| /* | |
| * 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 <cstddef> | |
| #include <cstdint> | |
| #include <limits> | |
| #include <optional> | |
| #include "rtc_base/rate_statistics.h" | |
| #include "rtc_base/synchronization/mutex.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(); | |
| 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 |