| /* |
| * 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, |
| * 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_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 webrtc { |
| |
| typedef rtc::RefCountedObject<DtlsIdentityStoreImpl> |
| RefCountedDtlsIdentityStore; |
| |
| class PeerConnectionFactory : public PeerConnectionFactoryInterface { |
| public: |
| virtual void SetOptions(const Options& options) { |
| options_ = options; |
| } |
| |
| // webrtc::PeerConnectionFactoryInterface override; |
| 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; |
| |
| 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_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_ |