OLD | NEW |
1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2012 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 471 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
482 // Returns the current SignalingState. | 482 // Returns the current SignalingState. |
483 virtual SignalingState signaling_state() = 0; | 483 virtual SignalingState signaling_state() = 0; |
484 | 484 |
485 // TODO(bemasc): Remove ice_state when callers are changed to | 485 // TODO(bemasc): Remove ice_state when callers are changed to |
486 // IceConnection/GatheringState. | 486 // IceConnection/GatheringState. |
487 // Returns the current IceState. | 487 // Returns the current IceState. |
488 virtual IceState ice_state() = 0; | 488 virtual IceState ice_state() = 0; |
489 virtual IceConnectionState ice_connection_state() = 0; | 489 virtual IceConnectionState ice_connection_state() = 0; |
490 virtual IceGatheringState ice_gathering_state() = 0; | 490 virtual IceGatheringState ice_gathering_state() = 0; |
491 | 491 |
492 // Starts RtcEventLog using existing file. Takes ownership of |file| and | |
493 // passes it on to Call, which will take the ownership. If the | |
494 // operation fails the file will be closed. The logging will stop | |
495 // automatically after 10 minutes have passed, or when the StopRtcEventLog | |
496 // function is called. | |
497 virtual bool StartRtcEventLog(rtc::PlatformFile file, | |
498 int64_t max_size_bytes) = 0; | |
499 | |
500 // Stops logging the RtcEventLog. | |
501 virtual void StopRtcEventLog() = 0; | |
502 | |
503 // Terminates all media and closes the transport. | 492 // Terminates all media and closes the transport. |
504 virtual void Close() = 0; | 493 virtual void Close() = 0; |
505 | 494 |
506 protected: | 495 protected: |
507 // Dtor protected as objects shouldn't be deleted via this interface. | 496 // Dtor protected as objects shouldn't be deleted via this interface. |
508 ~PeerConnectionInterface() {} | 497 ~PeerConnectionInterface() {} |
509 }; | 498 }; |
510 | 499 |
511 // PeerConnection callback interface. Application should implement these | 500 // PeerConnection callback interface. Application should implement these |
512 // methods. | 501 // methods. |
(...skipping 146 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
659 // the ownerhip. If the operation fails, the file will be closed. | 648 // the ownerhip. If the operation fails, the file will be closed. |
660 // A maximum file size in bytes can be specified. When the file size limit is | 649 // A maximum file size in bytes can be specified. When the file size limit is |
661 // reached, logging is stopped automatically. If max_size_bytes is set to a | 650 // reached, logging is stopped automatically. If max_size_bytes is set to a |
662 // value <= 0, no limit will be used, and logging will continue until the | 651 // value <= 0, no limit will be used, and logging will continue until the |
663 // StopAecDump function is called. | 652 // StopAecDump function is called. |
664 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; | 653 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; |
665 | 654 |
666 // Stops logging the AEC dump. | 655 // Stops logging the AEC dump. |
667 virtual void StopAecDump() = 0; | 656 virtual void StopAecDump() = 0; |
668 | 657 |
669 // This function is deprecated and will be removed when Chrome is updated to | 658 // Starts RtcEventLog using existing file. Takes ownership of |file| and |
670 // use the equivalent function on PeerConnectionInterface. | 659 // passes it on to VoiceEngine, which will take the ownership. If the |
671 // TODO(ivoc) Remove after Chrome is updated. | 660 // operation fails the file will be closed. The logging will stop |
| 661 // automatically after 10 minutes have passed, or when the StopRtcEventLog |
| 662 // function is called. A maximum filesize in bytes can be set, the logging |
| 663 // will be stopped before exceeding this limit. If max_size_bytes is set to a |
| 664 // value <= 0, no limit will be used. |
| 665 // This function as well as the StopRtcEventLog don't really belong on this |
| 666 // interface, this is a temporary solution until we move the logging object |
| 667 // from inside voice engine to webrtc::Call, which will happen when the VoE |
| 668 // restructuring effort is further along. |
| 669 // TODO(ivoc): Move this into being: |
| 670 // PeerConnection => MediaController => webrtc::Call. |
672 virtual bool StartRtcEventLog(rtc::PlatformFile file, | 671 virtual bool StartRtcEventLog(rtc::PlatformFile file, |
673 int64_t max_size_bytes) = 0; | 672 int64_t max_size_bytes) = 0; |
674 // This function is deprecated and will be removed when Chrome is updated to | 673 // Deprecated, use the version above. |
675 // use the equivalent function on PeerConnectionInterface. | |
676 // TODO(ivoc) Remove after Chrome is updated. | |
677 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0; | 674 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0; |
678 | 675 |
679 // This function is deprecated and will be removed when Chrome is updated to | 676 // Stops logging the RtcEventLog. |
680 // use the equivalent function on PeerConnectionInterface. | |
681 // TODO(ivoc) Remove after Chrome is updated. | |
682 virtual void StopRtcEventLog() = 0; | 677 virtual void StopRtcEventLog() = 0; |
683 | 678 |
684 protected: | 679 protected: |
685 // Dtor and ctor protected as objects shouldn't be created or deleted via | 680 // Dtor and ctor protected as objects shouldn't be created or deleted via |
686 // this interface. | 681 // this interface. |
687 PeerConnectionFactoryInterface() {} | 682 PeerConnectionFactoryInterface() {} |
688 ~PeerConnectionFactoryInterface() {} // NOLINT | 683 ~PeerConnectionFactoryInterface() {} // NOLINT |
689 }; | 684 }; |
690 | 685 |
691 // Create a new instance of PeerConnectionFactoryInterface. | 686 // Create a new instance of PeerConnectionFactoryInterface. |
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
726 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 721 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
727 cricket::WebRtcVideoDecoderFactory* decoder_factory) { | 722 cricket::WebRtcVideoDecoderFactory* decoder_factory) { |
728 return CreatePeerConnectionFactory( | 723 return CreatePeerConnectionFactory( |
729 worker_and_network_thread, worker_and_network_thread, signaling_thread, | 724 worker_and_network_thread, worker_and_network_thread, signaling_thread, |
730 default_adm, encoder_factory, decoder_factory); | 725 default_adm, encoder_factory, decoder_factory); |
731 } | 726 } |
732 | 727 |
733 } // namespace webrtc | 728 } // namespace webrtc |
734 | 729 |
735 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 730 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
OLD | NEW |