commit e618cc9c1ed030f33dd54396d04141b66dd24f97
author Artem Titov <titovartem@webrtc.org> Wed Mar 11 11:18:54 2020 +0100
committer Commit Bot <commit-bot@chromium.org> Wed Mar 11 12:08:32 2020 +0000
tree 712be5c4e5d62dd2360680316aa7ef1f7d4efed4
parent c46385c346d5a98b83ecef16bd4b4aa219a61d71

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}

modules/audio_coding/neteq/neteq_impl.cc

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.