Index: webrtc/base/rate_limiter.cc |
diff --git a/webrtc/base/rate_limiter.cc b/webrtc/base/rate_limiter.cc |
new file mode 100644 |
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+++ b/webrtc/base/rate_limiter.cc |
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+/* |
+ * 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), |
+ 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(); |
+ 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, |
+ clock_->TimeInMilliseconds()); |
+} |
+ |
+} // namespace webrtc |