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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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61 // | 61 // |
62 // 6. Provide the remote ICE candidates by calling AddIceCandidate. | 62 // 6. Provide the remote ICE candidates by calling AddIceCandidate. |
63 // | 63 // |
64 // 7. Once a candidate has been gathered, the PeerConnection will call the | 64 // 7. Once a candidate has been gathered, the PeerConnection will call the |
65 // observer function OnIceCandidate. Send these candidates to the remote peer. | 65 // observer function OnIceCandidate. Send these candidates to the remote peer. |
66 | 66 |
67 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 67 #ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
68 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 68 #define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
69 | 69 |
70 #include <memory> | 70 #include <memory> |
71 #include <ostream> | |
72 #include <string> | 71 #include <string> |
73 #include <utility> | 72 #include <utility> |
74 #include <vector> | 73 #include <vector> |
75 | 74 |
76 #include "webrtc/api/audio_codecs/audio_decoder_factory.h" | 75 #include "webrtc/api/audio_codecs/audio_decoder_factory.h" |
77 #include "webrtc/api/datachannelinterface.h" | 76 #include "webrtc/api/datachannelinterface.h" |
78 #include "webrtc/api/dtmfsenderinterface.h" | 77 #include "webrtc/api/dtmfsenderinterface.h" |
79 #include "webrtc/api/jsep.h" | 78 #include "webrtc/api/jsep.h" |
80 #include "webrtc/api/mediastreaminterface.h" | 79 #include "webrtc/api/mediastreaminterface.h" |
81 #include "webrtc/api/rtcerror.h" | 80 #include "webrtc/api/rtcerror.h" |
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894 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( | 893 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection( |
895 const PeerConnectionInterface::RTCConfiguration& configuration, | 894 const PeerConnectionInterface::RTCConfiguration& configuration, |
896 const MediaConstraintsInterface* constraints, | 895 const MediaConstraintsInterface* constraints, |
897 std::unique_ptr<cricket::PortAllocator> allocator, | 896 std::unique_ptr<cricket::PortAllocator> allocator, |
898 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, | 897 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
899 PeerConnectionObserver* observer) = 0; | 898 PeerConnectionObserver* observer) = 0; |
900 | 899 |
901 virtual rtc::scoped_refptr<MediaStreamInterface> | 900 virtual rtc::scoped_refptr<MediaStreamInterface> |
902 CreateLocalMediaStream(const std::string& label) = 0; | 901 CreateLocalMediaStream(const std::string& label) = 0; |
903 | 902 |
904 // Creates a AudioSourceInterface. | 903 // Creates an AudioSourceInterface. |
905 // |options| decides audio processing settings. | 904 // |options| decides audio processing settings. |
906 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 905 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
907 const cricket::AudioOptions& options) = 0; | 906 const cricket::AudioOptions& options) = 0; |
908 // Deprecated - use version above. | 907 // Deprecated - use version above. |
909 // Can use CopyConstraintsIntoAudioOptions to bridge the gap. | 908 // Can use CopyConstraintsIntoAudioOptions to bridge the gap. |
910 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( | 909 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( |
911 const MediaConstraintsInterface* constraints) = 0; | 910 const MediaConstraintsInterface* constraints) = 0; |
912 | 911 |
913 // Creates a VideoTrackSourceInterface from |capturer|. | 912 // Creates a VideoTrackSourceInterface from |capturer|. |
914 // TODO(deadbeef): We should aim to remove cricket::VideoCapturer from the | 913 // TODO(deadbeef): We should aim to remove cricket::VideoCapturer from the |
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1104 cricket::WebRtcVideoEncoderFactory* encoder_factory, | 1103 cricket::WebRtcVideoEncoderFactory* encoder_factory, |
1105 cricket::WebRtcVideoDecoderFactory* decoder_factory) { | 1104 cricket::WebRtcVideoDecoderFactory* decoder_factory) { |
1106 return CreatePeerConnectionFactory( | 1105 return CreatePeerConnectionFactory( |
1107 worker_and_network_thread, worker_and_network_thread, signaling_thread, | 1106 worker_and_network_thread, worker_and_network_thread, signaling_thread, |
1108 default_adm, encoder_factory, decoder_factory); | 1107 default_adm, encoder_factory, decoder_factory); |
1109 } | 1108 } |
1110 | 1109 |
1111 } // namespace webrtc | 1110 } // namespace webrtc |
1112 | 1111 |
1113 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ | 1112 #endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_ |
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