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

Issue 1640173004: Revert of Adding "first packet received" notification to PeerConnectionObserver. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months 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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303 303
304 bool IceRestartPending() const; 304 bool IceRestartPending() const;
305 305
306 void ResetIceRestartLatch(); 306 void ResetIceRestartLatch();
307 307
308 // Called when an RTCCertificate is generated or retrieved by 308 // Called when an RTCCertificate is generated or retrieved by
309 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. 309 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
310 void OnCertificateReady( 310 void OnCertificateReady(
311 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); 311 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
312 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); 312 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
313 void OnChannelFirstPacketReceived(cricket::BaseChannel*);
314 313
315 // For unit test. 314 // For unit test.
316 bool waiting_for_certificate_for_testing() const; 315 bool waiting_for_certificate_for_testing() const;
317 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); 316 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
318 317
319 void set_metrics_observer( 318 void set_metrics_observer(
320 webrtc::MetricsObserverInterface* metrics_observer) { 319 webrtc::MetricsObserverInterface* metrics_observer) {
321 metrics_observer_ = metrics_observer; 320 metrics_observer_ = metrics_observer;
322 } 321 }
323 322
324 // Called when voice_channel_, video_channel_ and data_channel_ are created 323 // Called when voice_channel_, video_channel_ and data_channel_ are created
325 // and destroyed. As a result of, for example, setting a new description. 324 // and destroyed. As a result of, for example, setting a new description.
326 sigslot::signal0<> SignalVoiceChannelCreated; 325 sigslot::signal0<> SignalVoiceChannelCreated;
327 sigslot::signal0<> SignalVoiceChannelDestroyed; 326 sigslot::signal0<> SignalVoiceChannelDestroyed;
328 sigslot::signal0<> SignalVideoChannelCreated; 327 sigslot::signal0<> SignalVideoChannelCreated;
329 sigslot::signal0<> SignalVideoChannelDestroyed; 328 sigslot::signal0<> SignalVideoChannelDestroyed;
330 sigslot::signal0<> SignalDataChannelCreated; 329 sigslot::signal0<> SignalDataChannelCreated;
331 sigslot::signal0<> SignalDataChannelDestroyed; 330 sigslot::signal0<> SignalDataChannelDestroyed;
332 331
333 // Called when a valid data channel OPEN message is received. 332 // Called when a valid data channel OPEN message is received.
334 // std::string represents the data channel label. 333 // std::string represents the data channel label.
335 sigslot::signal2<const std::string&, const InternalDataChannelInit&> 334 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
336 SignalDataChannelOpenMessage; 335 SignalDataChannelOpenMessage;
337 336
338 // Called when the first RTP packet is received.
339 sigslot::signal0<> SignalFirstMediaPacketReceived;
340
341 private: 337 private:
342 // Indicates the type of SessionDescription in a call to SetLocalDescription 338 // Indicates the type of SessionDescription in a call to SetLocalDescription
343 // and SetRemoteDescription. 339 // and SetRemoteDescription.
344 enum Action { 340 enum Action {
345 kOffer, 341 kOffer,
346 kPrAnswer, 342 kPrAnswer,
347 kAnswer, 343 kAnswer,
348 }; 344 };
349 345
350 // Log session state. 346 // Log session state.
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514 cricket::AudioOptions audio_options_; 510 cricket::AudioOptions audio_options_;
515 cricket::VideoOptions video_options_; 511 cricket::VideoOptions video_options_;
516 MetricsObserverInterface* metrics_observer_; 512 MetricsObserverInterface* metrics_observer_;
517 513
518 // Declares the bundle policy for the WebRTCSession. 514 // Declares the bundle policy for the WebRTCSession.
519 PeerConnectionInterface::BundlePolicy bundle_policy_; 515 PeerConnectionInterface::BundlePolicy bundle_policy_;
520 516
521 // Declares the RTCP mux policy for the WebRTCSession. 517 // Declares the RTCP mux policy for the WebRTCSession.
522 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; 518 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
523 519
524 bool has_received_media_packet_ = false;
525
526 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); 520 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
527 }; 521 };
528 } // namespace webrtc 522 } // namespace webrtc
529 523
530 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ 524 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_
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