Line data    Source code

       1             : /*
       2             :  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
       3             :  *
       4             :  *  Use of this source code is governed by a BSD-style license
       5             :  *  that can be found in the LICENSE file in the root of the source
       6             :  *  tree. An additional intellectual property rights grant can be found
       7             :  *  in the file PATENTS.  All contributing project authors may
       8             :  *  be found in the AUTHORS file in the root of the source tree.
       9             :  */
      10             : 
      11             : #include "webrtc/base/rate_limiter.h"
      12             : #include "webrtc/system_wrappers/include/clock.h"
      13             : 
      14             : namespace webrtc {
      15             : 
      16           0 : RateLimiter::RateLimiter(Clock* clock, int64_t max_window_ms)
      17             :     : clock_(clock),
      18             :       current_rate_(max_window_ms, RateStatistics::kBpsScale),
      19             :       window_size_ms_(max_window_ms),
      20           0 :       max_rate_bps_(std::numeric_limits<uint32_t>::max()) {}
      21             : 
      22           0 : RateLimiter::~RateLimiter() {}
      23             : 
      24             : // Usage note: This class is intended be usable in a scenario where different
      25             : // threads may call each of the the different method. For instance, a network
      26             : // thread trying to send data calling TryUseRate(), the bandwidth estimator
      27             : // calling SetMaxRate() and a timed maintenance thread periodically updating
      28             : // the RTT.
      29           0 : bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
      30           0 :   rtc::CritScope cs(&lock_);
      31           0 :   int64_t now_ms = clock_->TimeInMilliseconds();
      32           0 :   rtc::Optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
      33           0 :   if (current_rate) {
      34             :     // If there is a current rate, check if adding bytes would cause maximum
      35             :     // bitrate target to be exceeded. If there is NOT a valid current rate,
      36             :     // allow allocating rate even if target is exceeded. This prevents
      37             :     // problems
      38             :     // at very low rates, where for instance retransmissions would never be
      39             :     // allowed due to too high bitrate caused by a single packet.
      40             : 
      41             :     size_t bitrate_addition_bps =
      42           0 :         (packet_size_bytes * 8 * 1000) / window_size_ms_;
      43           0 :     if (*current_rate + bitrate_addition_bps > max_rate_bps_)
      44           0 :       return false;
      45             :   }
      46             : 
      47           0 :   current_rate_.Update(packet_size_bytes, now_ms);
      48           0 :   return true;
      49             : }
      50             : 
      51           0 : void RateLimiter::SetMaxRate(uint32_t max_rate_bps) {
      52           0 :   rtc::CritScope cs(&lock_);
      53           0 :   max_rate_bps_ = max_rate_bps;
      54           0 : }
      55             : 
      56             : // Set the window size over which to measure the current bitrate.
      57             : // For retransmissions, this is typically the RTT.
      58           0 : bool RateLimiter::SetWindowSize(int64_t window_size_ms) {
      59           0 :   rtc::CritScope cs(&lock_);
      60           0 :   window_size_ms_ = window_size_ms;
      61           0 :   return current_rate_.SetWindowSize(window_size_ms,
      62           0 :                                      clock_->TimeInMilliseconds());
      63             : }
      64             : 
      65             : }  // namespace webrtc