Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.

This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}

audio/audio_receive_stream.cc

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index e13e880..b4948ee 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -190,6 +190,8 @@
   stats.packets_lost = call_stats.cumulativeLost;
   stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
   stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
+  stats.last_packet_received_timestamp_ms =
+      call_stats.last_packet_received_timestamp_ms;
   stats.codec_name = receive_codec->second.name;
   stats.codec_payload_type = receive_codec->first;
   stats.ext_seqnum = call_stats.extendedMax;