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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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839 | 839 |
840 if (config_->periodic_alr_bandwidth_probing) { | 840 if (config_->periodic_alr_bandwidth_probing) { |
841 transport->send_side_cc()->EnablePeriodicAlrProbing(true); | 841 transport->send_side_cc()->EnablePeriodicAlrProbing(true); |
842 } | 842 } |
843 | 843 |
844 // RTP/RTCP initialization. | 844 // RTP/RTCP initialization. |
845 | 845 |
846 // We add the highest spatial layer first to ensure it'll be prioritized | 846 // We add the highest spatial layer first to ensure it'll be prioritized |
847 // when sending padding, with the hope that the packet rate will be smaller, | 847 // when sending padding, with the hope that the packet rate will be smaller, |
848 // and that it's more important to protect than the lower layers. | 848 // and that it's more important to protect than the lower layers. |
849 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 849 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
850 transport->packet_router()->AddSendRtpModule(rtp_rtcp); | 850 constexpr bool remb_candidate = true; |
| 851 transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate); |
| 852 } |
851 | 853 |
852 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { | 854 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { |
853 const std::string& extension = config_->rtp.extensions[i].uri; | 855 const std::string& extension = config_->rtp.extensions[i].uri; |
854 int id = config_->rtp.extensions[i].id; | 856 int id = config_->rtp.extensions[i].id; |
855 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 857 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
856 RTC_DCHECK_GE(id, 1); | 858 RTC_DCHECK_GE(id, 1); |
857 RTC_DCHECK_LE(id, 14); | 859 RTC_DCHECK_LE(id, 14); |
858 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 860 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
859 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType && | 861 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType && |
860 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) { | 862 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) { |
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1368 std::min(config_->rtp.max_packet_size, | 1370 std::min(config_->rtp.max_packet_size, |
1369 kPathMTU - transport_overhead_bytes_per_packet_); | 1371 kPathMTU - transport_overhead_bytes_per_packet_); |
1370 | 1372 |
1371 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1373 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
1372 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1374 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
1373 } | 1375 } |
1374 } | 1376 } |
1375 | 1377 |
1376 } // namespace internal | 1378 } // namespace internal |
1377 } // namespace webrtc | 1379 } // namespace webrtc |
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