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

Issue 1647483004: 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, 10 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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302 302
303 bool IceRestartPending() const; 303 bool IceRestartPending() const;
304 304
305 void ResetIceRestartLatch(); 305 void ResetIceRestartLatch();
306 306
307 // Called when an RTCCertificate is generated or retrieved by 307 // Called when an RTCCertificate is generated or retrieved by
308 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. 308 // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
309 void OnCertificateReady( 309 void OnCertificateReady(
310 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); 310 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
311 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp); 311 void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
312 void OnChannelFirstPacketReceived(cricket::BaseChannel*);
313 312
314 // For unit test. 313 // For unit test.
315 bool waiting_for_certificate_for_testing() const; 314 bool waiting_for_certificate_for_testing() const;
316 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); 315 const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing();
317 316
318 void set_metrics_observer( 317 void set_metrics_observer(
319 webrtc::MetricsObserverInterface* metrics_observer) { 318 webrtc::MetricsObserverInterface* metrics_observer) {
320 metrics_observer_ = metrics_observer; 319 metrics_observer_ = metrics_observer;
321 } 320 }
322 321
323 // Called when voice_channel_, video_channel_ and data_channel_ are created 322 // Called when voice_channel_, video_channel_ and data_channel_ are created
324 // and destroyed. As a result of, for example, setting a new description. 323 // and destroyed. As a result of, for example, setting a new description.
325 sigslot::signal0<> SignalVoiceChannelCreated; 324 sigslot::signal0<> SignalVoiceChannelCreated;
326 sigslot::signal0<> SignalVoiceChannelDestroyed; 325 sigslot::signal0<> SignalVoiceChannelDestroyed;
327 sigslot::signal0<> SignalVideoChannelCreated; 326 sigslot::signal0<> SignalVideoChannelCreated;
328 sigslot::signal0<> SignalVideoChannelDestroyed; 327 sigslot::signal0<> SignalVideoChannelDestroyed;
329 sigslot::signal0<> SignalDataChannelCreated; 328 sigslot::signal0<> SignalDataChannelCreated;
330 sigslot::signal0<> SignalDataChannelDestroyed; 329 sigslot::signal0<> SignalDataChannelDestroyed;
331 // Called when the whole session is destroyed. 330 // Called when the whole session is destroyed.
332 sigslot::signal0<> SignalDestroyed; 331 sigslot::signal0<> SignalDestroyed;
333 332
334 // Called when a valid data channel OPEN message is received. 333 // Called when a valid data channel OPEN message is received.
335 // std::string represents the data channel label. 334 // std::string represents the data channel label.
336 sigslot::signal2<const std::string&, const InternalDataChannelInit&> 335 sigslot::signal2<const std::string&, const InternalDataChannelInit&>
337 SignalDataChannelOpenMessage; 336 SignalDataChannelOpenMessage;
338 337
339 // Called when the first RTP packet is received.
340 sigslot::signal0<> SignalFirstMediaPacketReceived;
341
342 private: 338 private:
343 // Indicates the type of SessionDescription in a call to SetLocalDescription 339 // Indicates the type of SessionDescription in a call to SetLocalDescription
344 // and SetRemoteDescription. 340 // and SetRemoteDescription.
345 enum Action { 341 enum Action {
346 kOffer, 342 kOffer,
347 kPrAnswer, 343 kPrAnswer,
348 kAnswer, 344 kAnswer,
349 }; 345 };
350 346
351 // Log session state. 347 // Log session state.
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515 cricket::AudioOptions audio_options_; 511 cricket::AudioOptions audio_options_;
516 cricket::VideoOptions video_options_; 512 cricket::VideoOptions video_options_;
517 MetricsObserverInterface* metrics_observer_; 513 MetricsObserverInterface* metrics_observer_;
518 514
519 // Declares the bundle policy for the WebRTCSession. 515 // Declares the bundle policy for the WebRTCSession.
520 PeerConnectionInterface::BundlePolicy bundle_policy_; 516 PeerConnectionInterface::BundlePolicy bundle_policy_;
521 517
522 // Declares the RTCP mux policy for the WebRTCSession. 518 // Declares the RTCP mux policy for the WebRTCSession.
523 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; 519 PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
524 520
525 bool has_received_media_packet_ = false;
526
527 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); 521 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
528 }; 522 };
529 } // namespace webrtc 523 } // namespace webrtc
530 524
531 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_ 525 #endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_
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