commit 28a2c63526f471558bc93cdcae9fab42b84b10a5
author Minyue Li <minyue@webrtc.org> Wed Jul 07 15:53:38 2021 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Tue Jul 13 20:34:45 2021 +0000
tree 5760ce327547b8578473e6426e6ba23d0dbc0af4
parent 6a9dec9392e2854787dc936430915e8b2d1c9b74

Adding packetsDiscarded to RTCReceivedRtpStreamStats.

Bug: webrtc:12532, webrtc:7065, webrtc:8199
Change-Id: I3ba62ec65e5660e98787f629aec3ee7a0889207a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225261
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34468}

modules/audio_coding/acm2/acm_receiver.cc

diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 3214ce6..80cb3c5 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -307,6 +307,8 @@
       neteq_->GetOperationsAndState();
   acm_stat->packetBufferFlushes =
       neteq_operations_and_state.packet_buffer_flushes;
+  acm_stat->packetsDiscarded =
+      neteq_operations_and_state.discarded_primary_packets;
 }
 
 int AcmReceiver::EnableNack(size_t max_nack_list_size) {

modules/audio_coding/include/audio_coding_module_typedefs.h

diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index a7210da..e5598e3 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -81,19 +81,22 @@
   // adding extra delay due to "peaky jitter"
   bool jitterPeaksFound;
   // Stats below correspond to similarly-named fields in the WebRTC stats spec.
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
   uint64_t totalSamplesReceived;
   uint64_t concealedSamples;
   uint64_t silentConcealedSamples;
   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;
   uint64_t fecPacketsDiscarded;
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
+  uint64_t packetsDiscarded;
+  // Non standard stats propagated to spec complaint GetStats API.
+  uint64_t jitterBufferTargetDelayMs;
   // Stats below DO NOT correspond directly to anything in the WebRTC stats
   // fraction (of original stream) of synthesized audio inserted through
   // expansion (in Q14)