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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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83 bool initialized() const { return initialized_; } | 83 bool initialized() const { return initialized_; } |
84 // Starts up the media engine. | 84 // Starts up the media engine. |
85 bool Init(); | 85 bool Init(); |
86 // Shuts down the media engine. | 86 // Shuts down the media engine. |
87 void Terminate(); | 87 void Terminate(); |
88 | 88 |
89 // The operations below all occur on the worker thread. | 89 // The operations below all occur on the worker thread. |
90 // Creates a voice channel, to be associated with the specified session. | 90 // Creates a voice channel, to be associated with the specified session. |
91 VoiceChannel* CreateVoiceChannel( | 91 VoiceChannel* CreateVoiceChannel( |
92 webrtc::MediaControllerInterface* media_controller, | 92 webrtc::MediaControllerInterface* media_controller, |
93 TransportController* transport_controller, | 93 TransportChannel* rtp_transport, |
| 94 TransportChannel* rtcp_transport, |
| 95 rtc::Thread* signaling_thread, |
94 const std::string& content_name, | 96 const std::string& content_name, |
95 const std::string* bundle_transport_name, | 97 const std::string* bundle_transport_name, |
96 bool rtcp, | 98 bool rtcp, |
97 bool srtp_required, | 99 bool srtp_required, |
98 const AudioOptions& options); | 100 const AudioOptions& options); |
99 // Destroys a voice channel created with the Create API. | 101 // Destroys a voice channel created with the Create API. |
100 void DestroyVoiceChannel(VoiceChannel* voice_channel); | 102 void DestroyVoiceChannel(VoiceChannel* voice_channel); |
101 // Creates a video channel, synced with the specified voice channel, and | 103 // Creates a video channel, synced with the specified voice channel, and |
102 // associated with the specified session. | 104 // associated with the specified session. |
103 VideoChannel* CreateVideoChannel( | 105 VideoChannel* CreateVideoChannel( |
104 webrtc::MediaControllerInterface* media_controller, | 106 webrtc::MediaControllerInterface* media_controller, |
105 TransportController* transport_controller, | 107 TransportChannel* rtp_transport, |
| 108 TransportChannel* rtcp_transport, |
| 109 rtc::Thread* signaling_thread, |
106 const std::string& content_name, | 110 const std::string& content_name, |
107 const std::string* bundle_transport_name, | 111 const std::string* bundle_transport_name, |
108 bool rtcp, | 112 bool rtcp, |
109 bool srtp_required, | 113 bool srtp_required, |
110 const VideoOptions& options); | 114 const VideoOptions& options); |
111 // Destroys a video channel created with the Create API. | 115 // Destroys a video channel created with the Create API. |
112 void DestroyVideoChannel(VideoChannel* video_channel); | 116 void DestroyVideoChannel(VideoChannel* video_channel); |
113 RtpDataChannel* CreateRtpDataChannel( | 117 RtpDataChannel* CreateRtpDataChannel( |
114 webrtc::MediaControllerInterface* media_controller, | 118 webrtc::MediaControllerInterface* media_controller, |
115 TransportController* transport_controller, | 119 TransportChannel* rtp_transport, |
| 120 TransportChannel* rtcp_transport, |
| 121 rtc::Thread* signaling_thread, |
116 const std::string& content_name, | 122 const std::string& content_name, |
117 const std::string* bundle_transport_name, | 123 const std::string* bundle_transport_name, |
118 bool rtcp, | 124 bool rtcp, |
119 bool srtp_required); | 125 bool srtp_required); |
120 // Destroys a data channel created with the Create API. | 126 // Destroys a data channel created with the Create API. |
121 void DestroyRtpDataChannel(RtpDataChannel* data_channel); | 127 void DestroyRtpDataChannel(RtpDataChannel* data_channel); |
122 | 128 |
123 // Indicates whether any channels exist. | 129 // Indicates whether any channels exist. |
124 bool has_channels() const { | 130 bool has_channels() const { |
125 return (!voice_channels_.empty() || !video_channels_.empty()); | 131 return (!voice_channels_.empty() || !video_channels_.empty()); |
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154 void Construct(MediaEngineInterface* me, | 160 void Construct(MediaEngineInterface* me, |
155 DataEngineInterface* dme, | 161 DataEngineInterface* dme, |
156 rtc::Thread* worker_thread, | 162 rtc::Thread* worker_thread, |
157 rtc::Thread* network_thread); | 163 rtc::Thread* network_thread); |
158 bool InitMediaEngine_w(); | 164 bool InitMediaEngine_w(); |
159 void DestructorDeletes_w(); | 165 void DestructorDeletes_w(); |
160 void Terminate_w(); | 166 void Terminate_w(); |
161 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); | 167 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); |
162 VoiceChannel* CreateVoiceChannel_w( | 168 VoiceChannel* CreateVoiceChannel_w( |
163 webrtc::MediaControllerInterface* media_controller, | 169 webrtc::MediaControllerInterface* media_controller, |
164 TransportController* transport_controller, | 170 TransportChannel* rtp_transport, |
| 171 TransportChannel* rtcp_transport, |
| 172 rtc::Thread* signaling_thread, |
165 const std::string& content_name, | 173 const std::string& content_name, |
166 const std::string* bundle_transport_name, | 174 const std::string* bundle_transport_name, |
167 bool rtcp, | 175 bool rtcp, |
168 bool srtp_required, | 176 bool srtp_required, |
169 const AudioOptions& options); | 177 const AudioOptions& options); |
170 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); | 178 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); |
171 VideoChannel* CreateVideoChannel_w( | 179 VideoChannel* CreateVideoChannel_w( |
172 webrtc::MediaControllerInterface* media_controller, | 180 webrtc::MediaControllerInterface* media_controller, |
173 TransportController* transport_controller, | 181 TransportChannel* rtp_transport, |
| 182 TransportChannel* rtcp_transport, |
| 183 rtc::Thread* signaling_thread, |
174 const std::string& content_name, | 184 const std::string& content_name, |
175 const std::string* bundle_transport_name, | 185 const std::string* bundle_transport_name, |
176 bool rtcp, | 186 bool rtcp, |
177 bool srtp_required, | 187 bool srtp_required, |
178 const VideoOptions& options); | 188 const VideoOptions& options); |
179 void DestroyVideoChannel_w(VideoChannel* video_channel); | 189 void DestroyVideoChannel_w(VideoChannel* video_channel); |
180 RtpDataChannel* CreateRtpDataChannel_w( | 190 RtpDataChannel* CreateRtpDataChannel_w( |
181 webrtc::MediaControllerInterface* media_controller, | 191 webrtc::MediaControllerInterface* media_controller, |
182 TransportController* transport_controller, | 192 TransportChannel* rtp_transport, |
| 193 TransportChannel* rtcp_transport, |
| 194 rtc::Thread* signaling_thread, |
183 const std::string& content_name, | 195 const std::string& content_name, |
184 const std::string* bundle_transport_name, | 196 const std::string* bundle_transport_name, |
185 bool rtcp, | 197 bool rtcp, |
186 bool srtp_required); | 198 bool srtp_required); |
187 void DestroyRtpDataChannel_w(RtpDataChannel* data_channel); | 199 void DestroyRtpDataChannel_w(RtpDataChannel* data_channel); |
188 | 200 |
189 std::unique_ptr<MediaEngineInterface> media_engine_; | 201 std::unique_ptr<MediaEngineInterface> media_engine_; |
190 std::unique_ptr<DataEngineInterface> data_media_engine_; | 202 std::unique_ptr<DataEngineInterface> data_media_engine_; |
191 bool initialized_; | 203 bool initialized_; |
192 rtc::Thread* main_thread_; | 204 rtc::Thread* main_thread_; |
193 rtc::Thread* worker_thread_; | 205 rtc::Thread* worker_thread_; |
194 rtc::Thread* network_thread_; | 206 rtc::Thread* network_thread_; |
195 | 207 |
196 VoiceChannels voice_channels_; | 208 VoiceChannels voice_channels_; |
197 VideoChannels video_channels_; | 209 VideoChannels video_channels_; |
198 RtpDataChannels data_channels_; | 210 RtpDataChannels data_channels_; |
199 | 211 |
200 bool enable_rtx_; | 212 bool enable_rtx_; |
201 rtc::CryptoOptions crypto_options_; | 213 rtc::CryptoOptions crypto_options_; |
202 | 214 |
203 bool capturing_; | 215 bool capturing_; |
204 }; | 216 }; |
205 | 217 |
206 } // namespace cricket | 218 } // namespace cricket |
207 | 219 |
208 #endif // WEBRTC_PC_CHANNELMANAGER_H_ | 220 #endif // WEBRTC_PC_CHANNELMANAGER_H_ |
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