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Side by Side Diff: talk/app/webrtc/peerconnection.h

Issue 1406803004: Fixing some issues with the direction attribute of m-lines in offers. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing a comment. Created 5 years, 1 month ago
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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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44 44
45 class RemoteMediaStreamFactory; 45 class RemoteMediaStreamFactory;
46 46
47 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration> 47 typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
48 StunConfigurations; 48 StunConfigurations;
49 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration> 49 typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
50 TurnConfigurations; 50 TurnConfigurations;
51 51
52 // Populates |session_options| from |rtc_options|, and returns true if options 52 // Populates |session_options| from |rtc_options|, and returns true if options
53 // are valid. 53 // are valid.
54 // Send streams should already be added to |session_options| before this method
55 // is called, as this affects the values of recv_audio and recv_video.
56 bool ConvertRtcOptionsForOffer( 54 bool ConvertRtcOptionsForOffer(
57 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options, 55 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
58 cricket::MediaSessionOptions* session_options); 56 cricket::MediaSessionOptions* session_options);
59 57
60 // Populates |session_options| from |constraints|, and returns true if all 58 // Populates |session_options| from |constraints|, and returns true if all
61 // mandatory constraints are satisfied. 59 // mandatory constraints are satisfied.
62 bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints, 60 bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
63 cricket::MediaSessionOptions* session_options); 61 cricket::MediaSessionOptions* session_options);
64 62
65 // Parses the URLs for each server in |servers| to build |stun_config| and 63 // Parses the URLs for each server in |servers| to build |stun_config| and
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389 // because its destruction fires signals (such as VoiceChannelDestroyed) 387 // because its destruction fires signals (such as VoiceChannelDestroyed)
390 // which will trigger some final actions in PeerConnection... 388 // which will trigger some final actions in PeerConnection...
391 rtc::scoped_ptr<WebRtcSession> session_; 389 rtc::scoped_ptr<WebRtcSession> session_;
392 // ... But stats_ depends on session_ so it should be destroyed even earlier. 390 // ... But stats_ depends on session_ so it should be destroyed even earlier.
393 rtc::scoped_ptr<StatsCollector> stats_; 391 rtc::scoped_ptr<StatsCollector> stats_;
394 }; 392 };
395 393
396 } // namespace webrtc 394 } // namespace webrtc
397 395
398 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ 396 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_
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