/* * 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_LOCALAUDIOSOURCE_H_ #define TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_ #include "talk/app/webrtc/mediastreaminterface.h" #include "talk/app/webrtc/notifier.h" #include "talk/app/webrtc/peerconnectioninterface.h" #include "talk/media/base/mediachannel.h" #include "webrtc/base/scoped_ptr.h" // LocalAudioSource implements AudioSourceInterface. // This contains settings for switching audio processing on and off. namespace webrtc { class MediaConstraintsInterface; class LocalAudioSource : public Notifier { public: // Creates an instance of LocalAudioSource. static rtc::scoped_refptr Create( const PeerConnectionFactoryInterface::Options& options, const MediaConstraintsInterface* constraints); SourceState state() const override { return source_state_; } bool remote() const override { return false; } virtual const cricket::AudioOptions& options() const { return options_; } void AddSink(AudioTrackSinkInterface* sink) override {} void RemoveSink(AudioTrackSinkInterface* sink) override {} protected: LocalAudioSource() : source_state_(kInitializing) {} ~LocalAudioSource() override {} private: void Initialize(const PeerConnectionFactoryInterface::Options& options, const MediaConstraintsInterface* constraints); cricket::AudioOptions options_; SourceState source_state_; }; } // namespace webrtc #endif // TALK_APP_WEBRTC_LOCALAUDIOSOURCE_H_