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Unified Diff: webrtc/base/rate_limiter.cc

Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed nit Created 4 years, 5 months ago
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Index: webrtc/base/rate_limiter.cc
diff --git a/webrtc/base/rate_limiter.cc b/webrtc/base/rate_limiter.cc
new file mode 100644
index 0000000000000000000000000000000000000000..89bdb94e08bc3136853b544b60a0c866f30ed2a3
--- /dev/null
+++ b/webrtc/base/rate_limiter.cc
@@ -0,0 +1,65 @@
+/*
+ * 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
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