Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(635)

Side by Side Diff: webrtc/media/base/mediachannel.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/media/base/fakemediaengine.h ('k') | webrtc/media/engine/fakewebrtccall.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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
(...skipping 374 matching lines...) Expand 10 before | Expand all | Expand 10 after
385 } 385 }
386 // Called when a RTP packet is received. 386 // Called when a RTP packet is received.
387 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet, 387 virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
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(const std::string& transport_name, 395 virtual void OnNetworkRouteChanged(
396 const NetworkRoute& network_route) = 0; 396 const std::string& transport_name,
397 const rtc::NetworkRoute& network_route) = 0;
397 // Creates a new outgoing media stream with SSRCs and CNAME as described 398 // Creates a new outgoing media stream with SSRCs and CNAME as described
398 // by sp. 399 // by sp.
399 virtual bool AddSendStream(const StreamParams& sp) = 0; 400 virtual bool AddSendStream(const StreamParams& sp) = 0;
400 // Removes an outgoing media stream. 401 // Removes an outgoing media stream.
401 // ssrc must be the first SSRC of the media stream if the stream uses 402 // ssrc must be the first SSRC of the media stream if the stream uses
402 // multiple SSRCs. 403 // multiple SSRCs.
403 virtual bool RemoveSendStream(uint32_t ssrc) = 0; 404 virtual bool RemoveSendStream(uint32_t ssrc) = 0;
404 // Creates a new incoming media stream with SSRCs and CNAME as described 405 // Creates a new incoming media stream with SSRCs and CNAME as described
405 // by sp. 406 // by sp.
406 virtual bool AddRecvStream(const StreamParams& sp) = 0; 407 virtual bool AddRecvStream(const StreamParams& sp) = 0;
(...skipping 698 matching lines...) Expand 10 before | Expand all | Expand 10 after
1105 virtual bool SetSendParameters(const DataSendParameters& params) = 0; 1106 virtual bool SetSendParameters(const DataSendParameters& params) = 0;
1106 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0; 1107 virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
1107 1108
1108 // TODO(pthatcher): Implement this. 1109 // TODO(pthatcher): Implement this.
1109 virtual bool GetStats(DataMediaInfo* info) { return true; } 1110 virtual bool GetStats(DataMediaInfo* info) { return true; }
1110 1111
1111 virtual bool SetSend(bool send) = 0; 1112 virtual bool SetSend(bool send) = 0;
1112 virtual bool SetReceive(bool receive) = 0; 1113 virtual bool SetReceive(bool receive) = 0;
1113 1114
1114 virtual void OnNetworkRouteChanged(const std::string& transport_name, 1115 virtual void OnNetworkRouteChanged(const std::string& transport_name,
1115 const NetworkRoute& network_route) {} 1116 const rtc::NetworkRoute& network_route) {}
1116 1117
1117 virtual bool SendData( 1118 virtual bool SendData(
1118 const SendDataParams& params, 1119 const SendDataParams& params,
1119 const rtc::CopyOnWriteBuffer& payload, 1120 const rtc::CopyOnWriteBuffer& payload,
1120 SendDataResult* result = NULL) = 0; 1121 SendDataResult* result = NULL) = 0;
1121 // Signals when data is received (params, data, len) 1122 // Signals when data is received (params, data, len)
1122 sigslot::signal3<const ReceiveDataParams&, 1123 sigslot::signal3<const ReceiveDataParams&,
1123 const char*, 1124 const char*,
1124 size_t> SignalDataReceived; 1125 size_t> SignalDataReceived;
1125 // Signal when the media channel is ready to send the stream. Arguments are: 1126 // Signal when the media channel is ready to send the stream. Arguments are:
1126 // writable(bool) 1127 // writable(bool)
1127 sigslot::signal1<bool> SignalReadyToSend; 1128 sigslot::signal1<bool> SignalReadyToSend;
1128 // Signal for notifying that the remote side has closed the DataChannel. 1129 // Signal for notifying that the remote side has closed the DataChannel.
1129 sigslot::signal1<uint32_t> SignalStreamClosedRemotely; 1130 sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
1130 }; 1131 };
1131 1132
1132 } // namespace cricket 1133 } // namespace cricket
1133 1134
1134 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_ 1135 #endif // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
OLDNEW
« no previous file with comments | « webrtc/media/base/fakemediaengine.h ('k') | webrtc/media/engine/fakewebrtccall.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698