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}

call/audio_receive_stream.h

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 3f1a5ad..257042b 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -73,6 +73,10 @@
     int32_t decoding_plc_cng = 0;
     int32_t decoding_muted_output = 0;
     int64_t capture_start_ntp_time_ms = 0;
+    // 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;
     uint64_t jitter_buffer_flushes = 0;
     double relative_packet_arrival_delay_seconds = 0.0;
   };