Move DTMF sender to RtpSender (as opposed to WebRtcSession).

webrtc/pc/webrtcsession.h

 /*
  *  Copyright 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.
  */
 
 #ifndef WEBRTC_PC_WEBRTCSESSION_H_
 #define WEBRTC_PC_WEBRTCSESSION_H_
 
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/sslidentity.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/p2p/base/candidate.h"
 #include "webrtc/p2p/base/transportcontroller.h"
 #include "webrtc/pc/datachannel.h"
-#include "webrtc/pc/dtmfsender.h"
 #include "webrtc/pc/mediacontroller.h"
 #include "webrtc/pc/mediasession.h"
 
 #ifdef HAVE_QUIC
 #include "webrtc/pc/quicdatatransport.h"
 #endif  // HAVE_QUIC
 
 namespace cricket {
 
 class ChannelManager;
@@ skipping 92 matching lines @@
 };
 
 // A WebRtcSession manages general session state. This includes negotiation
 // of both the application-level and network-level protocols:  the former
 // defines what will be sent and the latter defines how it will be sent.  Each
 // network-level protocol is represented by a Transport object.  Each Transport
 // participates in the network-level negotiation.  The individual streams of
 // packets are represented by TransportChannels.  The application-level protocol
 // is represented by SessionDecription objects.
 class WebRtcSession :
-
-    public DtmfProviderInterface,
     public DataChannelProviderInterface,
     public sigslot::has_slots<> {
  public:
   enum State {
     STATE_INIT = 0,
     STATE_SENTOFFER,         // Sent offer, waiting for answer.
     STATE_RECEIVEDOFFER,     // Received an offer. Need to send answer.
     STATE_SENTPRANSWER,      // Sent provisional answer. Need to send answer.
     STATE_RECEIVEDPRANSWER,  // Received provisional answer, waiting for answer.
     STATE_INPROGRESS,        // Offer/answer exchange completed.
@@ skipping 123 matching lines @@
     return pending_local_description_.get();
   }
   const SessionDescriptionInterface* pending_remote_description() const {
     return pending_remote_description_.get();
   }
 
   // Get the id used as a media stream track's "id" field from ssrc.
   virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
   virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
 
-  // Implements DtmfProviderInterface.
-  bool CanInsertDtmf(const std::string& track_id) override;
-  bool InsertDtmf(const std::string& track_id,
-                  int code, int duration) override;
-  sigslot::signal0<>* GetOnDestroyedSignal() override;
-
   // Implements DataChannelProviderInterface.
   bool SendData(const cricket::SendDataParams& params,
                 const rtc::CopyOnWriteBuffer& payload,
                 cricket::SendDataResult* result) override;
   bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
   void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
   void AddSctpDataStream(int sid) override;
   void RemoveSctpDataStream(int sid) override;
   bool ReadyToSendData() const override;
 
@@ skipping 50 matching lines @@
 
   // Called when voice_channel_, video_channel_ and
   // rtp_data_channel_/sctp_transport_ are created and destroyed. As a result
   // of, for example, setting a new description.
   sigslot::signal0<> SignalVoiceChannelCreated;
   sigslot::signal0<> SignalVoiceChannelDestroyed;
   sigslot::signal0<> SignalVideoChannelCreated;
   sigslot::signal0<> SignalVideoChannelDestroyed;
   sigslot::signal0<> SignalDataChannelCreated;
   sigslot::signal0<> SignalDataChannelDestroyed;
-  // Called when the whole session is destroyed.
-  sigslot::signal0<> SignalDestroyed;
 
   // Called when a valid data channel OPEN message is received.
   // std::string represents the data channel label.
   sigslot::signal2<const std::string&, const InternalDataChannelInit&>
       SignalDataChannelOpenMessage;
 #ifdef HAVE_QUIC
   QuicDataTransport* quic_data_transport() {
     return quic_data_transport_.get();
   }
 #endif  // HAVE_QUIC
@@ skipping 269 matching lines @@
 
 #ifdef HAVE_QUIC
   std::unique_ptr<QuicDataTransport> quic_data_transport_;
 #endif  // HAVE_QUIC
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_WEBRTCSESSION_H_