| /* |
| * 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_ |