commit efbe75361705198cab3f4eb3c12368fd3b7930ef
author Fredrik Hernqvist <fhernqvist@google.com> Fri Jan 13 15:42:36 2023
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Mon Jan 16 13:19:45 2023
tree 73e31cbad64ac1dc071d131d56ac26bf9ea81b4f
parent f7e40717ab3c909f8b98251923dd54f465d59da7

Add RTCAudioPlayoutStats to GetStats().

This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.

Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}

pc/peer_connection.cc

diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 622870c..5633be3 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -22,6 +22,7 @@
 #include "absl/algorithm/container.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
 #include "api/jsep_ice_candidate.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -2500,6 +2501,13 @@
   }
 }
 
+absl::optional<AudioDeviceModule::Stats> PeerConnection::GetAudioDeviceStats() {
+  if (context_->media_engine()) {
+    return context_->media_engine()->voice().GetAudioDeviceStats();
+  }
+  return absl::nullopt;
+}
+
 bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
   DataChannelTransportInterface* transport =
       transport_controller_->GetDataChannelTransport(mid);