admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Avoid modifying BWE internal state on reception of REMB feedback.

Browse Source 
Instead, cap the final bandwidth estimate by the last received cap. This allows fast rampup after a REMB cap is lifted.

Bug: webrtc:12306
Change-Id: Ia99707134ce145275460524b3e46923876fdf62f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219696
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34109}
This commit is contained in:
Christoffer Rodbro 2021-05-24 21:20:23 +02:00 committed by WebRTC LUCI CQ
parent c09c58134a
commit 6396b48b18
3 changed files with 2 additions and 11 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
View File

@ -851,7 +851,6 @@ TEST_F(GoogCcNetworkControllerTest, IsFairToTCP) {
}
TEST(GoogCcScenario, RampupOnRembCapLifted) {
ScopedFieldTrials trial("WebRTC-Bwe-ReceiverLimitCapsOnly/Enabled/");
Scenario s("googcc_unit/rampup_ramb_cap_lifted");
NetworkSimulationConfig net_conf;
net_conf.bandwidth = DataRate::KilobitsPerSec(2000);

11
modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
View File

@ -226,8 +226,7 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(
low_loss_threshold_(kDefaultLowLossThreshold),
high_loss_threshold_(kDefaultHighLossThreshold),
bitrate_threshold_(kDefaultBitrateThreshold),
loss_based_bandwidth_estimation_(key_value_config),
receiver_limit_caps_only_("Enabled") {
loss_based_bandwidth_estimation_(key_value_config) {
RTC_DCHECK(event_log);
if (BweLossExperimentIsEnabled()) {
uint32_t bitrate_threshold_kbps;
@ -240,8 +239,6 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(
bitrate_threshold_ = DataRate::KilobitsPerSec(bitrate_threshold_kbps);
}
}
ParseFieldTrial({&receiver_limit_caps_only_},
key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly"));
}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@ -310,9 +307,7 @@ int SendSideBandwidthEstimation::GetMinBitrate() const {
}
DataRate SendSideBandwidthEstimation::target_rate() const {
DataRate target = current_target_;
if (receiver_limit_caps_only_)
target = std::min(target, receiver_limit_);
DataRate target = std::min(current_target_, receiver_limit_);
return std::max(min_bitrate_configured_, target);
}
@ -585,8 +580,6 @@ void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) {
DataRate SendSideBandwidthEstimation::GetUpperLimit() const {
DataRate upper_limit = delay_based_limit_;
if (!receiver_limit_caps_only_)
upper_limit = std::min(upper_limit, receiver_limit_);
upper_limit = std::min(upper_limit, max_bitrate_configured_);
return upper_limit;
}

1
modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
View File

@ -190,7 +190,6 @@ class SendSideBandwidthEstimation {
float high_loss_threshold_;
DataRate bitrate_threshold_;
LossBasedBandwidthEstimation loss_based_bandwidth_estimation_;
FieldTrialFlag receiver_limit_caps_only_;
};
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_