Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 20f57460a68ffc1cb30d84b43e46dd2361400c7f..5693fd1d69cd086787b8b62c2a6b4c9bbfd01b4d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -163,7 +163,11 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
-  UpdateSources();
+  auto audio_level =
+      rtp_header.extension.hasAudioLevel
+          ? rtc::Optional<uint8_t>(rtp_header.extension.audioLevel)
+          : rtc::Optional<uint8_t>();
+  UpdateSources(audio_level);
 
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
@@ -500,7 +504,7 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   }
 }
 
-void RtpReceiverImpl::UpdateSources() {
+void RtpReceiverImpl::UpdateSources(rtc::Optional<uint8_t> audio_level) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   int64_t now_ms = clock_->TimeInMilliseconds();
 
@@ -528,6 +532,8 @@ void RtpReceiverImpl::UpdateSources() {
   }
 
   RemoveOutdatedSources(now_ms);
+
+  ssrc_sources_.back().set_audio_level(audio_level);

[danilchap, 2017/08/16 10:31:40]

this line probably looks better before RemoveOutdatedSources.
then it is obvious why last element is updated and why it is safe to update. (as
it is now need to ensure RemoveOutdatedSources do not remove last element or
reorder them.)

[Zach Stein, 2017/08/21 17:55:06]

Done.

 }
 
 void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {