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

webrtc/api/rtpreceiverinterface.h

 /*
  *  Copyright 2015 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 21 matching lines @@
 };
 
 class RtpSource {
  public:
   RtpSource() = delete;
   RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type)
       : timestamp_ms_(timestamp_ms),
         source_id_(source_id),
         source_type_(source_type) {}
 
+  RtpSource(int64_t timestamp_ms,
+            uint32_t source_id,
+            RtpSourceType source_type,
+            uint8_t audio_level)
+      : timestamp_ms_(timestamp_ms),
+        source_id_(source_id),
+        source_type_(source_type),
+        audio_level_(audio_level) {}
+
   int64_t timestamp_ms() const { return timestamp_ms_; }
   void update_timestamp_ms(int64_t timestamp_ms) {
     RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
     timestamp_ms_ = timestamp_ms;
   }
 
   // The identifier of the source can be the CSRC or the SSRC.
   uint32_t source_id() const { return source_id_; }
 
   // The source can be either a contributing source or a synchronization source.
   RtpSourceType source_type() const { return source_type_; }
 
-  // This isn't implemented yet and will always return an empty Optional.
-  // TODO(zhihuang): Implement this to return real audio level.
-  rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
+  rtc::Optional<uint8_t> audio_level() const { return audio_level_; }
+  void set_audio_level(const rtc::Optional<uint8_t>& level) {
+    audio_level_ = level;
+  }
 
   bool operator==(const RtpSource& o) const {
     return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
-           source_type_ == o.source_type();
+           source_type_ == o.source_type() && audio_level_ == o.audio_level_;
   }
 
  private:
   int64_t timestamp_ms_;
   uint32_t source_id_;
   RtpSourceType source_type_;
+  rtc::Optional<uint8_t> audio_level_;
 };
 
 class RtpReceiverObserverInterface {
  public:
   // Note: Currently if there are multiple RtpReceivers of the same media type,
   // they will all call OnFirstPacketReceived at once.
   //
   // In the future, it's likely that an RtpReceiver will only call
   // OnFirstPacketReceived when a packet is received specifically for its
   // SSRC/mid.
@@ skipping 46 matching lines @@
   PROXY_CONSTMETHOD0(std::string, id)
   PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
   PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
   PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
   PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
   END_PROXY_MAP()
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTPRECEIVERINTERFACE_H_