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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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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164 void SetSource( | 164 void SetSource( |
165 uint32_t ssrc, | 165 uint32_t ssrc, |
166 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override; | 166 rtc::VideoSourceInterface<cricket::VideoFrame>* source) override; |
167 | 167 |
168 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, | 168 void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
169 const rtc::PacketTime& packet_time) override; | 169 const rtc::PacketTime& packet_time) override; |
170 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 170 void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
171 const rtc::PacketTime& packet_time) override; | 171 const rtc::PacketTime& packet_time) override; |
172 void OnReadyToSend(bool ready) override; | 172 void OnReadyToSend(bool ready) override; |
173 void OnNetworkRouteChanged(const std::string& transport_name, | 173 void OnNetworkRouteChanged(const std::string& transport_name, |
174 const NetworkRoute& network_route) override; | 174 const rtc::NetworkRoute& network_route) override; |
175 void SetInterface(NetworkInterface* iface) override; | 175 void SetInterface(NetworkInterface* iface) override; |
176 | 176 |
177 // Implemented for VideoMediaChannelTest. | 177 // Implemented for VideoMediaChannelTest. |
178 bool sending() const { return sending_; } | 178 bool sending() const { return sending_; } |
179 | 179 |
180 // AdaptReason is used for expressing why a WebRtcVideoSendStream request | 180 // AdaptReason is used for expressing why a WebRtcVideoSendStream request |
181 // a lower input frame size than the currently configured camera input frame | 181 // a lower input frame size than the currently configured camera input frame |
182 // size. There can be more than one reason OR:ed together. | 182 // size. There can be more than one reason OR:ed together. |
183 enum AdaptReason { | 183 enum AdaptReason { |
184 ADAPTREASON_NONE = 0, | 184 ADAPTREASON_NONE = 0, |
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533 // TODO(deadbeef): Don't duplicate information between | 533 // TODO(deadbeef): Don't duplicate information between |
534 // send_params/recv_params, rtp_extensions, options, etc. | 534 // send_params/recv_params, rtp_extensions, options, etc. |
535 VideoSendParameters send_params_; | 535 VideoSendParameters send_params_; |
536 VideoOptions default_send_options_; | 536 VideoOptions default_send_options_; |
537 VideoRecvParameters recv_params_; | 537 VideoRecvParameters recv_params_; |
538 }; | 538 }; |
539 | 539 |
540 } // namespace cricket | 540 } // namespace cricket |
541 | 541 |
542 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ | 542 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ |
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