webrtc_m130/talk/app/webrtc/peerconnectionfactory.h

/*
 * libjingle
 * Copyright 2011 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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 */

#ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_

#include <string>

#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"

namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
}

namespace webrtc {

typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
    RefCountedDtlsIdentityStore;

class PeerConnectionFactory : public PeerConnectionFactoryInterface {
 public:
  virtual void SetOptions(const Options& options) {
    options_ = options;
  }

  // webrtc::PeerConnectionFactoryInterface override;
  // TODO(deadbeef): Get rid of this overload once clients are moved to the
  // new version.
  rtc::scoped_refptr<PeerConnectionInterface>
      CreatePeerConnection(
          const PeerConnectionInterface::RTCConfiguration& configuration,
          const MediaConstraintsInterface* constraints,
          PortAllocatorFactoryInterface* allocator_factory,
          rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
          PeerConnectionObserver* observer) override;

  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
      const PeerConnectionInterface::RTCConfiguration& configuration,
      const MediaConstraintsInterface* constraints,
      rtc::scoped_ptr<cricket::PortAllocator> allocator,
      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
      PeerConnectionObserver* observer) override;

  bool Initialize();

  rtc::scoped_refptr<MediaStreamInterface>
      CreateLocalMediaStream(const std::string& label) override;

  rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
      const MediaConstraintsInterface* constraints) override;

  rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
      cricket::VideoCapturer* capturer,
      const MediaConstraintsInterface* constraints) override;

  rtc::scoped_refptr<VideoTrackInterface>
      CreateVideoTrack(const std::string& id,
                       VideoSourceInterface* video_source) override;

  rtc::scoped_refptr<AudioTrackInterface>
      CreateAudioTrack(const std::string& id,
                       AudioSourceInterface* audio_source) override;

  bool StartAecDump(rtc::PlatformFile file) override;
  void StopAecDump() override;
  bool StartRtcEventLog(rtc::PlatformFile file) override;
  void StopRtcEventLog() override;

  virtual webrtc::MediaControllerInterface* CreateMediaController() const;
  virtual rtc::Thread* signaling_thread();
  virtual rtc::Thread* worker_thread();
  const Options& options() const { return options_; }

 protected:
  PeerConnectionFactory();
  PeerConnectionFactory(
      rtc::Thread* worker_thread,
      rtc::Thread* signaling_thread,
      AudioDeviceModule* default_adm,
      cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
      cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
  virtual ~PeerConnectionFactory();

 private:
  cricket::MediaEngineInterface* CreateMediaEngine_w();

  bool owns_ptrs_;
  bool wraps_current_thread_;
  rtc::Thread* signaling_thread_;
  rtc::Thread* worker_thread_;
  Options options_;
  rtc::scoped_refptr<PortAllocatorFactoryInterface> default_allocator_factory_;
  // External Audio device used for audio playback.
  rtc::scoped_refptr<AudioDeviceModule> default_adm_;
  rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
  // External Video encoder factory. This can be NULL if the client has not
  // injected any. In that case, video engine will use the internal SW encoder.
  rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
      video_encoder_factory_;
  // External Video decoder factory. This can be NULL if the client has not
  // injected any. In that case, video engine will use the internal SW decoder.
  rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
      video_decoder_factory_;
  rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
  rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;

  rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
};

}  // namespace webrtc

#endif  // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_