commit 1e193faaf16ca34226f5a735db1d37347a57929a
author Ruslan Burakov <kuddai@google.com> Wed May 15 14:31:22 2019 +0200
committer Commit Bot <commit-bot@chromium.org> Fri May 17 08:08:12 2019 +0000
tree d03075f26cfe11a632b07935e26e16654b480b61
parent 8e1a0080d3a74b7d6f089175526e56b6852aef8e

Add DecelerationTargetLevelOffset Field Trial.

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}

modules/audio_coding/neteq/delay_manager_unittest.cc

diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index c57f074..eb1fabc 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -107,6 +107,7 @@
     tick_timer_.Increment();
   }
 }
+
 void DelayManagerTest::TearDown() {
   EXPECT_CALL(detector_, Die());
 }
@@ -725,4 +726,93 @@
   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