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Issue 1419673014: Remove frame time scheduing in IncomingVideoStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove unnecessary if Created 5 years 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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586 LOG(LS_INFO) << "Session: " << id() << " is destroyed."; 586 LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
587 } 587 }
588 588
589 bool WebRtcSession::Initialize( 589 bool WebRtcSession::Initialize(
590 const PeerConnectionFactoryInterface::Options& options, 590 const PeerConnectionFactoryInterface::Options& options,
591 const MediaConstraintsInterface* constraints, 591 const MediaConstraintsInterface* constraints,
592 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store, 592 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
593 const PeerConnectionInterface::RTCConfiguration& rtc_configuration) { 593 const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
594 bundle_policy_ = rtc_configuration.bundle_policy; 594 bundle_policy_ = rtc_configuration.bundle_policy;
595 rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy; 595 rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
596 video_options_.disable_prerenderer_smoothing =
597 rtc::Optional<bool>(rtc_configuration.disable_prerenderer_smoothing);
596 transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version); 598 transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
597 599
598 // Obtain a certificate from RTCConfiguration if any were provided (optional). 600 // Obtain a certificate from RTCConfiguration if any were provided (optional).
599 rtc::scoped_refptr<rtc::RTCCertificate> certificate; 601 rtc::scoped_refptr<rtc::RTCCertificate> certificate;
600 if (!rtc_configuration.certificates.empty()) { 602 if (!rtc_configuration.certificates.empty()) {
601 // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of 603 // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
602 // just picking the first one. The decision should be made based on the DTLS 604 // just picking the first one. The decision should be made based on the DTLS
603 // handshake. The DTLS negotiations need to know about all certificates. 605 // handshake. The DTLS negotiations need to know about all certificates.
604 certificate = rtc_configuration.certificates[0]; 606 certificate = rtc_configuration.certificates[0];
605 } 607 }
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2189 } 2191 }
2190 } 2192 }
2191 2193
2192 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel, 2194 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
2193 const rtc::SentPacket& sent_packet) { 2195 const rtc::SentPacket& sent_packet) {
2194 RTC_DCHECK(worker_thread()->IsCurrent()); 2196 RTC_DCHECK(worker_thread()->IsCurrent());
2195 media_controller_->call_w()->OnSentPacket(sent_packet); 2197 media_controller_->call_w()->OnSentPacket(sent_packet);
2196 } 2198 }
2197 2199
2198 } // namespace webrtc 2200 } // namespace webrtc
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