Move talk/app/webrtc to webrtc/api

talk/app/webrtc/dtmfsender.h

-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_APP_WEBRTC_DTMFSENDER_H_
-#define TALK_APP_WEBRTC_DTMFSENDER_H_
-
-#include <string>
-
-#include "talk/app/webrtc/dtmfsenderinterface.h"
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/proxy.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/messagehandler.h"
-#include "webrtc/base/refcount.h"
-
-// DtmfSender is the native implementation of the RTCDTMFSender defined by
-// the WebRTC W3C Editor's Draft.
-// http://dev.w3.org/2011/webrtc/editor/webrtc.html
-
-namespace rtc {
-class Thread;
-}
-
-namespace webrtc {
-
-// This interface is called by DtmfSender to talk to the actual audio channel
-// to send DTMF.
-class DtmfProviderInterface {
- public:
-  // Returns true if the audio track with given id (|track_id|) is capable
-  // of sending DTMF. Otherwise returns false.
-  virtual bool CanInsertDtmf(const std::string& track_id) = 0;
-  // Sends DTMF |code| via the audio track with given id (|track_id|).
-  // The |duration| indicates the length of the DTMF tone in ms.
-  // Returns true on success and false on failure.
-  virtual bool InsertDtmf(const std::string& track_id,
-                          int code, int duration) = 0;
-  // Returns a |sigslot::signal0<>| signal. The signal should fire before
-  // the provider is destroyed.
-  virtual sigslot::signal0<>* GetOnDestroyedSignal() = 0;
-
- protected:
-  virtual ~DtmfProviderInterface() {}
-};
-
-class DtmfSender
-    : public DtmfSenderInterface,
-      public sigslot::has_slots<>,
-      public rtc::MessageHandler {
- public:
-  static rtc::scoped_refptr<DtmfSender> Create(
-      AudioTrackInterface* track,
-      rtc::Thread* signaling_thread,
-      DtmfProviderInterface* provider);
-
-  // Implements DtmfSenderInterface.
-  void RegisterObserver(DtmfSenderObserverInterface* observer) override;
-  void UnregisterObserver() override;
-  bool CanInsertDtmf() override;
-  bool InsertDtmf(const std::string& tones,
-                  int duration,
-                  int inter_tone_gap) override;
-  const AudioTrackInterface* track() const override;
-  std::string tones() const override;
-  int duration() const override;
-  int inter_tone_gap() const override;
-
- protected:
-  DtmfSender(AudioTrackInterface* track,
-             rtc::Thread* signaling_thread,
-             DtmfProviderInterface* provider);
-  virtual ~DtmfSender();
-
- private:
-  DtmfSender();
-
-  // Implements MessageHandler.
-  virtual void OnMessage(rtc::Message* msg);
-
-  // The DTMF sending task.
-  void DoInsertDtmf();
-
-  void OnProviderDestroyed();
-
-  void StopSending();
-
-  rtc::scoped_refptr<AudioTrackInterface> track_;
-  DtmfSenderObserverInterface* observer_;
-  rtc::Thread* signaling_thread_;
-  DtmfProviderInterface* provider_;
-  std::string tones_;
-  int duration_;
-  int inter_tone_gap_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(DtmfSender);
-};
-
-// Define proxy for DtmfSenderInterface.
-BEGIN_PROXY_MAP(DtmfSender)
-  PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*)
-  PROXY_METHOD0(void, UnregisterObserver)
-  PROXY_METHOD0(bool, CanInsertDtmf)
-  PROXY_METHOD3(bool, InsertDtmf, const std::string&, int, int)
-  PROXY_CONSTMETHOD0(const AudioTrackInterface*, track)
-  PROXY_CONSTMETHOD0(std::string, tones)
-  PROXY_CONSTMETHOD0(int, duration)
-  PROXY_CONSTMETHOD0(int, inter_tone_gap)
-END_PROXY()
-
-// Get DTMF code from the DTMF event character.
-bool GetDtmfCode(char tone, int* code);
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_DTMFSENDER_H_