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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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| 821 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 821 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| 822 transport->packet_router()->AddSendRtpModule(rtp_rtcp); | 822 transport->packet_router()->AddSendRtpModule(rtp_rtcp); |
| 823 | 823 |
| 824 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { | 824 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { |
| 825 const std::string& extension = config_->rtp.extensions[i].uri; | 825 const std::string& extension = config_->rtp.extensions[i].uri; |
| 826 int id = config_->rtp.extensions[i].id; | 826 int id = config_->rtp.extensions[i].id; |
| 827 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 827 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
| 828 RTC_DCHECK_GE(id, 1); | 828 RTC_DCHECK_GE(id, 1); |
| 829 RTC_DCHECK_LE(id, 14); | 829 RTC_DCHECK_LE(id, 14); |
| 830 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 830 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
| 831 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType && |
| 832 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) { |
| 833 continue; |
| 834 } |
| 831 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 835 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 832 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( | 836 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( |
| 833 StringToRtpExtensionType(extension), id)); | 837 StringToRtpExtensionType(extension), id)); |
| 834 } | 838 } |
| 835 } | 839 } |
| 836 | 840 |
| 837 ConfigureProtection(); | 841 ConfigureProtection(); |
| 838 ConfigureSsrcs(); | 842 ConfigureSsrcs(); |
| 839 | 843 |
| 840 // TODO(pbos): Should we set CNAME on all RTP modules? | 844 // TODO(pbos): Should we set CNAME on all RTP modules? |
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| 1336 std::min(config_->rtp.max_packet_size, | 1340 std::min(config_->rtp.max_packet_size, |
| 1337 kPathMTU - transport_overhead_bytes_per_packet_); | 1341 kPathMTU - transport_overhead_bytes_per_packet_); |
| 1338 | 1342 |
| 1339 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 1343 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 1340 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); | 1344 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
| 1341 } | 1345 } |
| 1342 } | 1346 } |
| 1343 | 1347 |
| 1344 } // namespace internal | 1348 } // namespace internal |
| 1345 } // namespace webrtc | 1349 } // namespace webrtc |
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