commit 8af889659637e347eb3dc8b80295160cd323b676
author Ruslan Burakov <kuddai@google.com> Thu Nov 22 16:21:10 2018
committer Commit Bot <commit-bot@chromium.org> Fri Nov 23 11:41:43 2018
tree 19855b01dfbe81f31001b2bfcddaf32d01d176eb
parent b357e54dd5b2730ebc3d26e67b721e1aa57e3a5f

Expose jitter buffer flushes metric in new getStats api.

Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}

pc/rtcstatscollector_unittest.cc

diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index 3fc0127..e16c7e3 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1426,6 +1426,7 @@
   voice_receiver_info.concealed_samples = 123;
   voice_receiver_info.concealment_events = 12;
   voice_receiver_info.jitter_buffer_delay_seconds = 3456;
+  voice_receiver_info.jitter_buffer_flushes = 7;
 
   stats_->CreateMockRtpSendersReceiversAndChannels(
       {}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
@@ -1459,6 +1460,7 @@
   expected_remote_audio_track.concealed_samples = 123;
   expected_remote_audio_track.concealment_events = 12;
   expected_remote_audio_track.jitter_buffer_delay = 3456;
+  expected_remote_audio_track.jitter_buffer_flushes = 7;
   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())