Expose `jitterBufferEmittedCount` in addition to the existing `jitterBufferDelay` for `getStats()`.
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 2810294..ba36278 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -245,6 +245,8 @@
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+ acm_stat->jitterBufferEmittedCount =
+ neteq_lifetime_stat.jitter_buffer_emitted_count;
acm_stat->delayedPacketOutageSamples =
neteq_lifetime_stat.delayed_packet_outage_samples;
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index bafff72..0db423b 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -80,6 +80,7 @@
uint64_t concealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
+ uint64_t jitterBufferEmittedCount;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
uint16_t currentPacketLossRate;
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 6eecc65..d4e8423 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -70,6 +70,7 @@
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
uint64_t jitter_buffer_delay_ms = 0;
+ uint64_t jitter_buffer_emitted_count = 0;
// Below stat is not part of the spec.
uint64_t voice_concealed_samples = 0;
uint64_t delayed_packet_outage_samples = 0;
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index b562ede..b9b8d08 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1656,6 +1656,7 @@
int packets_sent = 0;
int packets_received = 0;
int expected_delay = 0;
+ uint64_t expected_emitted_count = 0;
while (packets_received < kNumPackets) {
// Insert packet.
if (packets_sent < kNumPackets) {
@@ -1679,6 +1680,7 @@
// number of samples that are sent for play out.
int current_delay_ms = packets_delay * kPacketLenMs;
expected_delay += current_delay_ms * kSamples;
+ expected_emitted_count += kSamples;
}
}
@@ -1690,6 +1692,7 @@
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
+ EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 5d94abd..7ad1a28 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -246,6 +246,7 @@
void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
uint64_t waiting_time_ms) {
lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
+ lifetime_stats_.jitter_buffer_emitted_count += num_samples;
}
void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {