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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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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 bool srtp_required, | 96 bool srtp_required, |
97 const AudioOptions& options); | 97 const AudioOptions& options); |
| 98 // Version of the above that takes PacketTransportInternal. |
| 99 VoiceChannel* CreateVoiceChannel( |
| 100 webrtc::MediaControllerInterface* media_controller, |
| 101 rtc::PacketTransportInternal* rtp_transport, |
| 102 rtc::PacketTransportInternal* rtcp_transport, |
| 103 rtc::Thread* signaling_thread, |
| 104 const std::string& content_name, |
| 105 bool srtp_required, |
| 106 const AudioOptions& options); |
98 // Destroys a voice channel created with the Create API. | 107 // Destroys a voice channel created with the Create API. |
99 void DestroyVoiceChannel(VoiceChannel* voice_channel); | 108 void DestroyVoiceChannel(VoiceChannel* voice_channel); |
100 // Creates a video channel, synced with the specified voice channel, and | 109 // Creates a video channel, synced with the specified voice channel, and |
101 // associated with the specified session. | 110 // associated with the specified session. |
102 VideoChannel* CreateVideoChannel( | 111 VideoChannel* CreateVideoChannel( |
103 webrtc::MediaControllerInterface* media_controller, | 112 webrtc::MediaControllerInterface* media_controller, |
104 DtlsTransportInternal* rtp_transport, | 113 DtlsTransportInternal* rtp_transport, |
105 DtlsTransportInternal* rtcp_transport, | 114 DtlsTransportInternal* rtcp_transport, |
106 rtc::Thread* signaling_thread, | 115 rtc::Thread* signaling_thread, |
107 const std::string& content_name, | 116 const std::string& content_name, |
108 bool srtp_required, | 117 bool srtp_required, |
109 const VideoOptions& options); | 118 const VideoOptions& options); |
| 119 // Version of the above that takes PacketTransportInternal. |
| 120 VideoChannel* CreateVideoChannel( |
| 121 webrtc::MediaControllerInterface* media_controller, |
| 122 rtc::PacketTransportInternal* rtp_transport, |
| 123 rtc::PacketTransportInternal* rtcp_transport, |
| 124 rtc::Thread* signaling_thread, |
| 125 const std::string& content_name, |
| 126 bool srtp_required, |
| 127 const VideoOptions& options); |
110 // Destroys a video channel created with the Create API. | 128 // Destroys a video channel created with the Create API. |
111 void DestroyVideoChannel(VideoChannel* video_channel); | 129 void DestroyVideoChannel(VideoChannel* video_channel); |
112 RtpDataChannel* CreateRtpDataChannel( | 130 RtpDataChannel* CreateRtpDataChannel( |
113 webrtc::MediaControllerInterface* media_controller, | 131 webrtc::MediaControllerInterface* media_controller, |
114 DtlsTransportInternal* rtp_transport, | 132 DtlsTransportInternal* rtp_transport, |
115 DtlsTransportInternal* rtcp_transport, | 133 DtlsTransportInternal* rtcp_transport, |
116 rtc::Thread* signaling_thread, | 134 rtc::Thread* signaling_thread, |
117 const std::string& content_name, | 135 const std::string& content_name, |
118 bool srtp_required); | 136 bool srtp_required); |
119 // Destroys a data channel created with the Create API. | 137 // Destroys a data channel created with the Create API. |
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153 void Construct(std::unique_ptr<MediaEngineInterface> me, | 171 void Construct(std::unique_ptr<MediaEngineInterface> me, |
154 std::unique_ptr<DataEngineInterface> dme, | 172 std::unique_ptr<DataEngineInterface> dme, |
155 rtc::Thread* worker_thread, | 173 rtc::Thread* worker_thread, |
156 rtc::Thread* network_thread); | 174 rtc::Thread* network_thread); |
157 bool InitMediaEngine_w(); | 175 bool InitMediaEngine_w(); |
158 void DestructorDeletes_w(); | 176 void DestructorDeletes_w(); |
159 void Terminate_w(); | 177 void Terminate_w(); |
160 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); | 178 bool SetCryptoOptions_w(const rtc::CryptoOptions& crypto_options); |
161 VoiceChannel* CreateVoiceChannel_w( | 179 VoiceChannel* CreateVoiceChannel_w( |
162 webrtc::MediaControllerInterface* media_controller, | 180 webrtc::MediaControllerInterface* media_controller, |
163 DtlsTransportInternal* rtp_transport, | 181 DtlsTransportInternal* rtp_dtls_transport, |
164 DtlsTransportInternal* rtcp_transport, | 182 DtlsTransportInternal* rtcp_dtls_transport, |
| 183 rtc::PacketTransportInternal* rtp_packet_transport, |
| 184 rtc::PacketTransportInternal* rtcp_packet_transport, |
165 rtc::Thread* signaling_thread, | 185 rtc::Thread* signaling_thread, |
166 const std::string& content_name, | 186 const std::string& content_name, |
167 bool srtp_required, | 187 bool srtp_required, |
168 const AudioOptions& options); | 188 const AudioOptions& options); |
169 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); | 189 void DestroyVoiceChannel_w(VoiceChannel* voice_channel); |
170 VideoChannel* CreateVideoChannel_w( | 190 VideoChannel* CreateVideoChannel_w( |
171 webrtc::MediaControllerInterface* media_controller, | 191 webrtc::MediaControllerInterface* media_controller, |
172 DtlsTransportInternal* rtp_transport, | 192 DtlsTransportInternal* rtp_dtls_transport, |
173 DtlsTransportInternal* rtcp_transport, | 193 DtlsTransportInternal* rtcp_dtls_transport, |
| 194 rtc::PacketTransportInternal* rtp_packet_transport, |
| 195 rtc::PacketTransportInternal* rtcp_packet_transport, |
174 rtc::Thread* signaling_thread, | 196 rtc::Thread* signaling_thread, |
175 const std::string& content_name, | 197 const std::string& content_name, |
176 bool srtp_required, | 198 bool srtp_required, |
177 const VideoOptions& options); | 199 const VideoOptions& options); |
178 void DestroyVideoChannel_w(VideoChannel* video_channel); | 200 void DestroyVideoChannel_w(VideoChannel* video_channel); |
179 RtpDataChannel* CreateRtpDataChannel_w( | 201 RtpDataChannel* CreateRtpDataChannel_w( |
180 webrtc::MediaControllerInterface* media_controller, | 202 webrtc::MediaControllerInterface* media_controller, |
181 DtlsTransportInternal* rtp_transport, | 203 DtlsTransportInternal* rtp_transport, |
182 DtlsTransportInternal* rtcp_transport, | 204 DtlsTransportInternal* rtcp_transport, |
183 rtc::Thread* signaling_thread, | 205 rtc::Thread* signaling_thread, |
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198 | 220 |
199 bool enable_rtx_; | 221 bool enable_rtx_; |
200 rtc::CryptoOptions crypto_options_; | 222 rtc::CryptoOptions crypto_options_; |
201 | 223 |
202 bool capturing_; | 224 bool capturing_; |
203 }; | 225 }; |
204 | 226 |
205 } // namespace cricket | 227 } // namespace cricket |
206 | 228 |
207 #endif // WEBRTC_PC_CHANNELMANAGER_H_ | 229 #endif // WEBRTC_PC_CHANNELMANAGER_H_ |
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