Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/dtmfsenderinterface.h b/talk/app/webrtc/dtmfsenderinterface.h
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+++ b/talk/app/webrtc/dtmfsenderinterface.h
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+/*
+ * 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 "talk/base/common.h"
+#include "talk/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 talk_base::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_
