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Side by Side Diff: talk/app/webrtc/peerconnectionfactory.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /*
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
28 #ifndef TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
29 #define TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
30
31 #include <string>
32
33 #include "talk/app/webrtc/dtlsidentitystore.h"
34 #include "talk/app/webrtc/mediacontroller.h"
35 #include "talk/app/webrtc/mediastreaminterface.h"
36 #include "talk/app/webrtc/peerconnectioninterface.h"
37 #include "talk/session/media/channelmanager.h"
38 #include "webrtc/base/scoped_ptr.h"
39 #include "webrtc/base/scoped_ref_ptr.h"
40 #include "webrtc/base/thread.h"
41
42 namespace rtc {
43 class BasicNetworkManager;
44 class BasicPacketSocketFactory;
45 }
46
47 namespace webrtc {
48
49 typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
50 RefCountedDtlsIdentityStore;
51
52 class PeerConnectionFactory : public PeerConnectionFactoryInterface {
53 public:
54 virtual void SetOptions(const Options& options) {
55 options_ = options;
56 }
57
58 rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
59 const PeerConnectionInterface::RTCConfiguration& configuration,
60 const MediaConstraintsInterface* constraints,
61 rtc::scoped_ptr<cricket::PortAllocator> allocator,
62 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
63 PeerConnectionObserver* observer) override;
64
65 bool Initialize();
66
67 rtc::scoped_refptr<MediaStreamInterface>
68 CreateLocalMediaStream(const std::string& label) override;
69
70 rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
71 const MediaConstraintsInterface* constraints) override;
72
73 rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
74 cricket::VideoCapturer* capturer,
75 const MediaConstraintsInterface* constraints) override;
76
77 rtc::scoped_refptr<VideoTrackInterface>
78 CreateVideoTrack(const std::string& id,
79 VideoSourceInterface* video_source) override;
80
81 rtc::scoped_refptr<AudioTrackInterface>
82 CreateAudioTrack(const std::string& id,
83 AudioSourceInterface* audio_source) override;
84
85 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
86 void StopAecDump() override;
87 bool StartRtcEventLog(rtc::PlatformFile file) override;
88 void StopRtcEventLog() override;
89
90 virtual webrtc::MediaControllerInterface* CreateMediaController() const;
91 virtual rtc::Thread* signaling_thread();
92 virtual rtc::Thread* worker_thread();
93 const Options& options() const { return options_; }
94
95 protected:
96 PeerConnectionFactory();
97 PeerConnectionFactory(
98 rtc::Thread* worker_thread,
99 rtc::Thread* signaling_thread,
100 AudioDeviceModule* default_adm,
101 cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
102 cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
103 virtual ~PeerConnectionFactory();
104
105 private:
106 cricket::MediaEngineInterface* CreateMediaEngine_w();
107
108 bool owns_ptrs_;
109 bool wraps_current_thread_;
110 rtc::Thread* signaling_thread_;
111 rtc::Thread* worker_thread_;
112 Options options_;
113 // External Audio device used for audio playback.
114 rtc::scoped_refptr<AudioDeviceModule> default_adm_;
115 rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
116 // External Video encoder factory. This can be NULL if the client has not
117 // injected any. In that case, video engine will use the internal SW encoder.
118 rtc::scoped_ptr<cricket::WebRtcVideoEncoderFactory>
119 video_encoder_factory_;
120 // External Video decoder factory. This can be NULL if the client has not
121 // injected any. In that case, video engine will use the internal SW decoder.
122 rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
123 video_decoder_factory_;
124 rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
125 rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
126
127 rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
128 };
129
130 } // namespace webrtc
131
132 #endif // TALK_APP_WEBRTC_PEERCONNECTIONFACTORY_H_
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