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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
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 TALK_SESSION_MEDIA_CALL_H_
29#define TALK_SESSION_MEDIA_CALL_H_
30
31#include <deque>
32#include <map>
33#include <string>
34#include <vector>
35
36#include "talk/base/messagequeue.h"
37#include "talk/media/base/mediachannel.h"
38#include "talk/media/base/screencastid.h"
39#include "talk/media/base/streamparams.h"
40#include "talk/media/base/videocommon.h"
41#include "talk/p2p/base/session.h"
42#include "talk/p2p/client/socketmonitor.h"
43#include "talk/session/media/audiomonitor.h"
44#include "talk/session/media/currentspeakermonitor.h"
45#include "talk/session/media/mediamessages.h"
46#include "talk/session/media/mediasession.h"
47#include "talk/xmpp/jid.h"
48
49namespace cricket {
50
51class MediaSessionClient;
52class BaseChannel;
53class VoiceChannel;
54class VideoChannel;
55class DataChannel;
56
57// Can't typedef this easily since it's forward declared as struct elsewhere.
58struct CallOptions : public MediaSessionOptions {
59};
60
61class Call : public talk_base::MessageHandler, public sigslot::has_slots<> {
62 public:
63 explicit Call(MediaSessionClient* session_client);
64 ~Call();
65
66 // |initiator| can be empty.
67 Session* InitiateSession(const buzz::Jid& to, const buzz::Jid& initiator,
68 const CallOptions& options);
wu@webrtc.org9dba5252013-08-05 20:36:57 +000069 Session* InitiateSession(const std::string& id, const buzz::Jid& to,
70 const CallOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071 void AcceptSession(Session* session, const CallOptions& options);
72 void RejectSession(Session* session);
73 void TerminateSession(Session* session);
74 void Terminate();
75 bool SendViewRequest(Session* session,
76 const ViewRequest& view_request);
77 void SetLocalRenderer(VideoRenderer* renderer);
78 void SetVideoRenderer(Session* session, uint32 ssrc,
79 VideoRenderer* renderer);
80 void StartConnectionMonitor(Session* session, int cms);
81 void StopConnectionMonitor(Session* session);
82 void StartAudioMonitor(Session* session, int cms);
83 void StopAudioMonitor(Session* session);
84 bool IsAudioMonitorRunning(Session* session);
85 void StartSpeakerMonitor(Session* session);
86 void StopSpeakerMonitor(Session* session);
87 void Mute(bool mute);
88 void MuteVideo(bool mute);
89 bool SendData(Session* session,
90 const SendDataParams& params,
91 const talk_base::Buffer& payload,
92 SendDataResult* result);
93 void PressDTMF(int event);
94 bool StartScreencast(Session* session,
95 const std::string& stream_name, uint32 ssrc,
96 const ScreencastId& screencastid, int fps);
97 bool StopScreencast(Session* session,
98 const std::string& stream_name, uint32 ssrc);
99
100 std::vector<Session*> sessions();
101 uint32 id();
102 bool has_video() const { return has_video_; }
103 bool has_data() const { return has_data_; }
104 bool muted() const { return muted_; }
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000105 bool video() const { return has_video_; }
106 bool secure() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107 bool video_muted() const { return video_muted_; }
108 const std::vector<StreamParams>* GetDataRecvStreams(Session* session) const {
109 MediaStreams* recv_streams = GetMediaStreams(session);
110 return recv_streams ? &recv_streams->data() : NULL;
111 }
112 const std::vector<StreamParams>* GetVideoRecvStreams(Session* session) const {
113 MediaStreams* recv_streams = GetMediaStreams(session);
114 return recv_streams ? &recv_streams->video() : NULL;
115 }
116 const std::vector<StreamParams>* GetAudioRecvStreams(Session* session) const {
117 MediaStreams* recv_streams = GetMediaStreams(session);
118 return recv_streams ? &recv_streams->audio() : NULL;
119 }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000120 VoiceChannel* GetVoiceChannel(Session* session) const;
121 VideoChannel* GetVideoChannel(Session* session) const;
122 DataChannel* GetDataChannel(Session* session) const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123 // Public just for unit tests
124 VideoContentDescription* CreateVideoStreamUpdate(const StreamParams& stream);
125 // Takes ownership of video.
126 void SendVideoStreamUpdate(Session* session, VideoContentDescription* video);
127
128 // Setting this to false will cause the call to have a longer timeout and
129 // for the SignalSetupToCallVoicemail to never fire.
130 void set_send_to_voicemail(bool send_to_voicemail) {
131 send_to_voicemail_ = send_to_voicemail;
132 }
133 bool send_to_voicemail() { return send_to_voicemail_; }
134 const VoiceMediaInfo& last_voice_media_info() const {
135 return last_voice_media_info_;
136 }
137
138 // Sets a flag on the chatapp that will redirect the call to voicemail once
139 // the call has been terminated
140 sigslot::signal0<> SignalSetupToCallVoicemail;
141 sigslot::signal2<Call*, Session*> SignalAddSession;
142 sigslot::signal2<Call*, Session*> SignalRemoveSession;
143 sigslot::signal3<Call*, Session*, Session::State>
144 SignalSessionState;
145 sigslot::signal3<Call*, Session*, Session::Error>
146 SignalSessionError;
147 sigslot::signal3<Call*, Session*, const std::string &>
148 SignalReceivedTerminateReason;
149 sigslot::signal2<Call*, const std::vector<ConnectionInfo> &>
150 SignalConnectionMonitor;
151 sigslot::signal2<Call*, const VoiceMediaInfo&> SignalMediaMonitor;
152 sigslot::signal2<Call*, const AudioInfo&> SignalAudioMonitor;
153 // Empty nick on StreamParams means "unknown".
154 // No ssrcs in StreamParams means "no current speaker".
155 sigslot::signal3<Call*,
156 Session*,
157 const StreamParams&> SignalSpeakerMonitor;
158 sigslot::signal2<Call*, const std::vector<ConnectionInfo> &>
159 SignalVideoConnectionMonitor;
160 sigslot::signal2<Call*, const VideoMediaInfo&> SignalVideoMediaMonitor;
161 // Gives added streams and removed streams, in that order.
162 sigslot::signal4<Call*,
163 Session*,
164 const MediaStreams&,
165 const MediaStreams&> SignalMediaStreamsUpdate;
166 sigslot::signal3<Call*,
167 const ReceiveDataParams&,
168 const talk_base::Buffer&> SignalDataReceived;
169
170 private:
171 void OnMessage(talk_base::Message* message);
172 void OnSessionState(BaseSession* base_session, BaseSession::State state);
173 void OnSessionError(BaseSession* base_session, Session::Error error);
174 void OnSessionInfoMessage(
175 Session* session, const buzz::XmlElement* action_elem);
176 void OnViewRequest(
177 Session* session, const ViewRequest& view_request);
178 void OnRemoteDescriptionUpdate(
179 BaseSession* base_session, const ContentInfos& updated_contents);
180 void OnReceivedTerminateReason(Session* session, const std::string &reason);
181 void IncomingSession(Session* session, const SessionDescription* offer);
182 // Returns true on success.
183 bool AddSession(Session* session, const SessionDescription* offer);
184 void RemoveSession(Session* session);
185 void EnableChannels(bool enable);
186 void EnableSessionChannels(Session* session, bool enable);
187 void Join(Call* call, bool enable);
188 void OnConnectionMonitor(VoiceChannel* channel,
189 const std::vector<ConnectionInfo> &infos);
190 void OnMediaMonitor(VoiceChannel* channel, const VoiceMediaInfo& info);
191 void OnAudioMonitor(VoiceChannel* channel, const AudioInfo& info);
192 void OnSpeakerMonitor(CurrentSpeakerMonitor* monitor, uint32 ssrc);
193 void OnConnectionMonitor(VideoChannel* channel,
194 const std::vector<ConnectionInfo> &infos);
195 void OnMediaMonitor(VideoChannel* channel, const VideoMediaInfo& info);
196 void OnDataReceived(DataChannel* channel,
197 const ReceiveDataParams& params,
198 const talk_base::Buffer& payload);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000199 MediaStreams* GetMediaStreams(Session* session) const;
200 void UpdateRemoteMediaStreams(Session* session,
201 const ContentInfos& updated_contents,
202 bool update_channels);
203 bool UpdateVoiceChannelRemoteContent(Session* session,
204 const AudioContentDescription* audio);
205 bool UpdateVideoChannelRemoteContent(Session* session,
206 const VideoContentDescription* video);
207 bool UpdateDataChannelRemoteContent(Session* session,
208 const DataContentDescription* data);
209 void UpdateRecvStreams(const std::vector<StreamParams>& update_streams,
210 BaseChannel* channel,
211 std::vector<StreamParams>* recv_streams,
212 std::vector<StreamParams>* added_streams,
213 std::vector<StreamParams>* removed_streams);
214 void AddRecvStreams(const std::vector<StreamParams>& added_streams,
215 BaseChannel* channel,
216 std::vector<StreamParams>* recv_streams);
217 void AddRecvStream(const StreamParams& stream,
218 BaseChannel* channel,
219 std::vector<StreamParams>* recv_streams);
220 void RemoveRecvStreams(const std::vector<StreamParams>& removed_streams,
221 BaseChannel* channel,
222 std::vector<StreamParams>* recv_streams);
223 void RemoveRecvStream(const StreamParams& stream,
224 BaseChannel* channel,
225 std::vector<StreamParams>* recv_streams);
226 void ContinuePlayDTMF();
227 bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc);
228 bool StopAllScreencastsWithoutSendingUpdate(Session* session);
wu@webrtc.org9dba5252013-08-05 20:36:57 +0000229 bool SessionDescriptionContainsCrypto(const SessionDescription* sdesc) const;
230 Session* InternalInitiateSession(const std::string& id,
231 const buzz::Jid& to,
232 const std::string& initiator_name,
233 const CallOptions& options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000234
235 uint32 id_;
236 MediaSessionClient* session_client_;
237
238 struct StartedCapture {
239 StartedCapture(cricket::VideoCapturer* capturer,
240 const cricket::VideoFormat& format) :
241 capturer(capturer),
242 format(format) {
243 }
244 cricket::VideoCapturer* capturer;
245 cricket::VideoFormat format;
246 };
247 typedef std::map<uint32, StartedCapture> StartedScreencastMap;
248
249 struct MediaSession {
250 Session* session;
251 VoiceChannel* voice_channel;
252 VideoChannel* video_channel;
253 DataChannel* data_channel;
254 MediaStreams* recv_streams;
255 StartedScreencastMap started_screencasts;
256 };
257
258 // Create a map of media sessions, keyed off session->id().
259 typedef std::map<std::string, MediaSession> MediaSessionMap;
260 MediaSessionMap media_session_map_;
261
262 std::map<std::string, CurrentSpeakerMonitor*> speaker_monitor_map_;
263 VideoRenderer* local_renderer_;
264 bool has_video_;
265 bool has_data_;
266 bool muted_;
267 bool video_muted_;
268 bool send_to_voicemail_;
269
270 // DTMF tones have to be queued up so that we don't flood the call. We
271 // keep a deque (doubely ended queue) of them around. While one is playing we
272 // set the playing_dtmf_ bit and schedule a message in XX msec to clear that
273 // bit or start the next tone playing.
274 std::deque<int> queued_dtmf_;
275 bool playing_dtmf_;
276
277 VoiceMediaInfo last_voice_media_info_;
278
279 friend class MediaSessionClient;
280};
281
282} // namespace cricket
283
284#endif // TALK_SESSION_MEDIA_CALL_H_