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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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