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Issue 2061423003: Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 4 years, 5 months ago
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1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/base/rate_limiter.h"
12 #include "webrtc/system_wrappers/include/clock.h"
13
14 namespace webrtc {
15
16 RateLimiter::RateLimiter(Clock* clock, int64_t max_window_ms)
17 : clock_(clock),
18 current_rate_(max_window_ms, RateStatistics::kBpsScale),
19 window_size_ms_(max_window_ms),
20 max_rate_bps_(std::numeric_limits<uint32_t>::max()) {}
21
22 RateLimiter::~RateLimiter() {}
23
24 bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
25 rtc::CritScope cs(&lock_);
26 rtc::Optional<uint32_t> current_rate =
27 current_rate_.Rate(clock_->TimeInMilliseconds());
28 if (current_rate) {
29 // If there is a current rate, check if adding bytes would cause maximum
30 // bitrate target to be exceeded. If there is NOT a valid current rate,
31 // allow allocating rate even if target is exceeded. This prevents
32 // problems
33 // at very low rates, where for instance retransmissions would never be
34 // allowed due to too high bitrate caused by a single packet.
35
36 size_t bitrate_addition_bps =
37 (packet_size_bytes * 8 * 1000) / window_size_ms_;
38 if (*current_rate + bitrate_addition_bps > max_rate_bps_)
39 return false;
40 }
41
42 current_rate_.Update(packet_size_bytes, clock_->TimeInMilliseconds());
tommi 2016/07/08 08:52:20 nit: just call TimeInMilliseconds once
sprang_webrtc 2016/07/08 11:46:58 Done.
43 return true;
44 }
45
46 void RateLimiter::SetMaxRate(uint32_t max_rate_bps) {
47 rtc::CritScope cs(&lock_);
48 max_rate_bps_ = max_rate_bps;
49 }
50
51 // Set the window size over which to measure the current bitrate.
52 // For retransmissions, this is typically the RTT.
53 void RateLimiter::SetWindowSize(int64_t window_size_ms) {
54 rtc::CritScope cs(&lock_);
55 window_size_ms_ = window_size_ms;
56 current_rate_.SetWindowSize(window_size_ms, clock_->TimeInMilliseconds());
57 }
58
59 } // namespace webrtc
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