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Default enable sending transport sequence numbers on audio packets.

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This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
This commit is contained in:
Jakob Ivarsson
2020-11-23 15:05:44 +01:00
committed by Commit Bot
parent d840c8fb5d
commit 47a03e8743
10 changed files with 8 additions and 57 deletions
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2
call/rampup_tests.cc
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@ -663,7 +663,6 @@ TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) {
UpDownUpAudioVideoTransportSequenceNumberRtx
#endif
TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) {
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(3, 1, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,
@ -672,7 +671,6 @@ TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) {
}
TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) {
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
std::vector<int> loss_rates = {0, 0, 0, 0};
RampUpDownUpTester test(0, 1, 0, kStartBitrateBps,
RtpExtension::kTransportSequenceNumberUri, true,