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| 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 RateLimiter::RateLimiter(const 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 max_rate_bps_(std::numeric_limits<uint32_t>::max()) {} | |
| 21 | |
| 22 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 bool RateLimiter::TryUseRate(size_t packet_size_bytes) { | |
| 30 rtc::CritScope cs(&lock_); | |
| 31 int64_t now_ms = clock_->TimeInMilliseconds(); | |
| 32 rtc::Optional<uint32_t> current_rate = current_rate_.Rate(now_ms); | |
| 33 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 (packet_size_bytes * 8 * 1000) / window_size_ms_; | |
| 43 if (*current_rate + bitrate_addition_bps > max_rate_bps_) | |
| 44 return false; | |
| 45 } | |
| 46 | |
| 47 current_rate_.Update(packet_size_bytes, now_ms); | |
| 48 return true; | |
| 49 } | |
| 50 | |
| 51 void RateLimiter::SetMaxRate(uint32_t max_rate_bps) { | |
| 52 rtc::CritScope cs(&lock_); | |
| 53 max_rate_bps_ = max_rate_bps; | |
| 54 } | |
| 55 | |
| 56 // Set the window size over which to measure the current bitrate. | |
| 57 // For retransmissions, this is typically the RTT. | |
| 58 bool RateLimiter::SetWindowSize(int64_t window_size_ms) { | |
| 59 rtc::CritScope cs(&lock_); | |
| 60 window_size_ms_ = window_size_ms; | |
| 61 return current_rate_.SetWindowSize(window_size_ms, | |
| 62 clock_->TimeInMilliseconds()); | |
| 63 } | |
| 64 | |
| 65 } // namespace webrtc | |
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