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0a1d189e50
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'rtc_base'
Then manually fix where Optional was used without rtc prefix (patchset#3)
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I825f80cc8089747876ba6316d9e7c30e05716974
Reviewed-on: https://webrtc-review.googlesource.com/84585
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23700}
65 lines
2.4 KiB
C++
65 lines
2.4 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtc_base/rate_limiter.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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RateLimiter::RateLimiter(const Clock* clock, int64_t max_window_ms)
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: clock_(clock),
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current_rate_(max_window_ms, RateStatistics::kBpsScale),
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window_size_ms_(max_window_ms),
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max_rate_bps_(std::numeric_limits<uint32_t>::max()) {}
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RateLimiter::~RateLimiter() {}
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// Usage note: This class is intended be usable in a scenario where different
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// threads may call each of the the different method. For instance, a network
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// thread trying to send data calling TryUseRate(), the bandwidth estimator
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// calling SetMaxRate() and a timed maintenance thread periodically updating
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// the RTT.
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bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
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rtc::CritScope cs(&lock_);
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int64_t now_ms = clock_->TimeInMilliseconds();
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absl::optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
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if (current_rate) {
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// If there is a current rate, check if adding bytes would cause maximum
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// bitrate target to be exceeded. If there is NOT a valid current rate,
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// allow allocating rate even if target is exceeded. This prevents
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// problems
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// at very low rates, where for instance retransmissions would never be
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// allowed due to too high bitrate caused by a single packet.
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size_t bitrate_addition_bps =
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(packet_size_bytes * 8 * 1000) / window_size_ms_;
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if (*current_rate + bitrate_addition_bps > max_rate_bps_)
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return false;
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}
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current_rate_.Update(packet_size_bytes, now_ms);
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return true;
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}
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void RateLimiter::SetMaxRate(uint32_t max_rate_bps) {
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rtc::CritScope cs(&lock_);
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max_rate_bps_ = max_rate_bps;
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}
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// Set the window size over which to measure the current bitrate.
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// For retransmissions, this is typically the RTT.
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bool RateLimiter::SetWindowSize(int64_t window_size_ms) {
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rtc::CritScope cs(&lock_);
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window_size_ms_ = window_size_ms;
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return current_rate_.SetWindowSize(window_size_ms,
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clock_->TimeInMilliseconds());
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}
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} // namespace webrtc
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