webrtc_m130/talk/app/webrtc/localaudiosource.h

/*
 * 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,
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 * 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<AudioSourceInterface> {
 public:
  // Creates an instance of LocalAudioSource.
  static rtc::scoped_refptr<LocalAudioSource> 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_