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

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 147 matching lines @@
     return false;
   }
 
   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
   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;
   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);
     if (HaveReceivedFrame()) {
       is_first_packet_in_frame =
           last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
           last_received_timestamp_ != rtp_header.timestamp;
     } else {
@@ skipping 366 matching lines @@
   for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
        ++vec_it) {
     if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
   }
   ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
 }
 
 }  // namespace webrtc