Populate jitter stats for video RTP streams

Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks!

Bug: webrtc:12487
Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33325}

pc/rtc_stats_integrationtest.cc

diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index e2d7c35..6288f60 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -836,7 +836,7 @@
     verifier.TestMemberIsUndefined(inbound_stream.frame_bit_depth);
     if (inbound_stream.media_type.is_defined() &&
         *inbound_stream.media_type == "video") {
-      verifier.TestMemberIsUndefined(inbound_stream.jitter);
+      verifier.TestMemberIsNonNegative<double>(inbound_stream.jitter);
       verifier.TestMemberIsUndefined(inbound_stream.jitter_buffer_delay);
       verifier.TestMemberIsUndefined(
           inbound_stream.jitter_buffer_emitted_count);