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Unified Diff: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc

Issue 2111813002: Revert of Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 months ago
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Index: webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index e47d4917e9fd66e46cf765b12aff157e16e8b571..a1b78a257c6620fae93bbfd9dc1a7f53fceda5e4 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -44,7 +44,7 @@
} // namespace
-SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
+SendSideBandwidthEstimation::SendSideBandwidthEstimation()
: lost_packets_since_last_loss_update_Q8_(0),
expected_packets_since_last_loss_update_(0),
bitrate_(0),
@@ -63,9 +63,7 @@
bitrate_at_2_seconds_kbps_(0),
uma_update_state_(kNoUpdate),
rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
- event_log_(event_log) {
- RTC_DCHECK(event_log);
-}
+ event_log_(nullptr) {}
SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
@@ -229,9 +227,11 @@
// rates).
bitrate_ += 1000;
- event_log_->LogBwePacketLossEvent(
- bitrate_, last_fraction_loss_,
- expected_packets_since_last_loss_update_);
+ if (event_log_) {
+ event_log_->LogBwePacketLossEvent(
+ bitrate_, last_fraction_loss_,
+ expected_packets_since_last_loss_update_);
+ }
} else if (last_fraction_loss_ <= 26) {
// Loss between 2% - 10%: Do nothing.
} else {
@@ -250,9 +250,11 @@
512.0);
has_decreased_since_last_fraction_loss_ = true;
}
- event_log_->LogBwePacketLossEvent(
- bitrate_, last_fraction_loss_,
- expected_packets_since_last_loss_update_);
+ if (event_log_) {
+ event_log_->LogBwePacketLossEvent(
+ bitrate_, last_fraction_loss_,
+ expected_packets_since_last_loss_update_);
+ }
}
}
bitrate_ = CapBitrateToThresholds(now_ms, bitrate_);
@@ -306,4 +308,9 @@
}
return bitrate;
}
+
+void SendSideBandwidthEstimation::SetEventLog(RtcEventLog* event_log) {
+ event_log_ = event_log;
+}
+
} // namespace webrtc

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