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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2011, Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27#ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
28#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
29
30#include <string>
31
32#include "talk/app/webrtc/mediastreaminterface.h"
33#include "talk/app/webrtc/peerconnectioninterface.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000034#include "talk/session/media/channelmanager.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000035#include "webrtc/base/scoped_ptr.h"
36#include "webrtc/base/thread.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037
38namespace webrtc {
39
40class PeerConnectionFactory : public PeerConnectionFactoryInterface,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000041 public rtc::MessageHandler {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +000043 virtual void SetOptions(const Options& options) {
44 options_ = options;
45 }
46
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000047 virtual rtc::scoped_refptr<PeerConnectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048 CreatePeerConnection(
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000049 const PeerConnectionInterface::RTCConfiguration& configuration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000050 const MediaConstraintsInterface* constraints,
51 PortAllocatorFactoryInterface* allocator_factory,
52 DTLSIdentityServiceInterface* dtls_identity_service,
53 PeerConnectionObserver* observer);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000054
henrike@webrtc.org28e20752013-07-10 00:45:36 +000055 bool Initialize();
56
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000057 virtual rtc::scoped_refptr<MediaStreamInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000058 CreateLocalMediaStream(const std::string& label);
59
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000060 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061 const MediaConstraintsInterface* constraints);
62
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000063 virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064 cricket::VideoCapturer* capturer,
65 const MediaConstraintsInterface* constraints);
66
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000067 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068 CreateVideoTrack(const std::string& id,
69 VideoSourceInterface* video_source);
70
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000071 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072 CreateAudioTrack(const std::string& id,
73 AudioSourceInterface* audio_source);
74
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000075 virtual bool StartAecDump(rtc::PlatformFile file);
wu@webrtc.orga9890802013-12-13 00:21:03 +000076
henrike@webrtc.org28e20752013-07-10 00:45:36 +000077 virtual cricket::ChannelManager* channel_manager();
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000078 virtual rtc::Thread* signaling_thread();
79 virtual rtc::Thread* worker_thread();
wu@webrtc.org97077a32013-10-25 21:18:33 +000080 const Options& options() const { return options_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +000081
82 protected:
83 PeerConnectionFactory();
84 PeerConnectionFactory(
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000085 rtc::Thread* worker_thread,
86 rtc::Thread* signaling_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000087 AudioDeviceModule* default_adm,
88 cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
89 cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
90 virtual ~PeerConnectionFactory();
91
henrike@webrtc.org28e20752013-07-10 00:45:36 +000092 private:
93 bool Initialize_s();
94 void Terminate_s();
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000095 rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource_s(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000096 const MediaConstraintsInterface* constraints);
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000097 rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource_s(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000098 cricket::VideoCapturer* capturer,
99 const MediaConstraintsInterface* constraints);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000100
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000101 rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_s(
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000102 const PeerConnectionInterface::RTCConfiguration& configuration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103 const MediaConstraintsInterface* constraints,
104 PortAllocatorFactoryInterface* allocator_factory,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000105 DTLSIdentityServiceInterface* dtls_identity_service,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106 PeerConnectionObserver* observer);
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000107
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000108 bool StartAecDump_s(rtc::PlatformFile file);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000109
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000110 // Implements rtc::MessageHandler.
111 void OnMessage(rtc::Message* msg);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000112
113 bool owns_ptrs_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000114 rtc::Thread* signaling_thread_;
115 rtc::Thread* worker_thread_;
wu@webrtc.org97077a32013-10-25 21:18:33 +0000116 Options options_;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000117 rtc::scoped_refptr<PortAllocatorFactoryInterface> allocator_factory_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 // External Audio device used for audio playback.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000119 rtc::scoped_refptr<AudioDeviceModule> default_adm_;
120 rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121 // External Video encoder factory. This can be NULL if the client has not
122 // injected any. In that case, video engine will use the internal SW encoder.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000123 rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000124 video_encoder_factory_;
125 // External Video decoder factory. This can be NULL if the client has not
126 // injected any. In that case, video engine will use the internal SW decoder.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000127 rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000128 video_decoder_factory_;
129};
130
131} // namespace webrtc
132
133#endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_