Populate qualityLimitationDurations stats for outbound RTP streams

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
Tested in chromium using wpt/webrtc-stats.

Bug: webrtc:10686
Change-Id: I05ac344e6caa7a663675de4c06ccfd17e1efb6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219300
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34179}
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index d92e7ff..032cbe9 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -973,6 +973,8 @@
       verifier.TestMemberIsNonNegative<double>(
           outbound_stream.total_packet_send_delay);
       verifier.TestMemberIsDefined(outbound_stream.quality_limitation_reason);
+      verifier.TestMemberIsDefined(
+          outbound_stream.quality_limitation_durations);
       verifier.TestMemberIsNonNegative<uint32_t>(
           outbound_stream.quality_limitation_resolution_changes);
       // The integration test is not set up to test screen share; don't require
@@ -1006,6 +1008,8 @@
       verifier.TestMemberIsUndefined(outbound_stream.total_packet_send_delay);
       verifier.TestMemberIsUndefined(outbound_stream.quality_limitation_reason);
       verifier.TestMemberIsUndefined(
+          outbound_stream.quality_limitation_durations);
+      verifier.TestMemberIsUndefined(
           outbound_stream.quality_limitation_resolution_changes);
       verifier.TestMemberIsUndefined(outbound_stream.content_type);
       // TODO(hbos): Implement for audio as well.