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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 |
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| 455 bool weak() const { return !(writable() && receiving() && connected()); } | 455 bool weak() const { return !(writable() && receiving() && connected()); } |
| 456 bool active() const { | 456 bool active() const { |
| 457 return write_state_ != STATE_WRITE_TIMEOUT; | 457 return write_state_ != STATE_WRITE_TIMEOUT; |
| 458 } | 458 } |
| 459 // A connection is dead if it can be safely deleted. | 459 // A connection is dead if it can be safely deleted. |
| 460 bool dead(int64_t now) const; | 460 bool dead(int64_t now) const; |
| 461 | 461 |
| 462 // Estimate of the round-trip time over this connection. | 462 // Estimate of the round-trip time over this connection. |
| 463 int rtt() const { return rtt_; } | 463 int rtt() const { return rtt_; } |
| 464 | 464 |
| 465 size_t sent_total_bytes(); | 465 ConnectionInfo stats(); |
| 466 size_t sent_bytes_second(); | 466 |
| 467 // Used to track how many packets are discarded in the application socket due | |
| 468 // to errors. | |
| 469 size_t sent_discarded_packets(); | |
| 470 size_t sent_total_packets(); | |
| 471 size_t recv_total_bytes(); | |
| 472 size_t recv_bytes_second(); | |
| 473 sigslot::signal1<Connection*> SignalStateChange; | 467 sigslot::signal1<Connection*> SignalStateChange; |
| 474 | 468 |
| 475 // Sent when the connection has decided that it is no longer of value. It | 469 // Sent when the connection has decided that it is no longer of value. It |
| 476 // will delete itself immediately after this call. | 470 // will delete itself immediately after this call. |
| 477 sigslot::signal1<Connection*> SignalDestroyed; | 471 sigslot::signal1<Connection*> SignalDestroyed; |
| 478 | 472 |
| 479 // The connection can send and receive packets asynchronously. This matches | 473 // The connection can send and receive packets asynchronously. This matches |
| 480 // the interface of AsyncPacketSocket, which may use UDP or TCP under the | 474 // the interface of AsyncPacketSocket, which may use UDP or TCP under the |
| 481 // covers. | 475 // covers. |
| 482 virtual int Send(const void* data, size_t size, | 476 virtual int Send(const void* data, size_t size, |
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| 631 int rtt_; | 625 int rtt_; |
| 632 int64_t last_ping_sent_; // last time we sent a ping to the other side | 626 int64_t last_ping_sent_; // last time we sent a ping to the other side |
| 633 int64_t last_ping_received_; // last time we received a ping from the other | 627 int64_t last_ping_received_; // last time we received a ping from the other |
| 634 // side | 628 // side |
| 635 int64_t last_data_received_; | 629 int64_t last_data_received_; |
| 636 int64_t last_ping_response_received_; | 630 int64_t last_ping_response_received_; |
| 637 std::vector<SentPing> pings_since_last_response_; | 631 std::vector<SentPing> pings_since_last_response_; |
| 638 | 632 |
| 639 rtc::RateTracker recv_rate_tracker_; | 633 rtc::RateTracker recv_rate_tracker_; |
| 640 rtc::RateTracker send_rate_tracker_; | 634 rtc::RateTracker send_rate_tracker_; |
| 641 uint32_t sent_packets_discarded_; | 635 |
| 642 uint32_t sent_packets_total_; | 636 ConnectionInfo stats_; |
| 643 | 637 |
| 644 private: | 638 private: |
| 645 void MaybeAddPrflxCandidate(ConnectionRequest* request, | 639 void MaybeAddPrflxCandidate(ConnectionRequest* request, |
| 646 StunMessage* response); | 640 StunMessage* response); |
| 647 | 641 |
| 648 bool reported_; | 642 bool reported_; |
| 649 State state_; | 643 State state_; |
| 650 // Time duration to switch from receiving to not receiving. | 644 // Time duration to switch from receiving to not receiving. |
| 651 int receiving_timeout_; | 645 int receiving_timeout_; |
| 652 int64_t time_created_ms_; | 646 int64_t time_created_ms_; |
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| 666 const rtc::PacketOptions& options) override; | 660 const rtc::PacketOptions& options) override; |
| 667 int GetError() override { return error_; } | 661 int GetError() override { return error_; } |
| 668 | 662 |
| 669 private: | 663 private: |
| 670 int error_ = 0; | 664 int error_ = 0; |
| 671 }; | 665 }; |
| 672 | 666 |
| 673 } // namespace cricket | 667 } // namespace cricket |
| 674 | 668 |
| 675 #endif // WEBRTC_P2P_BASE_PORT_H_ | 669 #endif // WEBRTC_P2P_BASE_PORT_H_ |
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