Don't include packetization overhead in protection bitrate.
If we do, the bitrate allocator will assume there can be a lot a FEC
and other things and bumps the max probing bitrate by 2x.
This caused a bunch of perf tests to change in a non-obvious way.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/115410
Bug: webrtc:10155, chromium:922396
Change-Id: I51d3611cb21d98a8fab1bfab2d8f167ed859696d
Reviewed-on: https://webrtc-review.googlesource.com/c/118043
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26319}diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 6d13821..dbc8831 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -636,16 +636,15 @@
encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
payload_bitrate_bps, framerate, fraction_loss, loss_mask_vector_, rtt);
+ uint32_t packetization_rate_bps = 0;
if (account_for_packetization_overhead_) {
- // Subtract packetization overhead from the encoder target. If rate is
- // really low, cap the overhead at 50%. Since packetization is measured over
- // an averaging window, it might intermittently be higher than encoder
- // target (eg encoder pause event), so cap it to target.
- const uint32_t packetization_rate_bps =
- std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_);
- encoder_target_rate_bps_ =
- std::max(encoder_target_rate_bps_ - packetization_rate_bps,
- encoder_target_rate_bps_ / 2);
+ // Subtract packetization overhead from the encoder target. If target rate
+ // is really low, cap the overhead at 50%. This also avoids the case where
+ // |encoder_target_rate_bps_| is 0 due to encoder pause event while the
+ // packetization rate is positive since packets are still flowing.
+ packetization_rate_bps =
+ std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
+ encoder_target_rate_bps_ -= packetization_rate_bps;
}
loss_mask_vector_.clear();
@@ -664,8 +663,11 @@
// When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
// protection_bitrate includes overhead.
- protection_bitrate_bps_ =
- bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps);
+ const uint32_t media_rate = encoder_target_rate_bps_ +
+ encoder_overhead_rate_bps +
+ packetization_rate_bps;
+ RTC_DCHECK_GE(bitrate_bps, media_rate);
+ protection_bitrate_bps_ = bitrate_bps - media_rate;
}
uint32_t RtpVideoSender::GetPayloadBitrateBps() const {
