Move talk/app/webrtc to webrtc/api

talk/app/webrtc/dtmfsenderinterface.h

Index: talk/app/webrtc/dtmfsenderinterface.h
diff --git a/talk/app/webrtc/dtmfsenderinterface.h b/talk/app/webrtc/dtmfsenderinterface.h
deleted file mode 100644
index 7fbf57af235f445d33104b4b1006737a85de72da..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/dtmfsenderinterface.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * 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_DTMFSENDERINTERFACE_H_
-#define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
-
-#include <string>
-
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "webrtc/base/common.h"
-#include "webrtc/base/refcount.h"
-
-// This file contains interfaces for DtmfSender.
-
-namespace webrtc {
-
-// DtmfSender callback interface. Application should implement this interface
-// to get notifications from the DtmfSender.
-class DtmfSenderObserverInterface {
- public:
-  // Triggered when DTMF |tone| is sent.
-  // If |tone| is empty that means the DtmfSender has sent out all the given
-  // tones.
-  virtual void OnToneChange(const std::string& tone) = 0;
-
- protected:
-  virtual ~DtmfSenderObserverInterface() {}
-};
-
-// The interface of native implementation of the RTCDTMFSender defined by the
-// WebRTC W3C Editor's Draft.
-class DtmfSenderInterface : public rtc::RefCountInterface {
- public:
-  virtual void RegisterObserver(DtmfSenderObserverInterface* observer) = 0;
-  virtual void UnregisterObserver() = 0;
-
-  // Returns true if this DtmfSender is capable of sending DTMF.
-  // Otherwise returns false.
-  virtual bool CanInsertDtmf() = 0;
-
-  // Queues a task that sends the DTMF |tones|. The |tones| parameter is treated
-  // as a series of characters. The characters 0 through 9, A through D, #, and
-  // * generate the associated DTMF tones. The characters a to d are equivalent
-  // to A to D. The character ',' indicates a delay of 2 seconds before
-  // processing the next character in the tones parameter.
-  // Unrecognized characters are ignored.
-  // The |duration| parameter indicates the duration in ms to use for each
-  // character passed in the |tones| parameter.
-  // The duration cannot be more than 6000 or less than 70.
-  // The |inter_tone_gap| parameter indicates the gap between tones in ms.
-  // The |inter_tone_gap| must be at least 50 ms but should be as short as
-  // possible.
-  // If InsertDtmf is called on the same object while an existing task for this
-  // object to generate DTMF is still running, the previous task is canceled.
-  // Returns true on success and false on failure.
-  virtual bool InsertDtmf(const std::string& tones, int duration,
-                          int inter_tone_gap) = 0;
-
-  // Returns the track given as argument to the constructor.
-  virtual const AudioTrackInterface* track() const = 0;
-
-  // Returns the tones remaining to be played out.
-  virtual std::string tones() const = 0;
-
-  // Returns the current tone duration value in ms.
-  // This value will be the value last set via the InsertDtmf() method, or the
-  // default value of 100 ms if InsertDtmf() was never called.
-  virtual int duration() const = 0;
-
-  // Returns the current value of the between-tone gap in ms.
-  // This value will be the value last set via the InsertDtmf() method, or the
-  // default value of 50 ms if InsertDtmf() was never called.
-  virtual int inter_tone_gap() const = 0;
-
- protected:
-  virtual ~DtmfSenderInterface() {}
-};
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_