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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 |
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486 // Returns the current SignalingState. | 486 // Returns the current SignalingState. |
487 virtual SignalingState signaling_state() = 0; | 487 virtual SignalingState signaling_state() = 0; |
488 | 488 |
489 // TODO(bemasc): Remove ice_state when callers are changed to | 489 // TODO(bemasc): Remove ice_state when callers are changed to |
490 // IceConnection/GatheringState. | 490 // IceConnection/GatheringState. |
491 // Returns the current IceState. | 491 // Returns the current IceState. |
492 virtual IceState ice_state() = 0; | 492 virtual IceState ice_state() = 0; |
493 virtual IceConnectionState ice_connection_state() = 0; | 493 virtual IceConnectionState ice_connection_state() = 0; |
494 virtual IceGatheringState ice_gathering_state() = 0; | 494 virtual IceGatheringState ice_gathering_state() = 0; |
495 | 495 |
| 496 // Starts RtcEventLog using existing file. Takes ownership of |file| and |
| 497 // passes it on to Call, which will take the ownership. If the |
| 498 // operation fails the file will be closed. The logging will stop |
| 499 // automatically after 10 minutes have passed, or when the StopRtcEventLog |
| 500 // function is called. |
| 501 // TODO(ivoc): Make this pure virtual when Chrome is updated. |
| 502 virtual bool StartRtcEventLog(rtc::PlatformFile file, |
| 503 int64_t max_size_bytes) { |
| 504 return false; |
| 505 } |
| 506 |
| 507 // Stops logging the RtcEventLog. |
| 508 // TODO(ivoc): Make this pure virtual when Chrome is updated. |
| 509 virtual void StopRtcEventLog() {} |
| 510 |
496 // Terminates all media and closes the transport. | 511 // Terminates all media and closes the transport. |
497 virtual void Close() = 0; | 512 virtual void Close() = 0; |
498 | 513 |
499 protected: | 514 protected: |
500 // Dtor protected as objects shouldn't be deleted via this interface. | 515 // Dtor protected as objects shouldn't be deleted via this interface. |
501 ~PeerConnectionInterface() {} | 516 ~PeerConnectionInterface() {} |
502 }; | 517 }; |
503 | 518 |
504 // PeerConnection callback interface. Application should implement these | 519 // PeerConnection callback interface. Application should implement these |
505 // methods. | 520 // methods. |
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652 // the ownerhip. If the operation fails, the file will be closed. | 667 // the ownerhip. If the operation fails, the file will be closed. |
653 // A maximum file size in bytes can be specified. When the file size limit is | 668 // A maximum file size in bytes can be specified. When the file size limit is |
654 // reached, logging is stopped automatically. If max_size_bytes is set to a | 669 // reached, logging is stopped automatically. If max_size_bytes is set to a |
655 // value <= 0, no limit will be used, and logging will continue until the | 670 // value <= 0, no limit will be used, and logging will continue until the |
656 // StopAecDump function is called. | 671 // StopAecDump function is called. |
657 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; | 672 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0; |
658 | 673 |
659 // Stops logging the AEC dump. | 674 // Stops logging the AEC dump. |
660 virtual void StopAecDump() = 0; | 675 virtual void StopAecDump() = 0; |
661 | 676 |
662 // Starts RtcEventLog using existing file. Takes ownership of |file| and | 677 // This function is deprecated and will be removed when Chrome is updated to |
663 // passes it on to VoiceEngine, which will take the ownership. If the | 678 // use the equivalent function on PeerConnectionInterface. |
664 // operation fails the file will be closed. The logging will stop | 679 // TODO(ivoc) Remove after Chrome is updated. |
665 // automatically after 10 minutes have passed, or when the StopRtcEventLog | |
666 // function is called. A maximum filesize in bytes can be set, the logging | |
667 // will be stopped before exceeding this limit. If max_size_bytes is set to a | |
668 // value <= 0, no limit will be used. | |
669 // This function as well as the StopRtcEventLog don't really belong on this | |
670 // interface, this is a temporary solution until we move the logging object | |
671 // from inside voice engine to webrtc::Call, which will happen when the VoE | |
672 // restructuring effort is further along. | |
673 // TODO(ivoc): Move this into being: | |
674 // PeerConnection => MediaController => webrtc::Call. | |
675 virtual bool StartRtcEventLog(rtc::PlatformFile file, | 680 virtual bool StartRtcEventLog(rtc::PlatformFile file, |
676 int64_t max_size_bytes) = 0; | 681 int64_t max_size_bytes) = 0; |
677 // Deprecated, use the version above. | 682 // This function is deprecated and will be removed when Chrome is updated to |
| 683 // use the equivalent function on PeerConnectionInterface. |
| 684 // TODO(ivoc) Remove after Chrome is updated. |
678 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0; | 685 virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0; |
679 | 686 |
680 // Stops logging the RtcEventLog. | 687 // This function is deprecated and will be removed when Chrome is updated to |
| 688 // use the equivalent function on PeerConnectionInterface. |
| 689 // TODO(ivoc) Remove after Chrome is updated. |
681 virtual void StopRtcEventLog() = 0; | 690 virtual void StopRtcEventLog() = 0; |
682 | 691 |
683 protected: | 692 protected: |
684 // Dtor and ctor protected as objects shouldn't be created or deleted via | 693 // Dtor and ctor protected as objects shouldn't be created or deleted via |
685 // this interface. | 694 // this interface. |
686 PeerConnectionFactoryInterface() {} | 695 PeerConnectionFactoryInterface() {} |
687 ~PeerConnectionFactoryInterface() {} // NOLINT | 696 ~PeerConnectionFactoryInterface() {} // NOLINT |
688 }; | 697 }; |
689 | 698 |
690 // Create a new instance of PeerConnectionFactoryInterface. | 699 // Create a new instance of PeerConnectionFactoryInterface. |
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725 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 734 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
726 cricket::WebRtcVideoDecoderFactory* decoder_factory) { | 735 cricket::WebRtcVideoDecoderFactory* decoder_factory) { |
727 return CreatePeerConnectionFactory( | 736 return CreatePeerConnectionFactory( |
728 worker_and_network_thread, worker_and_network_thread, signaling_thread, | 737 worker_and_network_thread, worker_and_network_thread, signaling_thread, |
729 default_adm, encoder_factory, decoder_factory); | 738 default_adm, encoder_factory, decoder_factory); |
730 } | 739 } |
731 | 740 |
732 } // namespace webrtc | 741 } // namespace webrtc |
733 | 742 |
734 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 743 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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