commit b0a020783890e526263ef9ecbbeb4048653559d0
author Gustaf Ullberg <gustaf@webrtc.org> Mon Oct 02 12:00:34 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Oct 02 10:47:00 2017 +0000
tree 546f03215501b9b6213bbfa0363a0a75432f0b3b
parent 652cc84069d3b2ff8d86405fe0b150cd1cb23264

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay

Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}

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 2d50225..36d6b27 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1950,7 +1950,8 @@
       assert(false);  // Should always be able to extract a packet here.
       return -1;
     }
-    stats_.StoreWaitingTime(packet->waiting_time->ElapsedMs());
+    const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
+    stats_.StoreWaitingTime(waiting_time_ms);
     RTC_DCHECK(!packet->empty());
 
     if (first_packet) {
@@ -1990,6 +1991,8 @@
     }
     extracted_samples = packet->timestamp - first_timestamp + packet_duration;
 
+    stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
+
     packet_list->push_back(std::move(*packet));  // Store packet in list.
     packet = rtc::Optional<Packet>();  // Ensure it's never used after the move.