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| 1 /* | 1 /* | 
| 2  * libjingle | 2  * libjingle | 
| 3  * Copyright 2012 Google Inc. | 3  * Copyright 2012 Google Inc. | 
| 4  * | 4  * | 
| 5  * Redistribution and use in source and binary forms, with or without | 5  * Redistribution and use in source and binary forms, with or without | 
| 6  * modification, are permitted provided that the following conditions are met: | 6  * modification, are permitted provided that the following conditions are met: | 
| 7  * | 7  * | 
| 8  *  1. Redistributions of source code must retain the above copyright notice, | 8  *  1. Redistributions of source code must retain the above copyright notice, | 
| 9  *     this list of conditions and the following disclaimer. | 9  *     this list of conditions and the following disclaimer. | 
| 10  *  2. Redistributions in binary form must reproduce the above copyright notice, | 10  *  2. Redistributions in binary form must reproduce the above copyright notice, | 
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| 74 #include "talk/app/webrtc/datachannelinterface.h" | 74 #include "talk/app/webrtc/datachannelinterface.h" | 
| 75 #include "talk/app/webrtc/dtlsidentitystore.h" | 75 #include "talk/app/webrtc/dtlsidentitystore.h" | 
| 76 #include "talk/app/webrtc/dtmfsenderinterface.h" | 76 #include "talk/app/webrtc/dtmfsenderinterface.h" | 
| 77 #include "talk/app/webrtc/dtlsidentitystore.h" | 77 #include "talk/app/webrtc/dtlsidentitystore.h" | 
| 78 #include "talk/app/webrtc/jsep.h" | 78 #include "talk/app/webrtc/jsep.h" | 
| 79 #include "talk/app/webrtc/mediastreaminterface.h" | 79 #include "talk/app/webrtc/mediastreaminterface.h" | 
| 80 #include "talk/app/webrtc/statstypes.h" | 80 #include "talk/app/webrtc/statstypes.h" | 
| 81 #include "talk/app/webrtc/umametrics.h" | 81 #include "talk/app/webrtc/umametrics.h" | 
| 82 #include "webrtc/base/fileutils.h" | 82 #include "webrtc/base/fileutils.h" | 
| 83 #include "webrtc/base/network.h" | 83 #include "webrtc/base/network.h" | 
|  | 84 #include "webrtc/base/rtccertificate.h" | 
| 84 #include "webrtc/base/sslstreamadapter.h" | 85 #include "webrtc/base/sslstreamadapter.h" | 
| 85 #include "webrtc/base/socketaddress.h" | 86 #include "webrtc/base/socketaddress.h" | 
| 86 | 87 | 
| 87 namespace rtc { | 88 namespace rtc { | 
| 88 class SSLIdentity; | 89 class SSLIdentity; | 
| 89 class Thread; | 90 class Thread; | 
| 90 } | 91 } | 
| 91 | 92 | 
| 92 namespace cricket { | 93 namespace cricket { | 
| 93 class PortAllocator; | 94 class PortAllocator; | 
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| 235     // Chromium at the same time. | 236     // Chromium at the same time. | 
| 236     IceTransportsType type; | 237     IceTransportsType type; | 
| 237     // TODO(pthatcher): Rename this ice_servers, but update Chromium | 238     // TODO(pthatcher): Rename this ice_servers, but update Chromium | 
| 238     // at the same time. | 239     // at the same time. | 
| 239     IceServers servers; | 240     IceServers servers; | 
| 240     BundlePolicy bundle_policy; | 241     BundlePolicy bundle_policy; | 
| 241     RtcpMuxPolicy rtcp_mux_policy; | 242     RtcpMuxPolicy rtcp_mux_policy; | 
| 242     TcpCandidatePolicy tcp_candidate_policy; | 243     TcpCandidatePolicy tcp_candidate_policy; | 
| 243     int audio_jitter_buffer_max_packets; | 244     int audio_jitter_buffer_max_packets; | 
| 244     bool audio_jitter_buffer_fast_accelerate; | 245     bool audio_jitter_buffer_fast_accelerate; | 
|  | 246     std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates; | 
| 245 | 247 | 
| 246     RTCConfiguration() | 248     RTCConfiguration() | 
| 247         : type(kAll), | 249         : type(kAll), | 
| 248           bundle_policy(kBundlePolicyBalanced), | 250           bundle_policy(kBundlePolicyBalanced), | 
| 249           rtcp_mux_policy(kRtcpMuxPolicyNegotiate), | 251           rtcp_mux_policy(kRtcpMuxPolicyNegotiate), | 
| 250           tcp_candidate_policy(kTcpCandidatePolicyEnabled), | 252           tcp_candidate_policy(kTcpCandidatePolicyEnabled), | 
| 251           audio_jitter_buffer_max_packets(50), | 253           audio_jitter_buffer_max_packets(50), | 
| 252           audio_jitter_buffer_fast_accelerate(false) {} | 254           audio_jitter_buffer_fast_accelerate(false) {} | 
| 253   }; | 255   }; | 
| 254 | 256 | 
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| 594 CreatePeerConnectionFactory( | 596 CreatePeerConnectionFactory( | 
| 595     rtc::Thread* worker_thread, | 597     rtc::Thread* worker_thread, | 
| 596     rtc::Thread* signaling_thread, | 598     rtc::Thread* signaling_thread, | 
| 597     AudioDeviceModule* default_adm, | 599     AudioDeviceModule* default_adm, | 
| 598     cricket::WebRtcVideoEncoderFactory* encoder_factory, | 600     cricket::WebRtcVideoEncoderFactory* encoder_factory, | 
| 599     cricket::WebRtcVideoDecoderFactory* decoder_factory); | 601     cricket::WebRtcVideoDecoderFactory* decoder_factory); | 
| 600 | 602 | 
| 601 }  // namespace webrtc | 603 }  // namespace webrtc | 
| 602 | 604 | 
| 603 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 605 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_ | 
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