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

webrtc/pc/rtpsender.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.
  */
 
 // This file contains classes that implement RtpSenderInterface.
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
 #ifndef WEBRTC_PC_RTPSENDER_H_
 #define WEBRTC_PC_RTPSENDER_H_
 
 #include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/pc/channel.h"
+#include "webrtc/pc/dtmfsender.h"
 #include "webrtc/pc/statscollector.h"
 
 namespace webrtc {
 
 // Internal interface used by PeerConnection.
 class RtpSenderInternal : public RtpSenderInterface {
  public:
   // Used to set the SSRC of the sender, once a local description has been set.
   // If |ssrc| is 0, this indiates that the sender should disconnect from the
   // underlying transport (this occurs if the sender isn't seen in a local
@@ skipping 24 matching lines @@
               size_t number_of_frames) override;
 
   // cricket::AudioSource implementation.
   void SetSink(cricket::AudioSource::Sink* sink) override;
 
   cricket::AudioSource::Sink* sink_;
   // Critical section protecting |sink_|.
   rtc::CriticalSection lock_;
 };
 
-class AudioRtpSender : public ObserverInterface,
+class AudioRtpSender : public DtmfProviderInterface,
+                       public ObserverInterface,
                        public rtc::RefCountedObject<RtpSenderInternal> {
  public:
   // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
   // at the appropriate times.
   // |channel| can be null if one does not exist yet.
   AudioRtpSender(AudioTrackInterface* track,
                  const std::string& stream_id,
                  cricket::VoiceChannel* channel,
                  StatsCollector* stats);
 
   // Randomly generates stream_id.
   // |channel| can be null if one does not exist yet.
   AudioRtpSender(AudioTrackInterface* track,
                  cricket::VoiceChannel* channel,
                  StatsCollector* stats);
 
   // Randomly generates id and stream_id.
   // |channel| can be null if one does not exist yet.
   AudioRtpSender(cricket::VoiceChannel* channel, StatsCollector* stats);
 
   virtual ~AudioRtpSender();
 
-  // ObserverInterface implementation
+  // DtmfSenderProvider implementation.
+  bool CanInsertDtmf() override;
+  bool InsertDtmf(int code, int duration) override;
+  sigslot::signal0<>* GetOnDestroyedSignal() override;
+
+  // ObserverInterface implementation.
   void OnChanged() override;
 
-  // RtpSenderInterface implementation
+  // RtpSenderInterface implementation.
   bool SetTrack(MediaStreamTrackInterface* track) override;
   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
     return track_;
   }
 
   uint32_t ssrc() const override { return ssrc_; }
 
   cricket::MediaType media_type() const override {
     return cricket::MEDIA_TYPE_AUDIO;
   }
 
   std::string id() const override { return id_; }
 
   std::vector<std::string> stream_ids() const override {
     std::vector<std::string> ret = {stream_id_};
     return ret;
   }
 
   RtpParameters GetParameters() const override;
   bool SetParameters(const RtpParameters& parameters) override;
 
+  rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
+
   // RtpSenderInternal implementation.
   void SetSsrc(uint32_t ssrc) override;
 
   void set_stream_id(const std::string& stream_id) override {
     stream_id_ = stream_id;
   }
   std::string stream_id() const override { return stream_id_; }
 
   void Stop() override;
 
   // Does not take ownership.
   // Should call SetChannel(nullptr) before |channel| is destroyed.
   void SetChannel(cricket::VoiceChannel* channel) { channel_ = channel; }
 
  private:
   // TODO(nisse): Since SSRC == 0 is technically valid, figure out
   // some other way to test if we have a valid SSRC.
   bool can_send_track() const { return track_ && ssrc_; }
   // Helper function to construct options for
   // AudioProviderInterface::SetAudioSend.
   void SetAudioSend();
   // Helper function to call SetAudioSend with "stop sending" parameters.
   void ClearAudioSend();
 
+  void CreateDtmfSender();
+
+  sigslot::signal0<> SignalDestroyed;
+
   std::string id_;
   std::string stream_id_;
   cricket::VoiceChannel* channel_ = nullptr;
   StatsCollector* stats_;
   rtc::scoped_refptr<AudioTrackInterface> track_;
+  rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   bool stopped_ = false;
 
   // Used to pass the data callback from the |track_| to the other end of
   // cricket::AudioSource.
   std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
 };
 
 class VideoRtpSender : public ObserverInterface,
@@ skipping 32 matching lines @@
   std::string id() const override { return id_; }
 
   std::vector<std::string> stream_ids() const override {
     std::vector<std::string> ret = {stream_id_};
     return ret;
   }
 
   RtpParameters GetParameters() const override;
   bool SetParameters(const RtpParameters& parameters) override;
 
+  rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
+
   // RtpSenderInternal implementation.
   void SetSsrc(uint32_t ssrc) override;
 
   void set_stream_id(const std::string& stream_id) override {
     stream_id_ = stream_id;
   }
   std::string stream_id() const override { return stream_id_; }
 
   void Stop() override;
 
@@ skipping 16 matching lines @@
   uint32_t ssrc_ = 0;
   bool cached_track_enabled_ = false;
   VideoTrackInterface::ContentHint cached_track_content_hint_ =
       VideoTrackInterface::ContentHint::kNone;
   bool stopped_ = false;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_RTPSENDER_H_