Chromium Code Reviews| 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..a3011c2cec8f42434b2395d26b47746b37783845 |
| --- /dev/null |
| +++ b/webrtc/base/rate_limiter.cc |
| @@ -0,0 +1,62 @@ |
| +/* |
| + * 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_retransmission_window_ms) |
| + : clock_(clock), |
| + current_rate_(max_retransmission_window_ms, RateStatistics::kBpsScale), |
| + window_size_ms_(max_retransmission_window_ms), |
| + max_rate_bps_(std::numeric_limits<uint32_t>::max()) {} |
| + |
| +RateLimiter::~RateLimiter() {} |
| + |
| +// Try to use rate to send bytes. Returns true on success and if so updates |
| +// current rate. |
|
tommi
2016/07/07 12:48:46
keep the function level documentation in the heade
sprang_webrtc
2016/07/07 13:46:50
Done.
|
| +bool RateLimiter::TryUseRate(size_t packet_size_bytes) { |
| + rtc::CritScope cs(&lock_); |
|
tommi
2016/07/07 12:48:46
can you add documentation about threading? Why not
sprang_webrtc
2016/07/07 13:46:50
In the initial use case, TryUseRate() will be call
tommi
2016/07/08 08:52:20
Thanks, that's useful information. What about put
sprang_webrtc
2016/07/08 11:46:58
Done.
|
| + rtc::Optional<uint32_t> current_rate = |
| + current_rate_.Rate(clock_->TimeInMilliseconds()); |
| + 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, clock_->TimeInMilliseconds()); |
|
tommi
2016/07/07 12:48:46
calling out to a different implementation while ho
sprang_webrtc
2016/07/07 13:46:50
I looked into bolting rate limiting logic onto Rat
tommi
2016/07/08 08:52:20
Might make sense at some point in the future. Thi
sprang_webrtc
2016/07/08 11:46:58
Acknowledged.
|
| + return true; |
| +} |
| + |
| +// Set the maximum bitrate, in bps, that this limiter allows to send. |
| +void RateLimiter::SetMaxRate(uint32_t max_rate_bps) { |
| + rtc::CritScope cs(&lock_); |
| + max_rate_bps_ = max_rate_bps; |
|
tommi
2016/07/07 12:48:46
If this variable is changed on the same thread as
sprang_webrtc
2016/07/07 13:46:50
For the initial use case at least, SetMaxRate() an
|
| +} |
| + |
| +// Set the window size over which to measure the current bitrate. |
| +// For retransmissions, this is typically the RTT. |
| +void RateLimiter::SetWindowSize(int64_t window_size_ms) { |
|
tommi
2016/07/07 12:48:46
same for this method. keep the method level docume
sprang_webrtc
2016/07/07 13:46:50
Done.
|
| + rtc::CritScope cs(&lock_); |
| + window_size_ms_ = window_size_ms; |
| + current_rate_.SetWindowSize(window_size_ms, clock_->TimeInMilliseconds()); |
|
tommi
2016/07/07 12:48:46
calling into a different implementation while hold
sprang_webrtc
2016/07/07 13:46:50
Nope. RateStatistics does not use any locks.
tommi
2016/07/08 08:52:20
Thanks. I took a closer look at how these classes
|
| +} |
| + |
| +} // namespace webrtc |