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Side by Side Diff: webrtc/p2p/base/transportchannel.h

Issue 1803063004: Reset the BWE when the network changes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Removed changes in call dir and leave that in a separate CL. Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright 2004 The WebRTC Project Authors. All rights reserved. 2 * Copyright 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_P2P_BASE_TRANSPORTCHANNEL_H_ 11 #ifndef WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
12 #define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_ 12 #define WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
13 13
14 #include <string> 14 #include <string>
15 #include <vector> 15 #include <vector>
16 16
17 #include "webrtc/p2p/base/candidate.h" 17 #include "webrtc/p2p/base/candidate.h"
18 #include "webrtc/p2p/base/candidatepairinterface.h"
18 #include "webrtc/p2p/base/transport.h" 19 #include "webrtc/p2p/base/transport.h"
19 #include "webrtc/p2p/base/transportdescription.h" 20 #include "webrtc/p2p/base/transportdescription.h"
20 #include "webrtc/base/asyncpacketsocket.h" 21 #include "webrtc/base/asyncpacketsocket.h"
21 #include "webrtc/base/basictypes.h" 22 #include "webrtc/base/basictypes.h"
22 #include "webrtc/base/dscp.h" 23 #include "webrtc/base/dscp.h"
23 #include "webrtc/base/sigslot.h" 24 #include "webrtc/base/sigslot.h"
24 #include "webrtc/base/socket.h" 25 #include "webrtc/base/socket.h"
25 #include "webrtc/base/sslidentity.h" 26 #include "webrtc/base/sslidentity.h"
26 #include "webrtc/base/sslstreamadapter.h" 27 #include "webrtc/base/sslstreamadapter.h"
27 28
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139 uint8_t* result, 140 uint8_t* result,
140 size_t result_len) = 0; 141 size_t result_len) = 0;
141 142
142 // Signalled each time a packet is received on this channel. 143 // Signalled each time a packet is received on this channel.
143 sigslot::signal5<TransportChannel*, const char*, 144 sigslot::signal5<TransportChannel*, const char*,
144 size_t, const rtc::PacketTime&, int> SignalReadPacket; 145 size_t, const rtc::PacketTime&, int> SignalReadPacket;
145 146
146 // Signalled each time a packet is sent on this channel. 147 // Signalled each time a packet is sent on this channel.
147 sigslot::signal2<TransportChannel*, const rtc::SentPacket&> SignalSentPacket; 148 sigslot::signal2<TransportChannel*, const rtc::SentPacket&> SignalSentPacket;
148 149
150 // Deprecated by SignalSelectedCandidatePairChanged
149 // This signal occurs when there is a change in the way that packets are 151 // This signal occurs when there is a change in the way that packets are
150 // being routed, i.e. to a different remote location. The candidate 152 // being routed, i.e. to a different remote location. The candidate
151 // indicates where and how we are currently sending media. 153 // indicates where and how we are currently sending media.
152 sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange; 154 sigslot::signal2<TransportChannel*, const Candidate&> SignalRouteChange;
153 155
156 // Signalled when the current selected candidate pair has changed.
157 // The first parameter is the transport channel that signals the event.
158 // The second parameter is the new selected candidate pair.
159 sigslot::signal2<TransportChannel*, CandidatePairInterface*>
160 SignalSelectedCandidatePairChanged;
161
154 // Invoked when the channel is being destroyed. 162 // Invoked when the channel is being destroyed.
155 sigslot::signal1<TransportChannel*> SignalDestroyed; 163 sigslot::signal1<TransportChannel*> SignalDestroyed;
156 164
157 // Debugging description of this transport channel. 165 // Debugging description of this transport channel.
158 std::string ToString() const; 166 std::string ToString() const;
159 167
160 protected: 168 protected:
161 // Sets the writable state, signaling if necessary. 169 // Sets the writable state, signaling if necessary.
162 void set_writable(bool writable); 170 void set_writable(bool writable);
163 171
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174 bool writable_; 182 bool writable_;
175 bool receiving_; 183 bool receiving_;
176 DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW; 184 DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
177 185
178 RTC_DISALLOW_COPY_AND_ASSIGN(TransportChannel); 186 RTC_DISALLOW_COPY_AND_ASSIGN(TransportChannel);
179 }; 187 };
180 188
181 } // namespace cricket 189 } // namespace cricket
182 190
183 #endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_ 191 #endif // WEBRTC_P2P_BASE_TRANSPORTCHANNEL_H_
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