Video jitter stats fix: Convert RTP timestamp

stats.rtp_stats.jitter is a RTP timestamp so we needed to convert it back to regular timestamps

See https://bugs.chromium.org/p/webrtc/issues/detail?id=12980#c7

Bug: webrtc:12980
Change-Id: I0d94a22e043ac6ecec4926d950abbdcf787b7168
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227100
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Di Wu <meetwudi@gmail.com>
Cr-Commit-Position: refs/heads/master@{#34590}
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index d42047a..8376b01 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -3161,7 +3161,7 @@
       stats.rtp_stats.packet_counter.padding_bytes;
   info.packets_rcvd = stats.rtp_stats.packet_counter.packets;
   info.packets_lost = stats.rtp_stats.packets_lost;
-  info.jitter_ms = stats.rtp_stats.jitter;
+  info.jitter_ms = stats.rtp_stats.jitter / (kVideoCodecClockrate / 1000);
 
   info.framerate_rcvd = stats.network_frame_rate;
   info.framerate_decoded = stats.decode_frame_rate;