henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 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 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 | |
| 49 | namespace cricket { |
| 50 | |
buildbot@webrtc.org | ca27236 | 2014-05-08 23:10:23 +0000 | [diff] [blame^] | 51 | struct AudioInfo; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 52 | class MediaSessionClient; |
| 53 | class BaseChannel; |
| 54 | class VoiceChannel; |
| 55 | class VideoChannel; |
| 56 | class DataChannel; |
| 57 | |
| 58 | // Can't typedef this easily since it's forward declared as struct elsewhere. |
| 59 | struct CallOptions : public MediaSessionOptions { |
| 60 | }; |
| 61 | |
buildbot@webrtc.org | ca27236 | 2014-05-08 23:10:23 +0000 | [diff] [blame^] | 62 | // CurrentSpeakerMonitor used to have a dependency on Call. To remove this |
| 63 | // dependency, we create AudioSourceContext. CurrentSpeakerMonitor depends on |
| 64 | // AudioSourceContext. |
| 65 | // AudioSourceProxy acts as a proxy so that when SignalAudioMonitor |
| 66 | // in Call is triggered, SignalAudioMonitor in AudioSourceContext is triggered. |
| 67 | // Likewise, when OnMediaStreamsUpdate in Call is triggered, |
| 68 | // OnMediaStreamsUpdate in AudioSourceContext is triggered. |
| 69 | class AudioSourceProxy: public AudioSourceContext, public sigslot::has_slots<> { |
| 70 | public: |
| 71 | explicit AudioSourceProxy(Call* call); |
| 72 | |
| 73 | private: |
| 74 | void OnAudioMonitor(Call* call, const AudioInfo& info); |
| 75 | void OnMediaStreamsUpdate(Call* call, cricket::Session*, |
| 76 | const cricket::MediaStreams&, const cricket::MediaStreams&); |
| 77 | |
| 78 | AudioSourceContext* audio_source_context_; |
| 79 | Call* call_; |
| 80 | }; |
| 81 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 82 | class Call : public talk_base::MessageHandler, public sigslot::has_slots<> { |
| 83 | public: |
| 84 | explicit Call(MediaSessionClient* session_client); |
| 85 | ~Call(); |
| 86 | |
| 87 | // |initiator| can be empty. |
| 88 | Session* InitiateSession(const buzz::Jid& to, const buzz::Jid& initiator, |
| 89 | const CallOptions& options); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 90 | Session* InitiateSession(const std::string& id, const buzz::Jid& to, |
| 91 | const CallOptions& options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 92 | void AcceptSession(Session* session, const CallOptions& options); |
| 93 | void RejectSession(Session* session); |
| 94 | void TerminateSession(Session* session); |
| 95 | void Terminate(); |
| 96 | bool SendViewRequest(Session* session, |
| 97 | const ViewRequest& view_request); |
| 98 | void SetLocalRenderer(VideoRenderer* renderer); |
| 99 | void SetVideoRenderer(Session* session, uint32 ssrc, |
| 100 | VideoRenderer* renderer); |
| 101 | void StartConnectionMonitor(Session* session, int cms); |
| 102 | void StopConnectionMonitor(Session* session); |
| 103 | void StartAudioMonitor(Session* session, int cms); |
| 104 | void StopAudioMonitor(Session* session); |
| 105 | bool IsAudioMonitorRunning(Session* session); |
| 106 | void StartSpeakerMonitor(Session* session); |
| 107 | void StopSpeakerMonitor(Session* session); |
| 108 | void Mute(bool mute); |
| 109 | void MuteVideo(bool mute); |
| 110 | bool SendData(Session* session, |
| 111 | const SendDataParams& params, |
| 112 | const talk_base::Buffer& payload, |
| 113 | SendDataResult* result); |
| 114 | void PressDTMF(int event); |
| 115 | bool StartScreencast(Session* session, |
| 116 | const std::string& stream_name, uint32 ssrc, |
| 117 | const ScreencastId& screencastid, int fps); |
| 118 | bool StopScreencast(Session* session, |
| 119 | const std::string& stream_name, uint32 ssrc); |
| 120 | |
| 121 | std::vector<Session*> sessions(); |
| 122 | uint32 id(); |
| 123 | bool has_video() const { return has_video_; } |
| 124 | bool has_data() const { return has_data_; } |
| 125 | bool muted() const { return muted_; } |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 126 | bool video() const { return has_video_; } |
| 127 | bool secure() const; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 128 | bool video_muted() const { return video_muted_; } |
| 129 | const std::vector<StreamParams>* GetDataRecvStreams(Session* session) const { |
| 130 | MediaStreams* recv_streams = GetMediaStreams(session); |
| 131 | return recv_streams ? &recv_streams->data() : NULL; |
| 132 | } |
| 133 | const std::vector<StreamParams>* GetVideoRecvStreams(Session* session) const { |
| 134 | MediaStreams* recv_streams = GetMediaStreams(session); |
| 135 | return recv_streams ? &recv_streams->video() : NULL; |
| 136 | } |
| 137 | const std::vector<StreamParams>* GetAudioRecvStreams(Session* session) const { |
| 138 | MediaStreams* recv_streams = GetMediaStreams(session); |
| 139 | return recv_streams ? &recv_streams->audio() : NULL; |
| 140 | } |
mallinath@webrtc.org | a27be8e | 2013-09-27 23:04:10 +0000 | [diff] [blame] | 141 | VoiceChannel* GetVoiceChannel(Session* session) const; |
| 142 | VideoChannel* GetVideoChannel(Session* session) const; |
| 143 | DataChannel* GetDataChannel(Session* session) const; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 144 | // Public just for unit tests |
| 145 | VideoContentDescription* CreateVideoStreamUpdate(const StreamParams& stream); |
| 146 | // Takes ownership of video. |
| 147 | void SendVideoStreamUpdate(Session* session, VideoContentDescription* video); |
| 148 | |
| 149 | // Setting this to false will cause the call to have a longer timeout and |
| 150 | // for the SignalSetupToCallVoicemail to never fire. |
| 151 | void set_send_to_voicemail(bool send_to_voicemail) { |
| 152 | send_to_voicemail_ = send_to_voicemail; |
| 153 | } |
| 154 | bool send_to_voicemail() { return send_to_voicemail_; } |
| 155 | const VoiceMediaInfo& last_voice_media_info() const { |
| 156 | return last_voice_media_info_; |
| 157 | } |
| 158 | |
| 159 | // Sets a flag on the chatapp that will redirect the call to voicemail once |
| 160 | // the call has been terminated |
| 161 | sigslot::signal0<> SignalSetupToCallVoicemail; |
| 162 | sigslot::signal2<Call*, Session*> SignalAddSession; |
| 163 | sigslot::signal2<Call*, Session*> SignalRemoveSession; |
| 164 | sigslot::signal3<Call*, Session*, Session::State> |
| 165 | SignalSessionState; |
| 166 | sigslot::signal3<Call*, Session*, Session::Error> |
| 167 | SignalSessionError; |
| 168 | sigslot::signal3<Call*, Session*, const std::string &> |
| 169 | SignalReceivedTerminateReason; |
| 170 | sigslot::signal2<Call*, const std::vector<ConnectionInfo> &> |
| 171 | SignalConnectionMonitor; |
| 172 | sigslot::signal2<Call*, const VoiceMediaInfo&> SignalMediaMonitor; |
| 173 | sigslot::signal2<Call*, const AudioInfo&> SignalAudioMonitor; |
| 174 | // Empty nick on StreamParams means "unknown". |
| 175 | // No ssrcs in StreamParams means "no current speaker". |
| 176 | sigslot::signal3<Call*, |
| 177 | Session*, |
| 178 | const StreamParams&> SignalSpeakerMonitor; |
| 179 | sigslot::signal2<Call*, const std::vector<ConnectionInfo> &> |
| 180 | SignalVideoConnectionMonitor; |
| 181 | sigslot::signal2<Call*, const VideoMediaInfo&> SignalVideoMediaMonitor; |
| 182 | // Gives added streams and removed streams, in that order. |
| 183 | sigslot::signal4<Call*, |
| 184 | Session*, |
| 185 | const MediaStreams&, |
| 186 | const MediaStreams&> SignalMediaStreamsUpdate; |
| 187 | sigslot::signal3<Call*, |
| 188 | const ReceiveDataParams&, |
| 189 | const talk_base::Buffer&> SignalDataReceived; |
| 190 | |
buildbot@webrtc.org | ca27236 | 2014-05-08 23:10:23 +0000 | [diff] [blame^] | 191 | AudioSourceProxy* GetAudioSourceProxy(); |
| 192 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 193 | private: |
| 194 | void OnMessage(talk_base::Message* message); |
| 195 | void OnSessionState(BaseSession* base_session, BaseSession::State state); |
| 196 | void OnSessionError(BaseSession* base_session, Session::Error error); |
| 197 | void OnSessionInfoMessage( |
| 198 | Session* session, const buzz::XmlElement* action_elem); |
| 199 | void OnViewRequest( |
| 200 | Session* session, const ViewRequest& view_request); |
| 201 | void OnRemoteDescriptionUpdate( |
| 202 | BaseSession* base_session, const ContentInfos& updated_contents); |
| 203 | void OnReceivedTerminateReason(Session* session, const std::string &reason); |
| 204 | void IncomingSession(Session* session, const SessionDescription* offer); |
| 205 | // Returns true on success. |
| 206 | bool AddSession(Session* session, const SessionDescription* offer); |
| 207 | void RemoveSession(Session* session); |
| 208 | void EnableChannels(bool enable); |
| 209 | void EnableSessionChannels(Session* session, bool enable); |
| 210 | void Join(Call* call, bool enable); |
| 211 | void OnConnectionMonitor(VoiceChannel* channel, |
| 212 | const std::vector<ConnectionInfo> &infos); |
| 213 | void OnMediaMonitor(VoiceChannel* channel, const VoiceMediaInfo& info); |
| 214 | void OnAudioMonitor(VoiceChannel* channel, const AudioInfo& info); |
| 215 | void OnSpeakerMonitor(CurrentSpeakerMonitor* monitor, uint32 ssrc); |
| 216 | void OnConnectionMonitor(VideoChannel* channel, |
| 217 | const std::vector<ConnectionInfo> &infos); |
| 218 | void OnMediaMonitor(VideoChannel* channel, const VideoMediaInfo& info); |
| 219 | void OnDataReceived(DataChannel* channel, |
| 220 | const ReceiveDataParams& params, |
| 221 | const talk_base::Buffer& payload); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 222 | MediaStreams* GetMediaStreams(Session* session) const; |
| 223 | void UpdateRemoteMediaStreams(Session* session, |
| 224 | const ContentInfos& updated_contents, |
| 225 | bool update_channels); |
| 226 | bool UpdateVoiceChannelRemoteContent(Session* session, |
| 227 | const AudioContentDescription* audio); |
| 228 | bool UpdateVideoChannelRemoteContent(Session* session, |
| 229 | const VideoContentDescription* video); |
| 230 | bool UpdateDataChannelRemoteContent(Session* session, |
| 231 | const DataContentDescription* data); |
| 232 | void UpdateRecvStreams(const std::vector<StreamParams>& update_streams, |
| 233 | BaseChannel* channel, |
| 234 | std::vector<StreamParams>* recv_streams, |
| 235 | std::vector<StreamParams>* added_streams, |
| 236 | std::vector<StreamParams>* removed_streams); |
| 237 | void AddRecvStreams(const std::vector<StreamParams>& added_streams, |
| 238 | BaseChannel* channel, |
| 239 | std::vector<StreamParams>* recv_streams); |
| 240 | void AddRecvStream(const StreamParams& stream, |
| 241 | BaseChannel* channel, |
| 242 | std::vector<StreamParams>* recv_streams); |
| 243 | void RemoveRecvStreams(const std::vector<StreamParams>& removed_streams, |
| 244 | BaseChannel* channel, |
| 245 | std::vector<StreamParams>* recv_streams); |
| 246 | void RemoveRecvStream(const StreamParams& stream, |
| 247 | BaseChannel* channel, |
| 248 | std::vector<StreamParams>* recv_streams); |
| 249 | void ContinuePlayDTMF(); |
| 250 | bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc); |
| 251 | bool StopAllScreencastsWithoutSendingUpdate(Session* session); |
wu@webrtc.org | 9dba525 | 2013-08-05 20:36:57 +0000 | [diff] [blame] | 252 | bool SessionDescriptionContainsCrypto(const SessionDescription* sdesc) const; |
| 253 | Session* InternalInitiateSession(const std::string& id, |
| 254 | const buzz::Jid& to, |
| 255 | const std::string& initiator_name, |
| 256 | const CallOptions& options); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 257 | |
| 258 | uint32 id_; |
| 259 | MediaSessionClient* session_client_; |
| 260 | |
| 261 | struct StartedCapture { |
| 262 | StartedCapture(cricket::VideoCapturer* capturer, |
| 263 | const cricket::VideoFormat& format) : |
| 264 | capturer(capturer), |
| 265 | format(format) { |
| 266 | } |
| 267 | cricket::VideoCapturer* capturer; |
| 268 | cricket::VideoFormat format; |
| 269 | }; |
| 270 | typedef std::map<uint32, StartedCapture> StartedScreencastMap; |
| 271 | |
| 272 | struct MediaSession { |
| 273 | Session* session; |
| 274 | VoiceChannel* voice_channel; |
| 275 | VideoChannel* video_channel; |
| 276 | DataChannel* data_channel; |
| 277 | MediaStreams* recv_streams; |
| 278 | StartedScreencastMap started_screencasts; |
| 279 | }; |
| 280 | |
| 281 | // Create a map of media sessions, keyed off session->id(). |
| 282 | typedef std::map<std::string, MediaSession> MediaSessionMap; |
| 283 | MediaSessionMap media_session_map_; |
| 284 | |
| 285 | std::map<std::string, CurrentSpeakerMonitor*> speaker_monitor_map_; |
| 286 | VideoRenderer* local_renderer_; |
| 287 | bool has_video_; |
| 288 | bool has_data_; |
| 289 | bool muted_; |
| 290 | bool video_muted_; |
| 291 | bool send_to_voicemail_; |
| 292 | |
| 293 | // DTMF tones have to be queued up so that we don't flood the call. We |
| 294 | // keep a deque (doubely ended queue) of them around. While one is playing we |
| 295 | // set the playing_dtmf_ bit and schedule a message in XX msec to clear that |
| 296 | // bit or start the next tone playing. |
| 297 | std::deque<int> queued_dtmf_; |
| 298 | bool playing_dtmf_; |
| 299 | |
| 300 | VoiceMediaInfo last_voice_media_info_; |
| 301 | |
buildbot@webrtc.org | ca27236 | 2014-05-08 23:10:23 +0000 | [diff] [blame^] | 302 | talk_base::scoped_ptr<AudioSourceProxy> audio_source_proxy_; |
| 303 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 304 | friend class MediaSessionClient; |
| 305 | }; |
| 306 | |
| 307 | } // namespace cricket |
| 308 | |
| 309 | #endif // TALK_SESSION_MEDIA_CALL_H_ |