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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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398 const rtc::PacketTime& packet_time) = 0; | 398 const rtc::PacketTime& packet_time) = 0; |
399 // Called when a RTCP packet is received. | 399 // Called when a RTCP packet is received. |
400 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 400 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
401 const rtc::PacketTime& packet_time) = 0; | 401 const rtc::PacketTime& packet_time) = 0; |
402 // Called when the socket's ability to send has changed. | 402 // Called when the socket's ability to send has changed. |
403 virtual void OnReadyToSend(bool ready) = 0; | 403 virtual void OnReadyToSend(bool ready) = 0; |
404 // Called when the network route used for sending packets changed. | 404 // Called when the network route used for sending packets changed. |
405 virtual void OnNetworkRouteChanged( | 405 virtual void OnNetworkRouteChanged( |
406 const std::string& transport_name, | 406 const std::string& transport_name, |
407 const rtc::NetworkRoute& network_route) = 0; | 407 const rtc::NetworkRoute& network_route) = 0; |
| 408 // Called when the rtp transport overhead changed. |
| 409 virtual void OnTransportOverheadChanged( |
| 410 int transport_overhead_per_packet) = 0; |
408 // Creates a new outgoing media stream with SSRCs and CNAME as described | 411 // Creates a new outgoing media stream with SSRCs and CNAME as described |
409 // by sp. | 412 // by sp. |
410 virtual bool AddSendStream(const StreamParams& sp) = 0; | 413 virtual bool AddSendStream(const StreamParams& sp) = 0; |
411 // Removes an outgoing media stream. | 414 // Removes an outgoing media stream. |
412 // ssrc must be the first SSRC of the media stream if the stream uses | 415 // ssrc must be the first SSRC of the media stream if the stream uses |
413 // multiple SSRCs. | 416 // multiple SSRCs. |
414 virtual bool RemoveSendStream(uint32_t ssrc) = 0; | 417 virtual bool RemoveSendStream(uint32_t ssrc) = 0; |
415 // Creates a new incoming media stream with SSRCs and CNAME as described | 418 // Creates a new incoming media stream with SSRCs and CNAME as described |
416 // by sp. | 419 // by sp. |
417 virtual bool AddRecvStream(const StreamParams& sp) = 0; | 420 virtual bool AddRecvStream(const StreamParams& sp) = 0; |
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1158 // Signal when the media channel is ready to send the stream. Arguments are: | 1161 // Signal when the media channel is ready to send the stream. Arguments are: |
1159 // writable(bool) | 1162 // writable(bool) |
1160 sigslot::signal1<bool> SignalReadyToSend; | 1163 sigslot::signal1<bool> SignalReadyToSend; |
1161 // Signal for notifying that the remote side has closed the DataChannel. | 1164 // Signal for notifying that the remote side has closed the DataChannel. |
1162 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1165 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1163 }; | 1166 }; |
1164 | 1167 |
1165 } // namespace cricket | 1168 } // namespace cricket |
1166 | 1169 |
1167 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1170 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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