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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
Download Patch File Download Diff File Expand all files Collapse all files

58
modules/audio_coding/neteq/delay_manager.cc
View File

@ -100,6 +100,29 @@ absl::optional<DelayHistogramConfig> GetDelayHistogramConfig() {
return absl::nullopt;
}
absl::optional<int> GetDecelerationTargetLevelOffsetMs() {
constexpr char kDecelerationTargetLevelOffsetFieldTrial[] =
"WebRTC-Audio-NetEqDecelerationTargetLevelOffset";
if (!webrtc::field_trial::IsEnabled(
kDecelerationTargetLevelOffsetFieldTrial)) {
return absl::nullopt;
}
const auto field_trial_string = webrtc::field_trial::FindFullName(
kDecelerationTargetLevelOffsetFieldTrial);
int deceleration_target_level_offset_ms = -1;
sscanf(field_trial_string.c_str(), "Enabled-%d",
&deceleration_target_level_offset_ms);
if (deceleration_target_level_offset_ms >= 0) {
RTC_LOG(LS_INFO) << "NetEq deceleration_target_level_offset "
<< "in milliseconds "
<< deceleration_target_level_offset_ms;
// Convert into Q8.
return deceleration_target_level_offset_ms << 8;
}
return absl::nullopt;
}
} // namespace
namespace webrtc {
@ -133,10 +156,14 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
last_pack_cng_or_dtmf_(1),
frame_length_change_experiment_(
field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
enable_rtx_handling_(enable_rtx_handling) {
enable_rtx_handling_(enable_rtx_handling),
deceleration_target_level_offset_ms_(
GetDecelerationTargetLevelOffsetMs()) {
assert(peak_detector); // Should never be NULL.
RTC_CHECK(histogram_);
RTC_DCHECK_GE(base_minimum_delay_ms_, 0);
RTC_DCHECK(!deceleration_target_level_offset_ms_ ||
*deceleration_target_level_offset_ms_ >= 0);
Reset();
}
@ -404,26 +431,39 @@ void DelayManager::ResetPacketIatCount() {
packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
}
void DelayManager::BufferLimits(int* lower_limit, int* higher_limit) const {
BufferLimits(target_level_, lower_limit, higher_limit);
}
// Note that |low_limit| and |higher_limit| are not assigned to
// |minimum_delay_ms_| and |maximum_delay_ms_| defined by the client of this
// class. They are computed from |target_level_| and used for decision making.
void DelayManager::BufferLimits(int* lower_limit, int* higher_limit) const {
// class. They are computed from |target_level| in Q8 and used for decision
// making.
void DelayManager::BufferLimits(int target_level,
int* lower_limit,
int* higher_limit) const {
if (!lower_limit || !higher_limit) {
RTC_LOG_F(LS_ERROR) << "NULL pointers supplied as input";
assert(false);
return;
}
// |target_level| is in Q8 already.
*lower_limit = (target_level * 3) / 4;
if (deceleration_target_level_offset_ms_ && packet_len_ms_ > 0) {
*lower_limit = std::max(
*lower_limit,
target_level - *deceleration_target_level_offset_ms_ / packet_len_ms_);
}
int window_20ms = 0x7FFF; // Default large value for legacy bit-exactness.
if (packet_len_ms_ > 0) {
window_20ms = (20 << 8) / packet_len_ms_;
}
// |target_level_| is in Q8 already.
*lower_limit = (target_level_ * 3) / 4;
// |higher_limit| is equal to |target_level_|, but should at
// least be 20 ms higher than |lower_limit_|.
*higher_limit = std::max(target_level_, *lower_limit + window_20ms);
// |higher_limit| is equal to |target_level|, but should at
// least be 20 ms higher than |lower_limit|.
*higher_limit = std::max(target_level, *lower_limit + window_20ms);
}
int DelayManager::TargetLevel() const {

13
modules/audio_coding/neteq/delay_manager.h
View File

@ -101,6 +101,9 @@ class DelayManager {
// within to the corresponding pointers. The values are in (fractions of)
// packets in Q8.
virtual void BufferLimits(int* lower_limit, int* higher_limit) const;
virtual void BufferLimits(int target_level,
int* lower_limit,
int* higher_limit) const;
// Gets the target buffer level, in (fractions of) packets in Q8.
virtual int TargetLevel() const;
@ -130,6 +133,11 @@ class DelayManager {
}
// This accessor is only intended for testing purposes.
absl::optional<int> deceleration_target_level_offset_ms() const {
return deceleration_target_level_offset_ms_;
}
// These accessors are only intended for testing purposes.
HistogramMode histogram_mode() const { return histogram_mode_; }
int histogram_quantile() const { return histogram_quantile_; }
int histogram_forget_factor() const { return histogram_->forget_factor(); }
@ -196,6 +204,11 @@ class DelayManager {
const bool enable_rtx_handling_;
int num_reordered_packets_ = 0; // Number of consecutive reordered packets.
std::deque<int> delay_history_;
// When current buffer level is more than
// |deceleration_target_level_offset_ms_| below the target level, NetEq will
// 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_;
RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
};

90
modules/audio_coding/neteq/delay_manager_unittest.cc
View File

@ -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