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Don't include packetization overhead in protection bitrate.

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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}
This commit is contained in:
Erik Språng 2019-01-18 10:47:07 +01:00 committed by Commit Bot
parent 466472796c
commit d15687d612
1 changed files with 13 additions and 11 deletions
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24
call/rtp_video_sender.cc
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@ -636,16 +636,15 @@ void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps,
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 @@ void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps,
// 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 {