Reland "Improve outbound-rtp statistics for simulcast"

This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.

Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".

Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Eldar Rello <elrello@microsoft.com>
> > Cr-Commit-Position: refs/heads/master@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}

TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com

# Not skipping CQ checks because this is a reland.

Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}

2 files changed, 8 insertions, 0 deletions

diff --git a/api/peer_connection_interface.h b/api/peer_connection_interface.h
index 0ae47b2a2f..1d81de74d8 100644
--- a/api/peer_connection_interface.h
+++ b/api/peer_connection_interface.h
@@ -666,6 +666,8 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // Whether network condition based codec switching is allowed.
     absl::optional<bool> allow_codec_switching;
 
+    bool enable_simulcast_stats = true;
+
     //
     // Don't forget to update operator== if adding something.
     //
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index e83c83d97e..28d841db09 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -469,6 +469,7 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
 
   RTCStatsMember<std::string> media_source_id;
   RTCStatsMember<std::string> remote_id;
+  RTCStatsMember<std::string> rid;
   RTCStatsMember<uint32_t> packets_sent;
   RTCStatsMember<uint64_t> retransmitted_packets_sent;
   RTCStatsMember<uint64_t> bytes_sent;
@@ -480,6 +481,11 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
   RTCStatsMember<uint32_t> key_frames_encoded;
   RTCStatsMember<double> total_encode_time;
   RTCStatsMember<uint64_t> total_encoded_bytes_target;
+  RTCStatsMember<uint32_t> frame_width;
+  RTCStatsMember<uint32_t> frame_height;
+  RTCStatsMember<double> frames_per_second;
+  RTCStatsMember<uint32_t> frames_sent;
+  RTCStatsMember<uint32_t> huge_frames_sent;
   // TODO(https://crbug.com/webrtc/10635): This is only implemented for video;
   // implement it for audio as well.
   RTCStatsMember<double> total_packet_send_delay;