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Cleaup WebRTC-AddRttToPlayoutDelay field trial

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Field trial is not used in any rollouts and should be removed.

R=mhoro@webrtc.org

Bug: webrtc:13264
Change-Id: Ib896dcdec81db7c3f4e68a8dda266d96dfdc6aed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234865
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35195}
This commit is contained in:
Evan Shrubsole 2021-10-13 11:19:48 +02:00 committed by WebRTC LUCI CQ
parent af1038d97c
commit 9146d765e2
3 changed files with 2 additions and 31 deletions
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4
modules/video_coding/frame_buffer2.cc
View File

@ -65,8 +65,6 @@ FrameBuffer::FrameBuffer(Clock* clock,
protection_mode_(kProtectionNack),
stats_callback_(stats_callback),
last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
add_rtt_to_playout_delay_(
webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")),
rtt_mult_settings_(RttMultExperiment::GetRttMultValue()),
zero_playout_delay_max_decode_queue_size_(
"max_decode_queue_size",
@ -309,7 +307,7 @@ EncodedFrame* FrameBuffer::GetNextFrame() {
jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms));
timing_->UpdateCurrentDelay(render_time_ms, now_ms);
} else {
if (RttMultExperiment::RttMultEnabled() || add_rtt_to_playout_delay_)
if (RttMultExperiment::RttMultEnabled())
jitter_estimator_.FrameNacked();
}

2
modules/video_coding/frame_buffer2.h
View File

@ -185,8 +185,6 @@ class FrameBuffer {
VCMReceiveStatisticsCallback* const stats_callback_;
int64_t last_log_non_decoded_ms_ RTC_GUARDED_BY(mutex_);
const bool add_rtt_to_playout_delay_;
// rtt_mult experiment settings.
const absl::optional<RttMultExperiment::Settings> rtt_mult_settings_;

27
modules/video_coding/frame_buffer2_unittest.cc
View File

@ -140,8 +140,7 @@ class TestFrameBuffer2 : public ::testing::Test {
static constexpr size_t kFrameSize = 10;
TestFrameBuffer2()
: trial_("WebRTC-AddRttToPlayoutDelay/Enabled/"),
time_controller_(Timestamp::Seconds(0)),
: time_controller_(Timestamp::Seconds(0)),
time_task_queue_(
time_controller_.GetTaskQueueFactory()->CreateTaskQueue(
"extract queue",
@ -232,8 +231,6 @@ class TestFrameBuffer2 : public ::testing::Test {
uint32_t Rand() { return rand_.Rand<uint32_t>(); }
// The ProtectionMode tests depends on rtt-multiplier experiment.
test::ScopedFieldTrials trial_;
webrtc::GlobalSimulatedTimeController time_controller_;
rtc::TaskQueue time_task_queue_;
VCMTimingFake timing_;
@ -459,28 +456,6 @@ TEST_F(TestFrameBuffer2, ProtectionModeNackFEC) {
EXPECT_LT(timing_.GetCurrentJitter(), kRttMs);
}
TEST_F(TestFrameBuffer2, ProtectionModeNack) {
uint16_t pid = Rand();
uint32_t ts = Rand();
constexpr int64_t kRttMs = 200;
buffer_->UpdateRtt(kRttMs);
// Jitter estimate includes RTT (after 3 retransmitted packets)
buffer_->SetProtectionMode(kProtectionNack);
InsertNackedFrame(pid, ts);
InsertNackedFrame(pid + 1, ts + 100);
InsertNackedFrame(pid + 2, ts + 200);
InsertFrame(pid + 3, 0, ts + 300, true, kFrameSize);
ExtractFrame();
ExtractFrame();
ExtractFrame();
ExtractFrame();
ASSERT_EQ(4u, frames_.size());
EXPECT_GT(timing_.GetCurrentJitter(), kRttMs);
}
TEST_F(TestFrameBuffer2, NoContinuousFrame) {
uint16_t pid = Rand();
uint32_t ts = Rand();