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}
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index b64803b..c21f89e 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -435,6 +435,10 @@
   int packets_rcvd = 0;
   int packets_lost = 0;
   float fraction_lost = 0.0f;
+  // The timestamp at which the last packet was received, i.e. the time of the
+  // local clock when it was received - not the RTP timestamp of that packet.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
+  absl::optional<int64_t> last_packet_received_timestamp_ms;
   std::string codec_name;
   absl::optional<int> codec_payload_type;
   std::vector<SsrcReceiverInfo> local_stats;