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 145 matching lines @@
     }
     LOG(LS_WARNING) << "Receiving invalid payload type.";
     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 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;
 
   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;
@@ skipping 316 matching lines @@
     // Using CSRC 0 to signal this event, not interop safe, other
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
-void RtpReceiverImpl::UpdateSources() {
+void RtpReceiverImpl::UpdateSources(
+    const rtc::Optional<uint8_t>& ssrc_audio_level) {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   int64_t now_ms = clock_->TimeInMilliseconds();
 
   for (size_t i = 0; i < num_csrcs_; ++i) {
     auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
     if (map_it == iterator_by_csrc_.end()) {
       // If it is a new CSRC, append a new object to the end of the list.
       csrc_sources_.emplace_back(now_ms, current_remote_csrc_[i],
                                  RtpSourceType::CSRC);
     } else {
       // If it is an existing CSRC, move the object to the end of the list.
       map_it->second->update_timestamp_ms(now_ms);
       csrc_sources_.splice(csrc_sources_.end(), csrc_sources_, map_it->second);
     }
     // Update the unordered_map.
     iterator_by_csrc_[current_remote_csrc_[i]] = std::prev(csrc_sources_.end());
   }
 
   // If this is the first packet or the SSRC is changed, insert a new
   // contributing source that uses the SSRC.
   if (ssrc_sources_.empty() || ssrc_sources_.rbegin()->source_id() != ssrc_) {
     ssrc_sources_.emplace_back(now_ms, ssrc_, RtpSourceType::SSRC);
   } else {
     ssrc_sources_.rbegin()->update_timestamp_ms(now_ms);
   }
 
+  ssrc_sources_.back().set_audio_level(ssrc_audio_level);
+
   RemoveOutdatedSources(now_ms);
 }
 
 void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {
   std::list<RtpSource>::iterator it;
   for (it = csrc_sources_.begin(); it != csrc_sources_.end(); ++it) {
     if ((now_ms - it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
     iterator_by_csrc_.erase(it->source_id());
   }
   csrc_sources_.erase(csrc_sources_.begin(), it);
 
   std::vector<RtpSource>::iterator vec_it;
   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