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

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 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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174 int id = config_.rtp.extensions[i].id; 174 int id = config_.rtp.extensions[i].id;
175 // One-byte-extension local identifiers are in the range 1-14 inclusive. 175 // One-byte-extension local identifiers are in the range 1-14 inclusive.
176 DCHECK_GE(id, 1); 176 DCHECK_GE(id, 1);
177 DCHECK_LE(id, 14); 177 DCHECK_LE(id, 14);
178 if (extension == RtpExtension::kTOffset) { 178 if (extension == RtpExtension::kTOffset) {
179 CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id)); 179 CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
180 } else if (extension == RtpExtension::kAbsSendTime) { 180 } else if (extension == RtpExtension::kAbsSendTime) {
181 CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id)); 181 CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id));
182 } else if (extension == RtpExtension::kVideoRotation) { 182 } else if (extension == RtpExtension::kVideoRotation) {
183 CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id)); 183 CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id));
184 } else if (extension == RtpExtension::kTransportSequenceNumber) {
185 CHECK_EQ(0, vie_channel_->SetReceiveTransportSequenceNumber(true, id));
184 } else { 186 } else {
185 RTC_NOTREACHED() << "Unsupported RTP extension."; 187 RTC_NOTREACHED() << "Unsupported RTP extension.";
186 } 188 }
187 } 189 }
188 190
189 if (config_.rtp.fec.ulpfec_payload_type != -1) { 191 if (config_.rtp.fec.ulpfec_payload_type != -1) {
190 // ULPFEC without RED doesn't make sense. 192 // ULPFEC without RED doesn't make sense.
191 DCHECK(config_.rtp.fec.red_payload_type != -1); 193 DCHECK(config_.rtp.fec.red_payload_type != -1);
192 VideoCodec codec; 194 VideoCodec codec;
193 memset(&codec, 0, sizeof(codec)); 195 memset(&codec, 0, sizeof(codec));
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335 case newapi::kRtcpCompound: 337 case newapi::kRtcpCompound:
336 vie_channel_->SetRTCPMode(kRtcpCompound); 338 vie_channel_->SetRTCPMode(kRtcpCompound);
337 break; 339 break;
338 case newapi::kRtcpReducedSize: 340 case newapi::kRtcpReducedSize:
339 vie_channel_->SetRTCPMode(kRtcpNonCompound); 341 vie_channel_->SetRTCPMode(kRtcpNonCompound);
340 break; 342 break;
341 } 343 }
342 } 344 }
343 } // namespace internal 345 } // namespace internal
344 } // namespace webrtc 346 } // namespace webrtc
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