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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 | 10 |
11 #include "webrtc/video/video_send_stream.h" | 11 #include "webrtc/video/video_send_stream.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <sstream> | 14 #include <sstream> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 19 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| 20 #include "webrtc/modules/pacing/include/packet_router.h" |
20 #include "webrtc/system_wrappers/interface/logging.h" | 21 #include "webrtc/system_wrappers/interface/logging.h" |
21 #include "webrtc/system_wrappers/interface/trace_event.h" | 22 #include "webrtc/system_wrappers/interface/trace_event.h" |
22 #include "webrtc/video/video_capture_input.h" | 23 #include "webrtc/video/video_capture_input.h" |
23 #include "webrtc/video_engine/vie_channel.h" | 24 #include "webrtc/video_engine/vie_channel.h" |
24 #include "webrtc/video_engine/vie_channel_group.h" | 25 #include "webrtc/video_engine/vie_channel_group.h" |
25 #include "webrtc/video_engine/vie_defines.h" | 26 #include "webrtc/video_engine/vie_defines.h" |
26 #include "webrtc/video_engine/vie_encoder.h" | 27 #include "webrtc/video_engine/vie_encoder.h" |
27 #include "webrtc/video_send_stream.h" | 28 #include "webrtc/video_send_stream.h" |
28 | 29 |
29 namespace webrtc { | 30 namespace webrtc { |
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130 int id = config_.rtp.extensions[i].id; | 131 int id = config_.rtp.extensions[i].id; |
131 // One-byte-extension local identifiers are in the range 1-14 inclusive. | 132 // One-byte-extension local identifiers are in the range 1-14 inclusive. |
132 DCHECK_GE(id, 1); | 133 DCHECK_GE(id, 1); |
133 DCHECK_LE(id, 14); | 134 DCHECK_LE(id, 14); |
134 if (extension == RtpExtension::kTOffset) { | 135 if (extension == RtpExtension::kTOffset) { |
135 CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id)); | 136 CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id)); |
136 } else if (extension == RtpExtension::kAbsSendTime) { | 137 } else if (extension == RtpExtension::kAbsSendTime) { |
137 CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id)); | 138 CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id)); |
138 } else if (extension == RtpExtension::kVideoRotation) { | 139 } else if (extension == RtpExtension::kVideoRotation) { |
139 CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id)); | 140 CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id)); |
| 141 } else if (extension == RtpExtension::kTransportSequenceNumber) { |
| 142 CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id)); |
140 } else { | 143 } else { |
141 RTC_NOTREACHED() << "Registering unsupported RTP extension."; | 144 RTC_NOTREACHED() << "Registering unsupported RTP extension."; |
142 } | 145 } |
143 } | 146 } |
144 | 147 |
145 // TODO(pbos): Consider configuring REMB in Call. | 148 // TODO(pbos): Consider configuring REMB in Call. |
146 channel_group_->SetChannelRembStatus(true, false, vie_channel_); | 149 channel_group_->SetChannelRembStatus(true, false, vie_channel_); |
147 | 150 |
148 // Enable NACK, FEC or both. | 151 // Enable NACK, FEC or both. |
149 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; | 152 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
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500 vie_channel_->IsSendingFecEnabled()); | 503 vie_channel_->IsSendingFecEnabled()); |
501 | 504 |
502 // Restart the media flow | 505 // Restart the media flow |
503 vie_encoder_->Restart(); | 506 vie_encoder_->Restart(); |
504 | 507 |
505 return true; | 508 return true; |
506 } | 509 } |
507 | 510 |
508 } // namespace internal | 511 } // namespace internal |
509 } // namespace webrtc | 512 } // namespace webrtc |
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