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Side by Side Diff: webrtc/libjingle/session/media/mediasessionclient.h

Issue 1175243003: Remove webrtc/libjingle/{examples,session}. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 6 months ago
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1 /*
2 * libjingle
3 * Copyright 2004 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 WEBRTC_LIBJINGLE_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
29 #define WEBRTC_LIBJINGLE_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
30
31 #include <algorithm>
32 #include <map>
33 #include <string>
34 #include <vector>
35
36 #include "talk/media/base/cryptoparams.h"
37 #include "talk/session/media/channelmanager.h"
38 #include "talk/session/media/mediasession.h"
39 #include "webrtc/base/messagequeue.h"
40 #include "webrtc/base/sigslot.h"
41 #include "webrtc/base/sigslotrepeater.h"
42 #include "webrtc/base/thread.h"
43 #include "webrtc/libjingle/session/media/call.h"
44 #include "webrtc/libjingle/session/sessionclient.h"
45 #include "webrtc/libjingle/session/sessionmanager.h"
46 #include "webrtc/p2p/base/sessiondescription.h"
47
48 namespace cricket {
49
50 class Call;
51
52 class MediaSessionClient : public SessionClient, public sigslot::has_slots<> {
53 public:
54 #if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
55 MediaSessionClient(const buzz::Jid& jid, SessionManager *manager);
56 #endif
57 // Alternative constructor, allowing injection of media_engine
58 // and device_manager.
59 MediaSessionClient(const buzz::Jid& jid, SessionManager *manager,
60 MediaEngineInterface* media_engine,
61 DataEngineInterface* data_media_engine,
62 DeviceManagerInterface* device_manager);
63 ~MediaSessionClient();
64
65 const buzz::Jid &jid() const { return jid_; }
66 SessionManager* session_manager() const { return session_manager_; }
67 ChannelManager* channel_manager() const { return channel_manager_; }
68
69 // Return mapping of call ids to Calls.
70 const std::map<uint32, Call *>& calls() const { return calls_; }
71
72 // The settings below combine with the settings on SessionManager to choose
73
74 // whether SDES-SRTP, DTLS-SRTP, or no security should be used. The possible
75 // combinations are shown in the following table. Note that where either DTLS
76 // or SDES is possible, DTLS is preferred. Thus to require either SDES or
77 // DTLS, but not mandate DTLS, set SDES to require and DTLS to enable.
78 //
79 // | SDES:Disable | SDES:Enable | SDES:Require |
80 // ----------------------------------------------------------------|
81 // DTLS:Disable | No SRTP | SDES Optional | SDES Mandatory |
82 // DTLS:Enable | DTLS Optional | DTLS/SDES Opt | DTLS/SDES Mand |
83 // DTLS:Require | DTLS Mandatory | DTLS Mandatory | DTLS Mandatory |
84
85 // Control use of SDES-SRTP.
86 SecurePolicy secure() const { return desc_factory_.secure(); }
87 void set_secure(SecurePolicy s) { desc_factory_.set_secure(s); }
88
89 // Control use of multiple sessions in a call.
90 void set_multisession_enabled(bool multisession_enabled) {
91 multisession_enabled_ = multisession_enabled;
92 }
93
94 int GetCapabilities() { return channel_manager_->GetCapabilities(); }
95
96 Call *CreateCall();
97 void DestroyCall(Call *call);
98
99 Call *GetFocus();
100 void SetFocus(Call *call);
101
102 void JoinCalls(Call *call_to_join, Call *call);
103
104 bool GetAudioInputDevices(std::vector<std::string>* names) {
105 return channel_manager_->GetAudioInputDevices(names);
106 }
107 bool GetAudioOutputDevices(std::vector<std::string>* names) {
108 return channel_manager_->GetAudioOutputDevices(names);
109 }
110 bool GetVideoCaptureDevices(std::vector<std::string>* names) {
111 return channel_manager_->GetVideoCaptureDevices(names);
112 }
113
114 bool SetAudioOptions(const std::string& in_name, const std::string& out_name,
115 const AudioOptions& options) {
116 return channel_manager_->SetAudioOptions(in_name, out_name, options);
117 }
118 bool SetOutputVolume(int level) {
119 return channel_manager_->SetOutputVolume(level);
120 }
121 bool SetCaptureDevice(const std::string& cam_device) {
122 return channel_manager_->SetCaptureDevice(cam_device);
123 }
124
125 SessionDescription* CreateOffer(const CallOptions& options) {
126 return desc_factory_.CreateOffer(options, NULL);
127 }
128 SessionDescription* CreateAnswer(const SessionDescription* offer,
129 const CallOptions& options) {
130 return desc_factory_.CreateAnswer(offer, options, NULL);
131 }
132
133 sigslot::signal2<Call *, Call *> SignalFocus;
134 sigslot::signal1<Call *> SignalCallCreate;
135 sigslot::signal1<Call *> SignalCallDestroy;
136 sigslot::repeater0<> SignalDevicesChange;
137
138 virtual bool ParseContent(SignalingProtocol protocol,
139 const buzz::XmlElement* elem,
140 ContentDescription** content,
141 ParseError* error);
142 virtual bool IsWritable(SignalingProtocol protocol,
143 const ContentDescription* content);
144 virtual bool WriteContent(SignalingProtocol protocol,
145 const ContentDescription* content,
146 buzz::XmlElement** elem,
147 WriteError* error);
148
149 private:
150 void Construct();
151 void OnSessionCreate(Session *session, bool received_initiate);
152 void OnSessionState(BaseSession *session, BaseSession::State state);
153 void OnSessionDestroy(Session *session);
154 Session *CreateSession(Call *call);
155 Session *CreateSession(const std::string& id, Call* call);
156 Call *FindCallByRemoteName(const std::string &remote_name);
157
158 buzz::Jid jid_;
159 SessionManager* session_manager_;
160 Call *focus_call_;
161 ChannelManager *channel_manager_;
162 MediaSessionDescriptionFactory desc_factory_;
163 bool multisession_enabled_;
164 std::map<uint32, Call *> calls_;
165
166 // Maintain a mapping of session id to call.
167 typedef std::map<std::string, Call *> SessionMap;
168 SessionMap session_map_;
169
170 friend class Call;
171 };
172
173 } // namespace cricket
174
175 #endif // WEBRTC_LIBJINGLE_SESSION_MEDIA_MEDIASESSIONCLIENT_H_
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