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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
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| 137 vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]); | 137 vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]); |
| 138 vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0); | 138 vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0); |
| 139 } | 139 } |
| 140 | 140 |
| 141 int32_t ViEChannel::Init() { | 141 int32_t ViEChannel::Init() { |
| 142 module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics()); | 142 module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics()); |
| 143 | 143 |
| 144 // RTP/RTCP initialization. | 144 // RTP/RTCP initialization. |
| 145 module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]); | 145 module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]); |
| 146 | 146 |
| 147 rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqFirRtp); | 147 rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
| 148 if (paced_sender_) { | 148 if (paced_sender_) { |
| 149 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 149 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| 150 rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); | 150 rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
| 151 } | 151 } |
| 152 packet_router_->AddRtpModule(rtp_rtcp_modules_[0]); | 152 packet_router_->AddRtpModule(rtp_rtcp_modules_[0]); |
| 153 if (sender_) { | 153 if (sender_) { |
| 154 std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]); | 154 std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]); |
| 155 send_payload_router_->SetSendingRtpModules(send_rtp_modules); | 155 send_payload_router_->SetSendingRtpModules(send_rtp_modules); |
| 156 RTC_DCHECK(!send_payload_router_->active()); | 156 RTC_DCHECK(!send_payload_router_->active()); |
| 157 } | 157 } |
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| 602 return 0; | 602 return 0; |
| 603 } | 603 } |
| 604 | 604 |
| 605 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { | 605 int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { |
| 606 // The max size of the nack list should be large enough to accommodate the | 606 // The max size of the nack list should be large enough to accommodate the |
| 607 // the number of packets (frames) resulting from the increased delay. | 607 // the number of packets (frames) resulting from the increased delay. |
| 608 // Roughly estimating for ~40 packets per frame @ 30fps. | 608 // Roughly estimating for ~40 packets per frame @ 30fps. |
| 609 return target_delay_ms * 40 * 30 / 1000; | 609 return target_delay_ms * 40 * 30 / 1000; |
| 610 } | 610 } |
| 611 | 611 |
| 612 int32_t ViEChannel::SetKeyFrameRequestMethod( | |
| 613 const KeyFrameRequestMethod method) { | |
| 614 return rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(method); | |
| 615 } | |
| 616 | |
| 617 void ViEChannel::EnableRemb(bool enable) { | 612 void ViEChannel::EnableRemb(bool enable) { |
| 618 rtp_rtcp_modules_[0]->SetREMBStatus(enable); | 613 rtp_rtcp_modules_[0]->SetREMBStatus(enable); |
| 619 } | 614 } |
| 620 | 615 |
| 621 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { | 616 int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { |
| 622 // Disable any previous registrations of this extension to avoid errors. | 617 // Disable any previous registrations of this extension to avoid errors. |
| 623 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 618 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 624 rtp_rtcp->DeregisterSendRtpHeaderExtension( | 619 rtp_rtcp->DeregisterSendRtpHeaderExtension( |
| 625 kRtpExtensionTransmissionTimeOffset); | 620 kRtpExtensionTransmissionTimeOffset); |
| 626 } | 621 } |
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| 695 } | 690 } |
| 696 | 691 |
| 697 int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) { | 692 int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) { |
| 698 return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1; | 693 return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1; |
| 699 } | 694 } |
| 700 | 695 |
| 701 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) { | 696 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) { |
| 702 rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable); | 697 rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable); |
| 703 } | 698 } |
| 704 | 699 |
| 705 void ViEChannel::SetTransmissionSmoothingStatus(bool enable) { | |
| 706 RTC_DCHECK(paced_sender_ && "No paced sender registered."); | |
| 707 paced_sender_->SetStatus(enable); | |
| 708 } | |
| 709 | |
| 710 void ViEChannel::EnableTMMBR(bool enable) { | 700 void ViEChannel::EnableTMMBR(bool enable) { |
| 711 rtp_rtcp_modules_[0]->SetTMMBRStatus(enable); | 701 rtp_rtcp_modules_[0]->SetTMMBRStatus(enable); |
| 712 } | 702 } |
| 713 | 703 |
| 714 int32_t ViEChannel::SetSSRC(const uint32_t SSRC, | 704 int32_t ViEChannel::SetSSRC(const uint32_t SSRC, |
| 715 const StreamType usage, | 705 const StreamType usage, |
| 716 const uint8_t simulcast_idx) { | 706 const uint8_t simulcast_idx) { |
| 717 RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx]; | 707 RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx]; |
| 718 if (usage == kViEStreamTypeRtx) { | 708 if (usage == kViEStreamTypeRtx) { |
| 719 rtp_rtcp->SetRtxSsrc(SSRC); | 709 rtp_rtcp->SetRtxSsrc(SSRC); |
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| 1255 CriticalSectionScoped cs(crit_.get()); | 1245 CriticalSectionScoped cs(crit_.get()); |
| 1256 receive_stats_callback_ = receive_statistics_proxy; | 1246 receive_stats_callback_ = receive_statistics_proxy; |
| 1257 } | 1247 } |
| 1258 | 1248 |
| 1259 void ViEChannel::SetIncomingVideoStream( | 1249 void ViEChannel::SetIncomingVideoStream( |
| 1260 IncomingVideoStream* incoming_video_stream) { | 1250 IncomingVideoStream* incoming_video_stream) { |
| 1261 CriticalSectionScoped cs(crit_.get()); | 1251 CriticalSectionScoped cs(crit_.get()); |
| 1262 incoming_video_stream_ = incoming_video_stream; | 1252 incoming_video_stream_ = incoming_video_stream; |
| 1263 } | 1253 } |
| 1264 } // namespace webrtc | 1254 } // namespace webrtc |
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