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Side by Side Diff: webrtc/video/video_receive_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, 9 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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223 remb_->AddReceiveChannel(rtp_rtcp_); 223 remb_->AddReceiveChannel(rtp_rtcp_);
224 } 224 }
225 225
226 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { 226 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
227 const std::string& extension = config_.rtp.extensions[i].name; 227 const std::string& extension = config_.rtp.extensions[i].name;
228 int id = config_.rtp.extensions[i].id; 228 int id = config_.rtp.extensions[i].id;
229 // One-byte-extension local identifiers are in the range 1-14 inclusive. 229 // One-byte-extension local identifiers are in the range 1-14 inclusive.
230 RTC_DCHECK_GE(id, 1); 230 RTC_DCHECK_GE(id, 1);
231 RTC_DCHECK_LE(id, 14); 231 RTC_DCHECK_LE(id, 14);
232 if (extension == RtpExtension::kTOffset) { 232 if (extension == RtpExtension::kTOffset) {
233 RTC_CHECK(vie_receiver_->SetReceiveTimestampOffsetStatus(true, id)); 233 RTC_CHECK(vie_receiver_->SetReceiveTimestampOffsetStatus(id));
234 } else if (extension == RtpExtension::kAbsSendTime) { 234 } else if (extension == RtpExtension::kAbsSendTime) {
235 RTC_CHECK(vie_receiver_->SetReceiveAbsoluteSendTimeStatus(true, id)); 235 RTC_CHECK(vie_receiver_->SetReceiveAbsoluteSendTimeStatus(id));
236 } else if (extension == RtpExtension::kVideoRotation) { 236 } else if (extension == RtpExtension::kVideoRotation) {
237 RTC_CHECK(vie_receiver_->SetReceiveVideoRotationStatus(true, id)); 237 RTC_CHECK(vie_receiver_->SetReceiveVideoRotationStatus(id));
238 } else if (extension == RtpExtension::kTransportSequenceNumber) { 238 } else if (extension == RtpExtension::kTransportSequenceNumber) {
239 RTC_CHECK(vie_receiver_->SetReceiveTransportSequenceNumber(true, id)); 239 RTC_CHECK(vie_receiver_->SetReceiveTransportSequenceNumber(id));
240 } else { 240 } else {
241 RTC_NOTREACHED() << "Unsupported RTP extension."; 241 RTC_NOTREACHED() << "Unsupported RTP extension.";
242 } 242 }
243 } 243 }
244 244
245 if (config_.rtp.fec.ulpfec_payload_type != -1) { 245 if (config_.rtp.fec.ulpfec_payload_type != -1) {
246 // ULPFEC without RED doesn't make sense. 246 // ULPFEC without RED doesn't make sense.
247 RTC_DCHECK(config_.rtp.fec.red_payload_type != -1); 247 RTC_DCHECK(config_.rtp.fec.red_payload_type != -1);
248 VideoCodec codec; 248 VideoCodec codec;
249 memset(&codec, 0, sizeof(codec)); 249 memset(&codec, 0, sizeof(codec));
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428 return true; 428 return true;
429 } 429 }
430 430
431 void VideoReceiveStream::Decode() { 431 void VideoReceiveStream::Decode() {
432 static const int kMaxDecodeWaitTimeMs = 50; 432 static const int kMaxDecodeWaitTimeMs = 50;
433 vcm_->Decode(kMaxDecodeWaitTimeMs); 433 vcm_->Decode(kMaxDecodeWaitTimeMs);
434 } 434 }
435 435
436 } // namespace internal 436 } // namespace internal
437 } // namespace webrtc 437 } // namespace webrtc
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