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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #ifndef CALL_CALL_H_ | 10 #ifndef CALL_CALL_H_ |
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124 int64_t rtt_ms = -1; | 124 int64_t rtt_ms = -1; |
125 }; | 125 }; |
126 | 126 |
127 static Call* Create(const Call::Config& config); | 127 static Call* Create(const Call::Config& config); |
128 | 128 |
129 // Allows mocking |transport_send| for testing. | 129 // Allows mocking |transport_send| for testing. |
130 static Call* Create( | 130 static Call* Create( |
131 const Call::Config& config, | 131 const Call::Config& config, |
132 std::unique_ptr<RtpTransportControllerSendInterface> transport_send); | 132 std::unique_ptr<RtpTransportControllerSendInterface> transport_send); |
133 | 133 |
| 134 // TODO(nisse): Should move to RtpTransportController. |
| 135 // Rtp header extensions can be renegotiated mid-call. |
| 136 virtual void SetVideoReceiveRtpHeaderExtensions( |
| 137 const std::vector<RtpExtension>& extensions) = 0; |
| 138 |
134 virtual AudioSendStream* CreateAudioSendStream( | 139 virtual AudioSendStream* CreateAudioSendStream( |
135 const AudioSendStream::Config& config) = 0; | 140 const AudioSendStream::Config& config) = 0; |
136 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; | 141 virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; |
137 | 142 |
138 virtual AudioReceiveStream* CreateAudioReceiveStream( | 143 virtual AudioReceiveStream* CreateAudioReceiveStream( |
139 const AudioReceiveStream::Config& config) = 0; | 144 const AudioReceiveStream::Config& config) = 0; |
140 virtual void DestroyAudioReceiveStream( | 145 virtual void DestroyAudioReceiveStream( |
141 AudioReceiveStream* receive_stream) = 0; | 146 AudioReceiveStream* receive_stream) = 0; |
142 | 147 |
143 virtual VideoSendStream* CreateVideoSendStream( | 148 virtual VideoSendStream* CreateVideoSendStream( |
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198 const rtc::NetworkRoute& network_route) = 0; | 203 const rtc::NetworkRoute& network_route) = 0; |
199 | 204 |
200 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; | 205 virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; |
201 | 206 |
202 virtual ~Call() {} | 207 virtual ~Call() {} |
203 }; | 208 }; |
204 | 209 |
205 } // namespace webrtc | 210 } // namespace webrtc |
206 | 211 |
207 #endif // CALL_CALL_H_ | 212 #endif // CALL_CALL_H_ |
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