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Side by Side Diff: webrtc/video/video_send_stream.cc

Issue 1735033003: Removes use of DeRegister Rtp Header Extension for video (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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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
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222 222
223 call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver()); 223 call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
224 224
225 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 225 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
226 const std::string& extension = config_.rtp.extensions[i].name; 226 const std::string& extension = config_.rtp.extensions[i].name;
227 int id = config_.rtp.extensions[i].id; 227 int id = config_.rtp.extensions[i].id;
228 // One-byte-extension local identifiers are in the range 1-14 inclusive. 228 // One-byte-extension local identifiers are in the range 1-14 inclusive.
229 RTC_DCHECK_GE(id, 1); 229 RTC_DCHECK_GE(id, 1);
230 RTC_DCHECK_LE(id, 14); 230 RTC_DCHECK_LE(id, 14);
231 if (extension == RtpExtension::kTOffset) { 231 if (extension == RtpExtension::kTOffset) {
232 RTC_CHECK_EQ(0, vie_channel_.SetSendTimestampOffsetStatus(true, id)); 232 RTC_CHECK_EQ(0, vie_channel_.SetSendTimestampOffsetStatus(id));
233 } else if (extension == RtpExtension::kAbsSendTime) { 233 } else if (extension == RtpExtension::kAbsSendTime) {
234 RTC_CHECK_EQ(0, vie_channel_.SetSendAbsoluteSendTimeStatus(true, id)); 234 RTC_CHECK_EQ(0, vie_channel_.SetSendAbsoluteSendTimeStatus(id));
235 } else if (extension == RtpExtension::kVideoRotation) { 235 } else if (extension == RtpExtension::kVideoRotation) {
236 RTC_CHECK_EQ(0, vie_channel_.SetSendVideoRotationStatus(true, id)); 236 RTC_CHECK_EQ(0, vie_channel_.SetSendVideoRotationStatus(id));
237 } else if (extension == RtpExtension::kTransportSequenceNumber) { 237 } else if (extension == RtpExtension::kTransportSequenceNumber) {
238 RTC_CHECK_EQ(0, vie_channel_.SetSendTransportSequenceNumber(true, id)); 238 RTC_CHECK_EQ(0, vie_channel_.SetSendTransportSequenceNumber(id));
239 } else { 239 } else {
240 RTC_NOTREACHED() << "Registering unsupported RTP extension."; 240 RTC_NOTREACHED() << "Registering unsupported RTP extension.";
241 } 241 }
242 } 242 }
243 243
244 remb_->AddRembSender(rtp_rtcp_modules_[0]); 244 remb_->AddRembSender(rtp_rtcp_modules_[0]);
245 rtp_rtcp_modules_[0]->SetREMBStatus(true); 245 rtp_rtcp_modules_[0]->SetREMBStatus(true);
246 246
247 // Enable NACK, FEC or both. 247 // Enable NACK, FEC or both.
248 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; 248 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
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617 return false; 617 return false;
618 } 618 }
619 619
620 // Restart the media flow 620 // Restart the media flow
621 vie_encoder_.Restart(); 621 vie_encoder_.Restart();
622 622
623 return true; 623 return true;
624 } 624 }
625 } // namespace internal 625 } // namespace internal
626 } // namespace webrtc 626 } // namespace webrtc
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