blob: 89bdb94e08bc3136853b544b60a0c866f30ed2a3 [file] [log] [blame]
* 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(Clock* clock, int64_t max_window_ms)
: clock_(clock),
current_rate_(max_window_ms, RateStatistics::kBpsScale),
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,
} // namespace webrtc