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Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.h

Issue 1844773002: Update the call when the network route changes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Merge with head Created 4 years, 8 months ago
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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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