Files
platform-external-webrtc/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
terelius e75d96b5bd Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
Reason for revert:
Resetting the estimate means that we need to start gathering data from scratch again. The combination of
1) DelayBasedEstimator not reacting to overuse unless there is a valid estimate of the acknowledged bitrate, and
2) AcknowledgedBitrateEstimator needing a significant amount of time/data to obtain an provide an estimate
causes poor performance in simulations/tests. It is not clear whether this will affect real networks negatively, but I suggest reverting this to be on the safe side.
See also https://bugs.chromium.org/p/webrtc/issues/detail?id=7884

Original issue's description:
> Test and fix for huge bwe drop after alr state.
>
> BUG=webrtc:7746
>
> Review-Url: https://codereview.webrtc.org/2931873002
> Cr-Commit-Position: refs/heads/master@{#18692}
> Committed: 37aa8ba616

TBR=solenberg@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org,holmer@chromium.org,philipel@webrtc.org,oprypin@webrtc.org,holmer@google.com,stefan@webrtc.org,tschumim@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7746

Review-Url: https://codereview.webrtc.org/2964213002
Cr-Commit-Position: refs/heads/master@{#18866}
2017-06-30 15:11:44 +00:00

116 lines
4.1 KiB
C++

/*
* Copyright (c) 2017 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/modules/congestion_controller/acknowledge_bitrate_estimator.h"
#include <cmath>
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
namespace {
constexpr int kInitialRateWindowMs = 500;
constexpr int kRateWindowMs = 150;
bool IsInSendTimeHistory(const PacketFeedback& packet) {
return packet.send_time_ms >= 0;
}
} // namespace
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
: sum_(0),
current_win_ms_(0),
prev_time_ms_(-1),
bitrate_estimate_(-1.0f),
bitrate_estimate_var_(50.0f) {}
void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
packet_feedback_vector.end(),
PacketFeedbackComparator()));
for (const auto& packet : packet_feedback_vector) {
if (IsInSendTimeHistory(packet))
Update(packet.arrival_time_ms, packet.payload_size);
}
}
void AcknowledgedBitrateEstimator::Update(int64_t now_ms, int bytes) {
int rate_window_ms = kRateWindowMs;
// We use a larger window at the beginning to get a more stable sample that
// we can use to initialize the estimate.
if (bitrate_estimate_ < 0.f)
rate_window_ms = kInitialRateWindowMs;
float bitrate_sample = UpdateWindow(now_ms, bytes, rate_window_ms);
if (bitrate_sample < 0.0f)
return;
if (bitrate_estimate_ < 0.0f) {
// This is the very first sample we get. Use it to initialize the estimate.
bitrate_estimate_ = bitrate_sample;
return;
}
// Define the sample uncertainty as a function of how far away it is from the
// current estimate.
float sample_uncertainty =
10.0f * std::abs(bitrate_estimate_ - bitrate_sample) / bitrate_estimate_;
float sample_var = sample_uncertainty * sample_uncertainty;
// Update a bayesian estimate of the rate, weighting it lower if the sample
// uncertainty is large.
// The bitrate estimate uncertainty is increased with each update to model
// that the bitrate changes over time.
float pred_bitrate_estimate_var = bitrate_estimate_var_ + 5.f;
bitrate_estimate_ = (sample_var * bitrate_estimate_ +
pred_bitrate_estimate_var * bitrate_sample) /
(sample_var + pred_bitrate_estimate_var);
bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
(sample_var + pred_bitrate_estimate_var);
BWE_TEST_LOGGING_PLOT(1, "acknowledged_bitrate", now_ms,
bitrate_estimate_ * 1000);
}
float AcknowledgedBitrateEstimator::UpdateWindow(int64_t now_ms,
int bytes,
int rate_window_ms) {
// Reset if time moves backwards.
if (now_ms < prev_time_ms_) {
prev_time_ms_ = -1;
sum_ = 0;
current_win_ms_ = 0;
}
if (prev_time_ms_ >= 0) {
current_win_ms_ += now_ms - prev_time_ms_;
// Reset if nothing has been received for more than a full window.
if (now_ms - prev_time_ms_ > rate_window_ms) {
sum_ = 0;
current_win_ms_ %= rate_window_ms;
}
}
prev_time_ms_ = now_ms;
float bitrate_sample = -1.0f;
if (current_win_ms_ >= rate_window_ms) {
bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
current_win_ms_ -= rate_window_ms;
sum_ = 0;
}
sum_ += bytes;
return bitrate_sample;
}
rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
if (bitrate_estimate_ < 0.f)
return rtc::Optional<uint32_t>();
return rtc::Optional<uint32_t>(bitrate_estimate_ * 1000);
}
} // namespace webrtc