Wire up non-sender RTT for audio, and implement related standardized stats.

The implemented stats are:
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
- https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements

Bug: webrtc:12951, webrtc:12714
Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34861}

modules/rtp_rtcp/source/rtp_rtcp_impl2.h

@@ -213,6 +213,7 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
   // which is the SSRC of the corresponding outbound RTP stream, is unique.
   std::vector<ReportBlockData> GetLatestReportBlockData() const override;
   absl::optional<SenderReportStats> GetSenderReportStats() const override;
+  absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override;
 
   // (REMB) Receiver Estimated Max Bitrate.
   void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override;