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..f0c5c99bda93606d3802527dae11c4fb4d006dbb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -165,6 +165,12 @@ bool RtpReceiverImpl::IncomingRtpPacket(
 
   UpdateSources();
 
+  auto source = ssrc_sources_.rbegin();

[danilchap, 2017/08/15 08:18:29]

though it is not annotated, it seems ssrc_sources_ is guarded by
critical_section_rtp_receiver_
it might be better to set audio_level inside UpdateSources function

[Zach Stein, 2017/08/15 21:44:59]

Done.

+  source->update_audio_level(

[danilchap, 2017/08/15 08:18:29]

minor suggestion to use back() instead of rbegin():
ssrc_sources_.back().set_audio_level(...
(feel free to ignore if you prefer rbegin)

[Zach Stein, 2017/08/15 21:44:59]

Done.

+      rtp_header.extension.hasAudioLevel
+          ? rtc::Optional<uint8_t>(rtp_header.extension.audioLevel)
+          : rtc::Optional<uint8_t>());
+
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
   bool is_first_packet_in_frame = false;