a3cca9de63..0000000000000000000000000000000000000000
--- a/webrtc/base/rate_limiter.cc
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- *  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 "webrtc/base/rate_limiter.h"
-#include "webrtc/system_wrappers/include/clock.h"
-
-namespace webrtc {
-
-RateLimiter::RateLimiter(const 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) {
-  rtc::CritScope cs(&lock_);
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  rtc::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) {
-  rtc::CritScope cs(&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) {
-  rtc::CritScope cs(&lock_);
-  window_size_ms_ = window_size_ms;
-  return current_rate_.SetWindowSize(window_size_ms,
-                                     clock_->TimeInMilliseconds());
-}
-
-}  // namespace webrtc