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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include "webrtc/video/video_send_stream.h" | 10 #include "webrtc/video/video_send_stream.h" |
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817 | 817 |
818 send_side_cc_->EnablePeriodicAlrProbing( | 818 send_side_cc_->EnablePeriodicAlrProbing( |
819 config_->periodic_alr_bandwidth_probing); | 819 config_->periodic_alr_bandwidth_probing); |
820 | 820 |
821 // RTP/RTCP initialization. | 821 // RTP/RTCP initialization. |
822 | 822 |
823 // We add the highest spatial layer first to ensure it'll be prioritized | 823 // We add the highest spatial layer first to ensure it'll be prioritized |
824 // when sending padding, with the hope that the packet rate will be smaller, | 824 // when sending padding, with the hope that the packet rate will be smaller, |
825 // and that it's more important to protect than the lower layers. | 825 // and that it's more important to protect than the lower layers. |
826 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 826 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
827 packet_router_->AddRtpModule(rtp_rtcp); | 827 packet_router_->AddSendRtpModule(rtp_rtcp); |
828 | 828 |
829 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { | 829 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { |
830 const std::string& extension = config_->rtp.extensions[i].uri; | 830 const std::string& extension = config_->rtp.extensions[i].uri; |
831 int id = config_->rtp.extensions[i].id; | 831 int id = config_->rtp.extensions[i].id; |
832 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 832 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
833 RTC_DCHECK_GE(id, 1); | 833 RTC_DCHECK_GE(id, 1); |
834 RTC_DCHECK_LE(id, 14); | 834 RTC_DCHECK_LE(id, 14); |
835 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 835 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
836 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 836 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
837 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( | 837 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( |
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896 VideoSendStreamImpl::~VideoSendStreamImpl() { | 896 VideoSendStreamImpl::~VideoSendStreamImpl() { |
897 RTC_DCHECK_RUN_ON(worker_queue_); | 897 RTC_DCHECK_RUN_ON(worker_queue_); |
898 RTC_DCHECK(!payload_router_.IsActive()) | 898 RTC_DCHECK(!payload_router_.IsActive()) |
899 << "VideoSendStreamImpl::Stop not called"; | 899 << "VideoSendStreamImpl::Stop not called"; |
900 LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString(); | 900 LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString(); |
901 | 901 |
902 rtp_rtcp_modules_[0]->SetREMBStatus(false); | 902 rtp_rtcp_modules_[0]->SetREMBStatus(false); |
903 remb_->RemoveRembSender(rtp_rtcp_modules_[0]); | 903 remb_->RemoveRembSender(rtp_rtcp_modules_[0]); |
904 | 904 |
905 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 905 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
906 packet_router_->RemoveRtpModule(rtp_rtcp); | 906 packet_router_->RemoveSendRtpModule(rtp_rtcp); |
907 delete rtp_rtcp; | 907 delete rtp_rtcp; |
908 } | 908 } |
909 } | 909 } |
910 | 910 |
911 bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { | 911 bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { |
912 // Runs on a network thread. | 912 // Runs on a network thread. |
913 RTC_DCHECK(!worker_queue_->IsCurrent()); | 913 RTC_DCHECK(!worker_queue_->IsCurrent()); |
914 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 914 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
915 rtp_rtcp->IncomingRtcpPacket(packet, length); | 915 rtp_rtcp->IncomingRtcpPacket(packet, length); |
916 return true; | 916 return true; |
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1340 std::min(config_->rtp.max_packet_size, | 1340 std::min(config_->rtp.max_packet_size, |
1341 kPathMTU - transport_overhead_bytes_per_packet_); | 1341 kPathMTU - transport_overhead_bytes_per_packet_); |
1342 | 1342 |
1343 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1343 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
1344 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1344 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
1345 } | 1345 } |
1346 } | 1346 } |
1347 | 1347 |
1348 } // namespace internal | 1348 } // namespace internal |
1349 } // namespace webrtc | 1349 } // namespace webrtc |
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