15583c19d7
The previously disabled warnings that were inherited from talk/build/common.gypi are now replaced by target-specific disabling of only the failing warnings. Additional disabling was needed since the stricter compilation warnings that applies to code in webrtc/. License headers will be updated in a follow-up CL. Other modifications: * Updated the header guards. * Sorted the includes using chromium/src/tools/sort-headers.py except for these files: talk/app/webrtc/peerconnectionendtoend_unittest.cc talk/app/webrtc/java/jni/androidmediadecoder_jni.cc talk/app/webrtc/java/jni/androidmediaencoder_jni.cc webrtc/media/devices/win32devicemanager.cc The HAVE_SCTP define was added for the peerconnection_unittests target in api_tests.gyp. I also checked that none of SRTP_RELATIVE_PATH HAVE_SRTP HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE were used by the talk/app/webrtc code. For Chromium, the following changes will need to be applied to the roll CL that updates the DEPS for WebRTC and libjingle: https://codereview.chromium.org/1615433002 BUG=webrtc:5418 NOPRESUBMIT=True R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1610243002 . Cr-Commit-Position: refs/heads/master@{#11545}
106 lines
4.4 KiB
C++
106 lines
4.4 KiB
C++
/*
|
|
* 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 WEBRTC_API_DTMFSENDERINTERFACE_H_
|
|
#define WEBRTC_API_DTMFSENDERINTERFACE_H_
|
|
|
|
#include <string>
|
|
|
|
#include "webrtc/api/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 // WEBRTC_API_DTMFSENDERINTERFACE_H_
|