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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 |
11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 12 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/api/rtpparameters.h" | 18 #include "webrtc/api/rtpparameters.h" |
19 #include "webrtc/base/basictypes.h" | 19 #include "webrtc/base/basictypes.h" |
20 #include "webrtc/base/copyonwritebuffer.h" | 20 #include "webrtc/base/copyonwritebuffer.h" |
21 #include "webrtc/base/dscp.h" | 21 #include "webrtc/base/dscp.h" |
22 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
| 23 #include "webrtc/base/networkroute.h" |
23 #include "webrtc/base/optional.h" | 24 #include "webrtc/base/optional.h" |
24 #include "webrtc/base/sigslot.h" | 25 #include "webrtc/base/sigslot.h" |
25 #include "webrtc/base/socket.h" | 26 #include "webrtc/base/socket.h" |
26 #include "webrtc/base/window.h" | 27 #include "webrtc/base/window.h" |
27 #include "webrtc/media/base/codec.h" | 28 #include "webrtc/media/base/codec.h" |
28 #include "webrtc/media/base/mediaconstants.h" | 29 #include "webrtc/media/base/mediaconstants.h" |
29 #include "webrtc/media/base/streamparams.h" | 30 #include "webrtc/media/base/streamparams.h" |
30 #include "webrtc/media/base/videosinkinterface.h" | 31 #include "webrtc/media/base/videosinkinterface.h" |
31 // TODO(juberti): re-evaluate this include | 32 // TODO(juberti): re-evaluate this include |
32 #include "webrtc/pc/audiomonitor.h" | 33 #include "webrtc/pc/audiomonitor.h" |
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380 return rtc::DSCP_DEFAULT; | 381 return rtc::DSCP_DEFAULT; |
381 } | 382 } |
382 // Called when a RTP packet is received. | 383 // Called when a RTP packet is received. |
383 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, | 384 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, |
384 const rtc::PacketTime& packet_time) = 0; | 385 const rtc::PacketTime& packet_time) = 0; |
385 // Called when a RTCP packet is received. | 386 // Called when a RTCP packet is received. |
386 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, | 387 virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet, |
387 const rtc::PacketTime& packet_time) = 0; | 388 const rtc::PacketTime& packet_time) = 0; |
388 // Called when the socket's ability to send has changed. | 389 // Called when the socket's ability to send has changed. |
389 virtual void OnReadyToSend(bool ready) = 0; | 390 virtual void OnReadyToSend(bool ready) = 0; |
| 391 // Called when the network route used for sending packets changed. |
| 392 virtual void OnNetworkRouteChanged(const std::string& transport_name, |
| 393 const NetworkRoute& network_route) = 0; |
390 // Creates a new outgoing media stream with SSRCs and CNAME as described | 394 // Creates a new outgoing media stream with SSRCs and CNAME as described |
391 // by sp. | 395 // by sp. |
392 virtual bool AddSendStream(const StreamParams& sp) = 0; | 396 virtual bool AddSendStream(const StreamParams& sp) = 0; |
393 // Removes an outgoing media stream. | 397 // Removes an outgoing media stream. |
394 // ssrc must be the first SSRC of the media stream if the stream uses | 398 // ssrc must be the first SSRC of the media stream if the stream uses |
395 // multiple SSRCs. | 399 // multiple SSRCs. |
396 virtual bool RemoveSendStream(uint32_t ssrc) = 0; | 400 virtual bool RemoveSendStream(uint32_t ssrc) = 0; |
397 // Creates a new incoming media stream with SSRCs and CNAME as described | 401 // Creates a new incoming media stream with SSRCs and CNAME as described |
398 // by sp. | 402 // by sp. |
399 virtual bool AddRecvStream(const StreamParams& sp) = 0; | 403 virtual bool AddRecvStream(const StreamParams& sp) = 0; |
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1097 | 1101 |
1098 virtual bool SetSendParameters(const DataSendParameters& params) = 0; | 1102 virtual bool SetSendParameters(const DataSendParameters& params) = 0; |
1099 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; | 1103 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; |
1100 | 1104 |
1101 // TODO(pthatcher): Implement this. | 1105 // TODO(pthatcher): Implement this. |
1102 virtual bool GetStats(DataMediaInfo* info) { return true; } | 1106 virtual bool GetStats(DataMediaInfo* info) { return true; } |
1103 | 1107 |
1104 virtual bool SetSend(bool send) = 0; | 1108 virtual bool SetSend(bool send) = 0; |
1105 virtual bool SetReceive(bool receive) = 0; | 1109 virtual bool SetReceive(bool receive) = 0; |
1106 | 1110 |
| 1111 virtual void OnNetworkRouteChanged(const std::string& transport_name, |
| 1112 const NetworkRoute& network_route) {} |
| 1113 |
1107 virtual bool SendData( | 1114 virtual bool SendData( |
1108 const SendDataParams& params, | 1115 const SendDataParams& params, |
1109 const rtc::CopyOnWriteBuffer& payload, | 1116 const rtc::CopyOnWriteBuffer& payload, |
1110 SendDataResult* result = NULL) = 0; | 1117 SendDataResult* result = NULL) = 0; |
1111 // Signals when data is received (params, data, len) | 1118 // Signals when data is received (params, data, len) |
1112 sigslot::signal3<const ReceiveDataParams&, | 1119 sigslot::signal3<const ReceiveDataParams&, |
1113 const char*, | 1120 const char*, |
1114 size_t> SignalDataReceived; | 1121 size_t> SignalDataReceived; |
1115 // Signal when the media channel is ready to send the stream. Arguments are: | 1122 // Signal when the media channel is ready to send the stream. Arguments are: |
1116 // writable(bool) | 1123 // writable(bool) |
1117 sigslot::signal1<bool> SignalReadyToSend; | 1124 sigslot::signal1<bool> SignalReadyToSend; |
1118 // Signal for notifying that the remote side has closed the DataChannel. | 1125 // Signal for notifying that the remote side has closed the DataChannel. |
1119 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; | 1126 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; |
1120 }; | 1127 }; |
1121 | 1128 |
1122 } // namespace cricket | 1129 } // namespace cricket |
1123 | 1130 |
1124 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ | 1131 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ |
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