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Add DecelerationTargetLevelOffset Field Trial.

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This change allows NetEq to reach preferred jitter buffer size much faster
for high target delays because it uses absolute units instead of relative ones
during computation of lower_limit.

More details can be found here:
https://docs.google.com/document/d/12qPMJYFhGXrA_o_nvz9VshpzAJX6aULxFig1fTzBzDI/edit

Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5

Bug: webrtc:10619
Change-Id: I21ce0e35e25166d935fdf0325c083bcf990899f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135745
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27970}
This commit is contained in:
Ruslan Burakov
2019-05-15 14:31:22 +02:00
committed by Commit Bot
parent 8e1a0080d3
commit 1e193faaf1
3 changed files with 152 additions and 9 deletions
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90
modules/audio_coding/neteq/delay_manager_unittest.cc
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@ -107,6 +107,7 @@ void DelayManagerTest::IncreaseTime(int inc_ms) {
tick_timer_.Increment();
}
}
void DelayManagerTest::TearDown() {
EXPECT_CALL(detector_, Die());
}
@ -725,4 +726,93 @@ TEST_F(DelayManagerTest, RelativeArrivalDelayStatistic) {
InsertNextPacket();
}
TEST_F(DelayManagerTest, DecelerationTargetLevelOffsetFieldTrial) {
{
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled-105/");
RecreateDelayManager();
EXPECT_EQ(dm_->deceleration_target_level_offset_ms().value(), 105 << 8);
}
{
// Negative number.
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled--105/");
RecreateDelayManager();
EXPECT_FALSE(dm_->deceleration_target_level_offset_ms().has_value());
}
{
// Disabled.
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Disabled/");
RecreateDelayManager();
EXPECT_FALSE(dm_->deceleration_target_level_offset_ms().has_value());
}
{
// Float number.
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled-105.5/");
RecreateDelayManager();
EXPECT_EQ(dm_->deceleration_target_level_offset_ms().value(), 105 << 8);
}
{
// Several numbers.
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled-20-40/");
RecreateDelayManager();
EXPECT_EQ(dm_->deceleration_target_level_offset_ms().value(), 20 << 8);
}
}
TEST_F(DelayManagerTest, DecelerationTargetLevelOffset) {
// Border value where 1/4 target buffer level meets
// WebRTC-Audio-NetEqDecelerationTargetLevelOffset.
constexpr int kBoarderTargetLevel = 100 * 4;
{
// Test that for a low target level, default behaviour is intact.
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled-100/");
const int target_level_ms = ((kBoarderTargetLevel - 1) << 8) / kFrameSizeMs;
RecreateDelayManager();
SetPacketAudioLength(kFrameSizeMs);
int lower, higher; // In Q8.
dm_->BufferLimits(target_level_ms, &lower, &higher);
// Default behaviour of taking 75% of target level.
EXPECT_EQ(target_level_ms * 3 / 4, lower);
EXPECT_EQ(target_level_ms, higher);
}
{
// Test that for the high target level, |lower| is below target level by
// fixed constant (100 ms in this Field Trial setup).
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset/Enabled-100/");
const int target_level_ms = ((kBoarderTargetLevel + 1) << 8) / kFrameSizeMs;
RecreateDelayManager();
SetPacketAudioLength(kFrameSizeMs);
int lower, higher; // In Q8.
dm_->BufferLimits(target_level_ms, &lower, &higher);
EXPECT_EQ(target_level_ms - ((100 << 8) / kFrameSizeMs), lower);
EXPECT_EQ(target_level_ms, higher);
}
{
// Test that for the high target level, without Field Trial the behaviour
// will remain the same.
const int target_level_ms = ((kBoarderTargetLevel + 1) << 8) / kFrameSizeMs;
RecreateDelayManager();
SetPacketAudioLength(kFrameSizeMs);
int lower, higher; // In Q8.
dm_->BufferLimits(target_level_ms, &lower, &higher);
// Default behaviour of taking 75% of target level.
EXPECT_EQ(target_level_ms * 3 / 4, lower);
EXPECT_EQ(target_level_ms, higher);
}
}
} // namespace webrtc