1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
|
/*
* 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_
|
