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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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| 735 transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate); | 735 transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate); |
| 736 } | 736 } |
| 737 | 737 |
| 738 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { | 738 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { |
| 739 const std::string& extension = config_->rtp.extensions[i].uri; | 739 const std::string& extension = config_->rtp.extensions[i].uri; |
| 740 int id = config_->rtp.extensions[i].id; | 740 int id = config_->rtp.extensions[i].id; |
| 741 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 741 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
| 742 RTC_DCHECK_GE(id, 1); | 742 RTC_DCHECK_GE(id, 1); |
| 743 RTC_DCHECK_LE(id, 14); | 743 RTC_DCHECK_LE(id, 14); |
| 744 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 744 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
| 745 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType && | |
| 746 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) { | |
| 747 continue; | |
| 748 } | |
| 749 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 745 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 750 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( | 746 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( |
| 751 StringToRtpExtensionType(extension), id)); | 747 StringToRtpExtensionType(extension), id)); |
| 752 } | 748 } |
| 753 } | 749 } |
| 754 | 750 |
| 755 ConfigureProtection(); | 751 ConfigureProtection(); |
| 756 ConfigureSsrcs(); | 752 ConfigureSsrcs(); |
| 757 | 753 |
| 758 // TODO(pbos): Should we set CNAME on all RTP modules? | 754 // TODO(pbos): Should we set CNAME on all RTP modules? |
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| 1256 std::min(config_->rtp.max_packet_size, | 1252 std::min(config_->rtp.max_packet_size, |
| 1257 kPathMTU - transport_overhead_bytes_per_packet_); | 1253 kPathMTU - transport_overhead_bytes_per_packet_); |
| 1258 | 1254 |
| 1259 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1255 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 1260 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1256 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
| 1261 } | 1257 } |
| 1262 } | 1258 } |
| 1263 | 1259 |
| 1264 } // namespace internal | 1260 } // namespace internal |
| 1265 } // namespace webrtc | 1261 } // namespace webrtc |
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