From 81df62b4568cf377da50f0ebc0401a0c2ebfad73 Mon Sep 17 00:00:00 2001
From: Jakob Ivarsson <jakobi@webrtc.org>
Date: Wed, 17 Jul 2019 14:09:04 +0200
Subject: [PATCH] Add field trial to introduce extra delay after target level
 calculation.

Bug: webrtc:10817
Change-Id: Id9eced821df2859b2cb7174062b6f5e29e145f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145902
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28825}
---
 modules/audio_coding/neteq/delay_manager.cc   | 24 ++++++++++++++++++-
 modules/audio_coding/neteq/delay_manager.h    |  1 +
 .../neteq/delay_manager_unittest.cc           | 23 ++++++++++++++++++
 3 files changed, 47 insertions(+), 1 deletion(-)

diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index bdaa28c9dc..b101759496 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -131,6 +131,23 @@ absl::optional<int> GetDecelerationTargetLevelOffsetMs() {
   return absl::nullopt;
 }
 
+absl::optional<int> GetExtraDelayMs() {
+  constexpr char kExtraDelayFieldTrial[] = "WebRTC-Audio-NetEqExtraDelay";
+  if (!webrtc::field_trial::IsEnabled(kExtraDelayFieldTrial)) {
+    return absl::nullopt;
+  }
+
+  const auto field_trial_string =
+      webrtc::field_trial::FindFullName(kExtraDelayFieldTrial);
+  int extra_delay_ms = -1;
+  sscanf(field_trial_string.c_str(), "Enabled-%d", &extra_delay_ms);
+  if (extra_delay_ms >= 0) {
+    RTC_LOG(LS_INFO) << "NetEq extra delay in milliseconds: " << extra_delay_ms;
+    return extra_delay_ms;
+  }
+  return absl::nullopt;
+}
+
 }  // namespace
 
 namespace webrtc {
@@ -166,7 +183,8 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
           field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
       enable_rtx_handling_(enable_rtx_handling),
       deceleration_target_level_offset_ms_(
-          GetDecelerationTargetLevelOffsetMs()) {
+          GetDecelerationTargetLevelOffsetMs()),
+      extra_delay_ms_(GetExtraDelayMs()) {
   assert(peak_detector);  // Should never be NULL.
   RTC_CHECK(histogram_);
   RTC_DCHECK_GE(base_minimum_delay_ms_, 0);
@@ -385,6 +403,10 @@ int DelayManager::CalculateTargetLevel(int iat_packets, bool reordered) {
   target_level = std::max(target_level, 1);
   // Scale to Q8 and assign to member variable.
   target_level_ = target_level << 8;
+  if (extra_delay_ms_ && packet_len_ms_ > 0) {
+    int extra_delay = (extra_delay_ms_.value() << 8) / packet_len_ms_;
+    target_level_ += extra_delay;
+  }
   return target_level_;
 }
 
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index adefea1c61..851ed46147 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -209,6 +209,7 @@ class DelayManager {
   // impose deceleration to increase the buffer level. The value is in Q8, and
   // measured in milliseconds.
   const absl::optional<int> deceleration_target_level_offset_ms_;
+  const absl::optional<int> extra_delay_ms_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
 };
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index a8e2b3d18e..2d15e479e7 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -850,4 +850,27 @@ TEST_F(DelayManagerTest, DecelerationTargetLevelOffset) {
   }
 }
 
+TEST_F(DelayManagerTest, ExtraDelay) {
+  {
+    // Default behavior. Insert two packets so that a new target level is
+    // calculated.
+    SetPacketAudioLength(kFrameSizeMs);
+    InsertNextPacket();
+    IncreaseTime(kFrameSizeMs);
+    InsertNextPacket();
+    EXPECT_EQ(dm_->TargetLevel(), 1 << 8);
+  }
+  {
+    // Add 80 ms extra delay and calculate a new target level.
+    test::ScopedFieldTrials field_trial(
+        "WebRTC-Audio-NetEqExtraDelay/Enabled-80/");
+    RecreateDelayManager();
+    SetPacketAudioLength(kFrameSizeMs);
+    InsertNextPacket();
+    IncreaseTime(kFrameSizeMs);
+    InsertNextPacket();
+    EXPECT_EQ(dm_->TargetLevel(), 5 << 8);
+  }
+}
+
 }  // namespace webrtc