|  | /* | 
|  | *  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 <optional> | 
|  |  | 
|  | #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(); | 
|  | std::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 |