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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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86 void Terminate(); | 86 void Terminate(); |
87 | 87 |
88 // The operations below all occur on the worker thread. | 88 // The operations below all occur on the worker thread. |
89 // Creates a voice channel, to be associated with the specified session. | 89 // Creates a voice channel, to be associated with the specified session. |
90 VoiceChannel* CreateVoiceChannel( | 90 VoiceChannel* CreateVoiceChannel( |
91 webrtc::MediaControllerInterface* media_controller, | 91 webrtc::MediaControllerInterface* media_controller, |
92 DtlsTransportInternal* rtp_transport, | 92 DtlsTransportInternal* rtp_transport, |
93 DtlsTransportInternal* rtcp_transport, | 93 DtlsTransportInternal* rtcp_transport, |
94 rtc::Thread* signaling_thread, | 94 rtc::Thread* signaling_thread, |
95 const std::string& content_name, | 95 const std::string& content_name, |
96 const std::string* bundle_transport_name, | |
97 bool rtcp_mux_required, | |
98 bool srtp_required, | 96 bool srtp_required, |
99 const AudioOptions& options); | 97 const AudioOptions& options); |
100 // Destroys a voice channel created with the Create API. | 98 // Destroys a voice channel created with the Create API. |
101 void DestroyVoiceChannel(VoiceChannel* voice_channel); | 99 void DestroyVoiceChannel(VoiceChannel* voice_channel); |
102 // Creates a video channel, synced with the specified voice channel, and | 100 // Creates a video channel, synced with the specified voice channel, and |
103 // associated with the specified session. | 101 // associated with the specified session. |
104 VideoChannel* CreateVideoChannel( | 102 VideoChannel* CreateVideoChannel( |
105 webrtc::MediaControllerInterface* media_controller, | 103 webrtc::MediaControllerInterface* media_controller, |
106 DtlsTransportInternal* rtp_transport, | 104 DtlsTransportInternal* rtp_transport, |
107 DtlsTransportInternal* rtcp_transport, | 105 DtlsTransportInternal* rtcp_transport, |
108 rtc::Thread* signaling_thread, | 106 rtc::Thread* signaling_thread, |
109 const std::string& content_name, | 107 const std::string& content_name, |
110 const std::string* bundle_transport_name, | |
111 bool rtcp_mux_required, | |
112 bool srtp_required, | 108 bool srtp_required, |
113 const VideoOptions& options); | 109 const VideoOptions& options); |
114 // Destroys a video channel created with the Create API. | 110 // Destroys a video channel created with the Create API. |
115 void DestroyVideoChannel(VideoChannel* video_channel); | 111 void DestroyVideoChannel(VideoChannel* video_channel); |
116 RtpDataChannel* CreateRtpDataChannel( | 112 RtpDataChannel* CreateRtpDataChannel( |
117 webrtc::MediaControllerInterface* media_controller, | 113 webrtc::MediaControllerInterface* media_controller, |
118 DtlsTransportInternal* rtp_transport, | 114 DtlsTransportInternal* rtp_transport, |
119 DtlsTransportInternal* rtcp_transport, | 115 DtlsTransportInternal* rtcp_transport, |
120 rtc::Thread* signaling_thread, | 116 rtc::Thread* signaling_thread, |
121 const std::string& content_name, | 117 const std::string& content_name, |
122 const std::string* bundle_transport_name, | |
123 bool rtcp_mux_required, | |
124 bool srtp_required); | 118 bool srtp_required); |
125 // Destroys a data channel created with the Create API. | 119 // Destroys a data channel created with the Create API. |
126 void DestroyRtpDataChannel(RtpDataChannel* data_channel); | 120 void DestroyRtpDataChannel(RtpDataChannel* data_channel); |
127 | 121 |
128 // Indicates whether any channels exist. | 122 // Indicates whether any channels exist. |
129 bool has_channels() const { | 123 bool has_channels() const { |
130 return (!voice_channels_.empty() || !video_channels_.empty()); | 124 return (!voice_channels_.empty() || !video_channels_.empty()); |
131 } | 125 } |
132 | 126 |
133 // RTX will be enabled/disabled in engines that support it. The supporting | 127 // RTX will be enabled/disabled in engines that support it. The supporting |
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163 bool InitMediaEngine_w(); | 157 bool InitMediaEngine_w(); |
164 void DestructorDeletes_w(); | 158 void DestructorDeletes_w(); |
165 void Terminate_w(); | 159 void Terminate_w(); |
166 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); | 160 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); |
167 VoiceChannel* CreateVoiceChannel_w( | 161 VoiceChannel* CreateVoiceChannel_w( |
168 webrtc::MediaControllerInterface* media_controller, | 162 webrtc::MediaControllerInterface* media_controller, |
169 DtlsTransportInternal* rtp_transport, | 163 DtlsTransportInternal* rtp_transport, |
170 DtlsTransportInternal* rtcp_transport, | 164 DtlsTransportInternal* rtcp_transport, |
171 rtc::Thread* signaling_thread, | 165 rtc::Thread* signaling_thread, |
172 const std::string& content_name, | 166 const std::string& content_name, |
173 const std::string* bundle_transport_name, | |
174 bool rtcp_mux_required, | |
175 bool srtp_required, | 167 bool srtp_required, |
176 const AudioOptions& options); | 168 const AudioOptions& options); |
177 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); | 169 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); |
178 VideoChannel* CreateVideoChannel_w( | 170 VideoChannel* CreateVideoChannel_w( |
179 webrtc::MediaControllerInterface* media_controller, | 171 webrtc::MediaControllerInterface* media_controller, |
180 DtlsTransportInternal* rtp_transport, | 172 DtlsTransportInternal* rtp_transport, |
181 DtlsTransportInternal* rtcp_transport, | 173 DtlsTransportInternal* rtcp_transport, |
182 rtc::Thread* signaling_thread, | 174 rtc::Thread* signaling_thread, |
183 const std::string& content_name, | 175 const std::string& content_name, |
184 const std::string* bundle_transport_name, | |
185 bool rtcp_mux_required, | |
186 bool srtp_required, | 176 bool srtp_required, |
187 const VideoOptions& options); | 177 const VideoOptions& options); |
188 void DestroyVideoChannel_w(VideoChannel* video_channel); | 178 void DestroyVideoChannel_w(VideoChannel* video_channel); |
189 RtpDataChannel* CreateRtpDataChannel_w( | 179 RtpDataChannel* CreateRtpDataChannel_w( |
190 webrtc::MediaControllerInterface* media_controller, | 180 webrtc::MediaControllerInterface* media_controller, |
191 DtlsTransportInternal* rtp_transport, | 181 DtlsTransportInternal* rtp_transport, |
192 DtlsTransportInternal* rtcp_transport, | 182 DtlsTransportInternal* rtcp_transport, |
193 rtc::Thread* signaling_thread, | 183 rtc::Thread* signaling_thread, |
194 const std::string& content_name, | 184 const std::string& content_name, |
195 const std::string* bundle_transport_name, | |
196 bool rtcp_mux_required, | |
197 bool srtp_required); | 185 bool srtp_required); |
198 void DestroyRtpDataChannel_w(RtpDataChannel* data_channel); | 186 void DestroyRtpDataChannel_w(RtpDataChannel* data_channel); |
199 | 187 |
200 std::unique_ptr<MediaEngineInterface> media_engine_; | 188 std::unique_ptr<MediaEngineInterface> media_engine_; |
201 std::unique_ptr<DataEngineInterface> data_media_engine_; | 189 std::unique_ptr<DataEngineInterface> data_media_engine_; |
202 bool initialized_; | 190 bool initialized_; |
203 rtc::Thread* main_thread_; | 191 rtc::Thread* main_thread_; |
204 rtc::Thread* worker_thread_; | 192 rtc::Thread* worker_thread_; |
205 rtc::Thread* network_thread_; | 193 rtc::Thread* network_thread_; |
206 | 194 |
207 VoiceChannels voice_channels_; | 195 VoiceChannels voice_channels_; |
208 VideoChannels video_channels_; | 196 VideoChannels video_channels_; |
209 RtpDataChannels data_channels_; | 197 RtpDataChannels data_channels_; |
210 | 198 |
211 bool enable_rtx_; | 199 bool enable_rtx_; |
212 rtc::CryptoOptions crypto_options_; | 200 rtc::CryptoOptions crypto_options_; |
213 | 201 |
214 bool capturing_; | 202 bool capturing_; |
215 }; | 203 }; |
216 | 204 |
217 } // namespace cricket | 205 } // namespace cricket |
218 | 206 |
219 #endif // WEBRTC_PC_CHANNELMANAGER_H_ | 207 #endif // WEBRTC_PC_CHANNELMANAGER_H_ |
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