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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: Rebase, again Created 5 years, 4 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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172 int id = config_.rtp.extensions[i].id; 172 int id = config_.rtp.extensions[i].id;
173 // One-byte-extension local identifiers are in the range 1-14 inclusive. 173 // One-byte-extension local identifiers are in the range 1-14 inclusive.
174 DCHECK_GE(id, 1); 174 DCHECK_GE(id, 1);
175 DCHECK_LE(id, 14); 175 DCHECK_LE(id, 14);
176 if (extension == RtpExtension::kTOffset) { 176 if (extension == RtpExtension::kTOffset) {
177 CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id)); 177 CHECK_EQ(0, vie_channel_->SetReceiveTimestampOffsetStatus(true, id));
178 } else if (extension == RtpExtension::kAbsSendTime) { 178 } else if (extension == RtpExtension::kAbsSendTime) {
179 CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id)); 179 CHECK_EQ(0, vie_channel_->SetReceiveAbsoluteSendTimeStatus(true, id));
180 } else if (extension == RtpExtension::kVideoRotation) { 180 } else if (extension == RtpExtension::kVideoRotation) {
181 CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id)); 181 CHECK_EQ(0, vie_channel_->SetReceiveVideoRotationStatus(true, id));
182 } else if (extension == RtpExtension::kTransportSequenceNumber) {
183 CHECK_EQ(0, vie_channel_->SetReceiveTransportSequenceNumber(true, id));
182 } else { 184 } else {
183 RTC_NOTREACHED() << "Unsupported RTP extension."; 185 RTC_NOTREACHED() << "Unsupported RTP extension.";
184 } 186 }
185 } 187 }
186 188
187 if (config_.rtp.fec.ulpfec_payload_type != -1) { 189 if (config_.rtp.fec.ulpfec_payload_type != -1) {
188 // ULPFEC without RED doesn't make sense. 190 // ULPFEC without RED doesn't make sense.
189 DCHECK(config_.rtp.fec.red_payload_type != -1); 191 DCHECK(config_.rtp.fec.red_payload_type != -1);
190 VideoCodec codec; 192 VideoCodec codec;
191 memset(&codec, 0, sizeof(codec)); 193 memset(&codec, 0, sizeof(codec));
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333 case newapi::kRtcpCompound: 335 case newapi::kRtcpCompound:
334 vie_channel_->SetRTCPMode(kRtcpCompound); 336 vie_channel_->SetRTCPMode(kRtcpCompound);
335 break; 337 break;
336 case newapi::kRtcpReducedSize: 338 case newapi::kRtcpReducedSize:
337 vie_channel_->SetRTCPMode(kRtcpNonCompound); 339 vie_channel_->SetRTCPMode(kRtcpNonCompound);
338 break; 340 break;
339 } 341 }
340 } 342 }
341 } // namespace internal 343 } // namespace internal
342 } // namespace webrtc 344 } // namespace webrtc
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