Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics.

Bug: webrtc:10333
Change-Id: I415e2286b426cbca940fe3a187957531847272ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124780
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26976}
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index 7b57324..c57f074 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -18,6 +18,7 @@
 #include "modules/audio_coding/neteq/histogram.h"
 #include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
 #include "modules/audio_coding/neteq/mock/mock_histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
 #include "rtc_base/checks.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"
@@ -53,6 +54,7 @@
 
   std::unique_ptr<DelayManager> dm_;
   TickTimer tick_timer_;
+  MockStatisticsCalculator stats_;
   MockDelayPeakDetector detector_;
   MockHistogram* mock_histogram_;
   uint16_t seq_no_;
@@ -81,10 +83,11 @@
     dm_ = absl::make_unique<DelayManager>(
         kMaxNumberOfPackets, kMinDelayMs, kDefaultHistogramQuantile,
         histogram_mode_, enable_rtx_handling_, &detector_, &tick_timer_,
-        std::move(histogram));
+        &stats_, std::move(histogram));
   } else {
     dm_ = DelayManager::Create(kMaxNumberOfPackets, kMinDelayMs,
-                               enable_rtx_handling_, &detector_, &tick_timer_);
+                               enable_rtx_handling_, &detector_, &tick_timer_,
+                               &stats_);
   }
 }
 
@@ -709,4 +712,17 @@
   EXPECT_EQ(0, dm_->Update(seq_no_, ts_, kFs));
 }
 
+TEST_F(DelayManagerTest, RelativeArrivalDelayStatistic) {
+  SetPacketAudioLength(kFrameSizeMs);
+  InsertNextPacket();
+
+  IncreaseTime(kFrameSizeMs);
+  EXPECT_CALL(stats_, RelativePacketArrivalDelay(0));
+  InsertNextPacket();
+
+  IncreaseTime(2 * kFrameSizeMs);
+  EXPECT_CALL(stats_, RelativePacketArrivalDelay(20));
+  InsertNextPacket();
+}
+
 }  // namespace webrtc