Only return stats for the most recent unsignaled audio stream.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index d202106be09025ed3215525b70eae77c055e2388..fc2213c6525b0b6b9b558d3a6ee153c7742a7f66 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -2267,6 +2267,23 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
   // Get SSRC and stats for each receiver.
   RTC_DCHECK_EQ(info->receivers.size(), 0U);
   for (const auto& stream : recv_streams_) {
+    uint32_t ssrc = stream.first;
+    // When SSRCs are unsignaled, there's only one audio MediaStreamTrack, but
+    // multiple RTP streams can be received over time (if the SSRC changes for
+    // whatever reason). We only want the RTCMediaStreamTrackStats to represent
+    // the stats for the most recent stream (the one whose audio is actually
+    // routed to the MediaStreamTrack), so here we ignore any unsignaled SSRCs
+    // except for the most recent one (last in the vector). This is somewhat of
+    // a hack, and means you don't get *any* stats for these inactive streams,
+    // but it's slightly better than the previous behavior, which was "highest
+    // SSRC wins".
+    // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
+    if (!unsignaled_recv_ssrcs_.empty() &&
+        std::find(unsignaled_recv_ssrcs_.begin(),
+                  --unsignaled_recv_ssrcs_.end(),

[the sun, 2017/09/18 18:59:35]

nit: I would've done it

auto end_it = --unsignaled_recv_ssrcs_.end();
if (std::find(unsignaled_recv_ssrcs.begin(), endit, ssrc) != end_it) {
  continue;
}

which is as many lines but easier to read

[Taylor Brandstetter, 2017/09/20 00:19:45]

Can't quite do that since if the list is empty end() == begin() and decrementing
it is technically undefined behavior. But I'll do something similar.

+                  ssrc) != --unsignaled_recv_ssrcs_.end()) {
+      continue;
+    }
     webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
     VoiceReceiverInfo rinfo;
     rinfo.add_ssrc(stats.remote_ssrc);