Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
Bug: webrtc:11381
Change-Id: I7df3450e50da49d178e1e3a5d9f4970672d91aac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30758}
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 9cecb98..29eff19 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -272,6 +272,8 @@
neteq_lifetime_stat.silent_concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+ acm_stat->jitterBufferTargetDelayMs =
+ neteq_lifetime_stat.jitter_buffer_target_delay_ms;
acm_stat->jitterBufferEmittedCount =
neteq_lifetime_stat.jitter_buffer_emitted_count;
acm_stat->delayedPacketOutageSamples =
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 2f40acd..a5d4b24 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -90,6 +90,8 @@
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
+ // Non standard stats propagated to spec complaint GetStats API.
+ uint64_t jitterBufferTargetDelayMs;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 265043a..0b7510d 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1987,7 +1987,9 @@
}
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
- stats_->JitterBufferDelay(packet_duration, waiting_time_ms);
+ RTC_DCHECK(controller_);
+ stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
+ controller_->TargetLevelMs());
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 701a3c5..d78e2c6 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -986,6 +986,7 @@
int packets_sent = 0;
int packets_received = 0;
int expected_delay = 0;
+ int expected_target_delay = 0;
uint64_t expected_emitted_count = 0;
while (packets_received < kNumPackets) {
// Insert packet.
@@ -1010,6 +1011,7 @@
// number of samples that are sent for play out.
int current_delay_ms = packets_delay * kPacketLenMs;
expected_delay += current_delay_ms * kSamples;
+ expected_target_delay += neteq_->TargetDelayMs() * kSamples;
expected_emitted_count += kSamples;
}
}
@@ -1021,8 +1023,11 @@
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
- EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
+ EXPECT_EQ(expected_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_delay_ms));
EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
+ EXPECT_EQ(expected_target_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
@@ -1043,6 +1048,7 @@
rtp_info.markerBit = 0;
const uint8_t payload[kPayloadBytes] = {0};
+ int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
neteq_->InsertPacket(rtp_info, payload);
bool muted;
@@ -1055,6 +1061,7 @@
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload);
+ expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
// We have two packets in the buffer and kAccelerate operation will
// extract 20 ms of data.
neteq_->GetAudio(&out_frame_, &muted, NetEq::Operation::kAccelerate);
@@ -1063,6 +1070,8 @@
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(10 * kSamples * 3, stats.jitter_buffer_delay_ms);
EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count);
+ EXPECT_EQ(expected_target_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
namespace test {
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 081ec33..fa2925c 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -275,8 +275,11 @@
}
void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
- uint64_t waiting_time_ms) {
+ uint64_t waiting_time_ms,
+ uint64_t target_delay_ms) {
lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
+ lifetime_stats_.jitter_buffer_target_delay_ms +=
+ target_delay_ms * num_samples;
lifetime_stats_.jitter_buffer_emitted_count += num_samples;
}
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 38e463c..333f4a7 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -83,7 +83,9 @@
void IncreaseCounter(size_t num_samples, int fs_hz);
// Update jitter buffer delay counter.
- void JitterBufferDelay(size_t num_samples, uint64_t waiting_time_ms);
+ void JitterBufferDelay(size_t num_samples,
+ uint64_t waiting_time_ms,
+ uint64_t target_delay_ms);
// Stores new packet waiting time in waiting time statistics.
void StoreWaitingTime(int waiting_time_ms);