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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 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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388 const rtc::PacketTime& packet_time) = 0; | 388 const rtc::PacketTime& packet_time) = 0; |
389 // Called when a RTCP packet is received. | 389 // Called when a RTCP packet is received. |
390 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 390 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
391 const rtc::PacketTime& packet_time) = 0; | 391 const rtc::PacketTime& packet_time) = 0; |
392 // Called when the socket's ability to send has changed. | 392 // Called when the socket's ability to send has changed. |
393 virtual void OnReadyToSend(bool ready) = 0; | 393 virtual void OnReadyToSend(bool ready) = 0; |
394 // Called when the network route used for sending packets changed. | 394 // Called when the network route used for sending packets changed. |
395 virtual void OnNetworkRouteChanged( | 395 virtual void OnNetworkRouteChanged( |
396 const std::string& transport_name, | 396 const std::string& transport_name, |
397 const rtc::NetworkRoute& network_route) = 0; | 397 const rtc::NetworkRoute& network_route) = 0; |
| 398 // Called when the rtp transport overhead changed. |
| 399 virtual void OnTransportOverheadChange(int transport_overhead_per_packet) = 0; |
398 // Creates a new outgoing media stream with SSRCs and CNAME as described | 400 // Creates a new outgoing media stream with SSRCs and CNAME as described |
399 // by sp. | 401 // by sp. |
400 virtual bool AddSendStream(const StreamParams& sp) = 0; | 402 virtual bool AddSendStream(const StreamParams& sp) = 0; |
401 // Removes an outgoing media stream. | 403 // Removes an outgoing media stream. |
402 // ssrc must be the first SSRC of the media stream if the stream uses | 404 // ssrc must be the first SSRC of the media stream if the stream uses |
403 // multiple SSRCs. | 405 // multiple SSRCs. |
404 virtual bool RemoveSendStream(uint32_t ssrc) = 0; | 406 virtual bool RemoveSendStream(uint32_t ssrc) = 0; |
405 // Creates a new incoming media stream with SSRCs and CNAME as described | 407 // Creates a new incoming media stream with SSRCs and CNAME as described |
406 // by sp. | 408 // by sp. |
407 virtual bool AddRecvStream(const StreamParams& sp) = 0; | 409 virtual bool AddRecvStream(const StreamParams& sp) = 0; |
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1142 // Signal when the media channel is ready to send the stream. Arguments are: | 1144 // Signal when the media channel is ready to send the stream. Arguments are: |
1143 // writable(bool) | 1145 // writable(bool) |
1144 sigslot::signal1<bool> SignalReadyToSend; | 1146 sigslot::signal1<bool> SignalReadyToSend; |
1145 // Signal for notifying that the remote side has closed the DataChannel. | 1147 // Signal for notifying that the remote side has closed the DataChannel. |
1146 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1148 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1147 }; | 1149 }; |
1148 | 1150 |
1149 } // namespace cricket | 1151 } // namespace cricket |
1150 | 1152 |
1151 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1153 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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