webrtc/rtc_base/rate_limiter.cc
Sebastian Jansson aa01f27667 Removes all const Clock*.
This prepares for making the Clock interface fully mutable.

Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.

Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
2019-01-30 13:03:37 +00:00

69 lines
2.5 KiB
C++

/*
* 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 "absl/types/optional.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) {
rtc::CritScope cs(&lock_);
int64_t now_ms = clock_->TimeInMilliseconds();
absl::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,
clock_->TimeInMilliseconds());
}
} // namespace webrtc