NetEq: Use TickTimer in DelayManager
This change replaces packet_iat_count_ms_ and max_timer_ms_, two
time-counting member variables in DelayManager, with Stopwatch objects
obtained from a TickTimer.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1929863002
Cr-Commit-Position: refs/heads/master@{#12554}
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic.cc b/webrtc/modules/audio_coding/neteq/decision_logic.cc
index 39bb466..b702e6d 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -152,10 +152,6 @@
void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
Modes prev_mode) {
- const int elapsed_time_ms =
- static_cast<int>(output_size_samples_ / (8 * fs_mult_));
- delay_manager_->UpdateCounters(elapsed_time_ms);
-
// Do not update buffer history if currently playing CNG since it will bias
// the filtered buffer level.
if ((prev_mode != kModeRfc3389Cng) && (prev_mode != kModeCodecInternalCng)) {
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
index 350821c..7165d93 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -28,7 +28,7 @@
TickTimer tick_timer;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer);
- DelayManager delay_manager(240, &delay_peak_detector);
+ DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(fs_hz, output_size_samples,
kPlayoutOn, &decoder_database,
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index e955f17..84bda7c 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -24,12 +24,13 @@
namespace webrtc {
DelayManager::DelayManager(size_t max_packets_in_buffer,
- DelayPeakDetector* peak_detector)
+ DelayPeakDetector* peak_detector,
+ const TickTimer* tick_timer)
: first_packet_received_(false),
max_packets_in_buffer_(max_packets_in_buffer),
iat_vector_(kMaxIat + 1, 0),
iat_factor_(0),
- packet_iat_count_ms_(0),
+ tick_timer_(tick_timer),
base_target_level_(4), // In Q0 domain.
target_level_(base_target_level_ << 8), // In Q8 domain.
packet_len_ms_(0),
@@ -41,7 +42,6 @@
maximum_delay_ms_(target_level_),
iat_cumulative_sum_(0),
max_iat_cumulative_sum_(0),
- max_timer_ms_(0),
peak_detector_(*peak_detector),
last_pack_cng_or_dtmf_(1) {
assert(peak_detector); // Should never be NULL.
@@ -79,7 +79,7 @@
if (!first_packet_received_) {
// Prepare for next packet arrival.
- packet_iat_count_ms_ = 0;
+ packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
last_seq_no_ = sequence_number;
last_timestamp_ = timestamp;
first_packet_received_ = true;
@@ -106,7 +106,7 @@
// Calculate inter-arrival time (IAT) in integer "packet times"
// (rounding down). This is the value used as index to the histogram
// vector |iat_vector_|.
- int iat_packets = packet_iat_count_ms_ / packet_len_ms;
+ int iat_packets = packet_iat_stopwatch_->ElapsedMs() / packet_len_ms;
if (streaming_mode_) {
UpdateCumulativeSums(packet_len_ms, sequence_number);
@@ -137,7 +137,7 @@
} // End if (packet_len_ms > 0).
// Prepare for next packet arrival.
- packet_iat_count_ms_ = 0;
+ packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
last_seq_no_ = sequence_number;
last_timestamp_ = timestamp;
return 0;
@@ -147,7 +147,8 @@
uint16_t sequence_number) {
// Calculate IAT in Q8, including fractions of a packet (i.e., more
// accurate than |iat_packets|.
- int iat_packets_q8 = (packet_iat_count_ms_ << 8) / packet_len_ms;
+ int iat_packets_q8 =
+ (packet_iat_stopwatch_->ElapsedMs() << 8) / packet_len_ms;
// Calculate cumulative sum IAT with sequence number compensation. The sum
// is zero if there is no clock-drift.
iat_cumulative_sum_ += (iat_packets_q8 -
@@ -159,9 +160,9 @@
if (iat_cumulative_sum_ > max_iat_cumulative_sum_) {
// Found a new maximum.
max_iat_cumulative_sum_ = iat_cumulative_sum_;
- max_timer_ms_ = 0;
+ max_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
}
- if (max_timer_ms_ > kMaxStreamingPeakPeriodMs) {
+ if (max_iat_stopwatch_->ElapsedMs() > kMaxStreamingPeakPeriodMs) {
// Too long since the last maximum was observed; decrease max value.
max_iat_cumulative_sum_ -= kCumulativeSumDrift;
}
@@ -299,7 +300,7 @@
}
packet_len_ms_ = length_ms;
peak_detector_.SetPacketAudioLength(packet_len_ms_);
- packet_iat_count_ms_ = 0;
+ packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
last_pack_cng_or_dtmf_ = 1; // TODO(hlundin): Legacy. Remove?
return 0;
}
@@ -311,8 +312,8 @@
peak_detector_.Reset();
ResetHistogram(); // Resets target levels too.
iat_factor_ = 0; // Adapt the histogram faster for the first few packets.
- packet_iat_count_ms_ = 0;
- max_timer_ms_ = 0;
+ packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
+ max_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
iat_cumulative_sum_ = 0;
max_iat_cumulative_sum_ = 0;
last_pack_cng_or_dtmf_ = 1;
@@ -340,13 +341,10 @@
return peak_detector_.peak_found();
}
-void DelayManager::UpdateCounters(int elapsed_time_ms) {
- packet_iat_count_ms_ += elapsed_time_ms;
- max_timer_ms_ += elapsed_time_ms;
+void DelayManager::ResetPacketIatCount() {
+ packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
}
-void DelayManager::ResetPacketIatCount() { packet_iat_count_ms_ = 0; }
-
// 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.
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.h b/webrtc/modules/audio_coding/neteq/delay_manager.h
index 785fced..6f3c14a 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.h
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.h
@@ -13,10 +13,12 @@
#include <string.h> // Provide access to size_t.
+#include <memory>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -32,7 +34,9 @@
// buffer can hold no more than |max_packets_in_buffer| packets (i.e., this
// is the number of packet slots in the buffer). Supply a PeakDetector
// object to the DelayManager.
- DelayManager(size_t max_packets_in_buffer, DelayPeakDetector* peak_detector);
+ DelayManager(size_t max_packets_in_buffer,
+ DelayPeakDetector* peak_detector,
+ const TickTimer* tick_timer);
virtual ~DelayManager();
@@ -75,10 +79,6 @@
// DelayPeakDetector object.
virtual bool PeakFound() const;
- // Notifies the counters in DelayManager and DelayPeakDetector that
- // |elapsed_time_ms| have elapsed.
- virtual void UpdateCounters(int elapsed_time_ms);
-
// Reset the inter-arrival time counter to 0.
virtual void ResetPacketIatCount();
@@ -135,7 +135,9 @@
const size_t max_packets_in_buffer_; // Capacity of the packet buffer.
IATVector iat_vector_; // Histogram of inter-arrival times.
int iat_factor_; // Forgetting factor for updating the IAT histogram (Q15).
- int packet_iat_count_ms_; // Milliseconds elapsed since last packet.
+ const TickTimer* tick_timer_;
+ // Time elapsed since last packet.
+ std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
int base_target_level_; // Currently preferred buffer level before peak
// detection and streaming mode (Q0).
// TODO(turajs) change the comment according to the implementation of
@@ -153,7 +155,8 @@
int maximum_delay_ms_; // Externally set maximum allowed delay.
int iat_cumulative_sum_; // Cumulative sum of delta inter-arrival times.
int max_iat_cumulative_sum_; // Max of |iat_cumulative_sum_|.
- int max_timer_ms_; // Time elapsed since maximum was observed.
+ // Time elapsed since maximum was observed.
+ std::unique_ptr<TickTimer::Stopwatch> max_iat_stopwatch_;
DelayPeakDetector& peak_detector_;
int last_pack_cng_or_dtmf_;
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
index 05d6f3e..3290e9c 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -51,7 +51,7 @@
void DelayManagerTest::SetUp() {
EXPECT_CALL(detector_, Reset())
.Times(1);
- dm_ = new DelayManager(kMaxNumberOfPackets, &detector_);
+ dm_ = new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_);
}
void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {
@@ -67,7 +67,6 @@
void DelayManagerTest::IncreaseTime(int inc_ms) {
for (int t = 0; t < inc_ms; t += kTimeStepMs) {
- dm_->UpdateCounters(kTimeStepMs);
tick_timer_.Increment();
}
}
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
index 6fb8585..7ceea70 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -20,8 +20,9 @@
class MockDelayManager : public DelayManager {
public:
MockDelayManager(size_t max_packets_in_buffer,
- DelayPeakDetector* peak_detector)
- : DelayManager(max_packets_in_buffer, peak_detector) {}
+ DelayPeakDetector* peak_detector,
+ const TickTimer* tick_timer)
+ : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
virtual ~MockDelayManager() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_CONST_METHOD0(iat_vector,
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 7bdb23c..ef1e6cb 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -60,7 +60,8 @@
decoder_database(new DecoderDatabase),
delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
delay_manager(new DelayManager(config.max_packets_in_buffer,
- delay_peak_detector.get())),
+ delay_peak_detector.get(),
+ tick_timer.get())),
dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
dtmf_tone_generator(new DtmfToneGenerator),
packet_buffer(
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index ed6dc20..42f2c1e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -91,7 +91,7 @@
if (use_mock_delay_manager_) {
std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
- config_.max_packets_in_buffer, delay_peak_detector_));
+ config_.max_packets_in_buffer, delay_peak_detector_, tick_timer_));
mock_delay_manager_ = mock.get();
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
deps.delay_manager = std::move(mock);
