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

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

 /*
  *  Copyright (c) 2013 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 61 matching lines @@
  private:
   bool HaveReceivedFrame() const;
 
   void CheckSSRCChanged(const RTPHeader& rtp_header);
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
                               bool* is_red,
                               PayloadUnion* payload);
 
-  void UpdateSources();
+  void UpdateSources(const rtc::Optional<uint8_t>& ssrc_audio_level);
   void RemoveOutdatedSources(int64_t now_ms);
 
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
 
   RtpFeedback* cb_rtp_feedback_;
 
   rtc::CriticalSection critical_section_rtp_receiver_;
   int64_t last_receive_time_;
   size_t last_received_payload_length_;
 
   // SSRCs.
   uint32_t ssrc_;
   uint8_t num_csrcs_;
   uint32_t current_remote_csrc_[kRtpCsrcSize];
 
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
 
   std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
       iterator_by_csrc_;
   // The RtpSource objects are sorted chronologically.
   std::list<RtpSource> csrc_sources_;
   std::vector<RtpSource> ssrc_sources_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_