Add LoudestFilter in ConferenceTransport

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  Copyright (c) 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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
 
 #include <deque>
 #include <map>
 #include <utility>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_file.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-
+#include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
 
 static const size_t kMaxPacketSizeByte = 1500;
 
 namespace voetest {
 
 // This class is to simulate a conference call. There are two Voice Engines, one
 // for local channels and the other for remote channels. There is a simulated
 // reflector, which exchanges RTCP with local channels. For simplicity, it
 // also uses the Voice Engine for remote channels. One can add streams by
 // calling AddStream(), which creates a remote sender channel and a local
 // receive channel. The remote sender channel plays a file as microphone in a
 // looped fashion. Received streams are mixed and played.
 
 class ConferenceTransport: public webrtc::Transport {
  public:
   ConferenceTransport();
   virtual ~ConferenceTransport();
 
   /* SetRtt()
    * Set RTT between local channels and reflector.
    *
    * Input:
    *   rtt_ms : RTT in milliseconds.
    */
   void SetRtt(unsigned int rtt_ms);
 
   /* AddStream()
    * Adds a stream in the conference.
    *
+   * Input:
+   *   file_name : name of the file to be added as microphone input.
+   *   format    : format of the input file.
+   *
    * Returns stream id.
    */
-  unsigned int AddStream();
+  unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
 
   /* RemoveStream()
    * Removes a stream with specified ID from the conference.
    *
    * Input:
    *   id : stream id.
    *
    * Returns false if the specified stream does not exist, true if succeeds.
    */
   bool RemoveStream(unsigned int id);
@@ skipping 43 matching lines @@
     uint32 send_time_ms_;
   };
 
   static bool Run(void* transport) {
     return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
   }
 
   int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
   void StorePacket(Packet::Type type, int channel, const void* data,
                    size_t len);
-  void SendPacket(const Packet& packet) const;
+  void SendPacket(const Packet& packet);
   bool DispatchPackets();
 
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> pq_crit_;
   const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> stream_crit_;
   const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
   const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
 
   unsigned int rtt_ms_;
   unsigned int stream_count_;
 
   std::map<unsigned int, std::pair<int, int>> streams_
     GUARDED_BY(stream_crit_.get());
   std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_.get());
 
   int local_sender_;  // Channel Id of local sender
   int reflector_;
 
   webrtc::VoiceEngine* local_voe_;
   webrtc::VoEBase* local_base_;
   webrtc::VoERTP_RTCP* local_rtp_rtcp_;
   webrtc::VoENetwork* local_network_;
 
   webrtc::VoiceEngine* remote_voe_;
   webrtc::VoEBase* remote_base_;
   webrtc::VoECodec* remote_codec_;
   webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
+
+  LoudestFilter loudest_filter_;
+
+  const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_