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

Issue 1649493004: Support multiple rtx codecs. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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359 if (packet_length > sizeof(restored_packet_)) 359 if (packet_length > sizeof(restored_packet_))
360 return false; 360 return false;
361 rtc::CritScope lock(&receive_cs_); 361 rtc::CritScope lock(&receive_cs_);
362 if (restored_packet_in_use_) { 362 if (restored_packet_in_use_) {
363 LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet."; 363 LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
364 return false; 364 return false;
365 } 365 }
366 if (!rtp_payload_registry_->RestoreOriginalPacket( 366 if (!rtp_payload_registry_->RestoreOriginalPacket(
367 restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(), 367 restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
368 header)) { 368 header)) {
369 LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header"; 369 LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
370 << header.ssrc << " payload type: "
371 << static_cast<int>(header.payloadType);
370 return false; 372 return false;
371 } 373 }
372 restored_packet_in_use_ = true; 374 restored_packet_in_use_ = true;
373 bool ret = OnRecoveredPacket(restored_packet_, packet_length); 375 bool ret = OnRecoveredPacket(restored_packet_, packet_length);
374 restored_packet_in_use_ = false; 376 restored_packet_in_use_ = false;
375 return ret; 377 return ret;
376 } 378 }
377 return false; 379 return false;
378 } 380 }
379 381
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471 rtp_receive_statistics_->GetStatistician(header.ssrc); 473 rtp_receive_statistics_->GetStatistician(header.ssrc);
472 if (!statistician) 474 if (!statistician)
473 return false; 475 return false;
474 // Check if this is a retransmission. 476 // Check if this is a retransmission.
475 int64_t min_rtt = 0; 477 int64_t min_rtt = 0;
476 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); 478 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
477 return !in_order && 479 return !in_order &&
478 statistician->IsRetransmitOfOldPacket(header, min_rtt); 480 statistician->IsRetransmitOfOldPacket(header, min_rtt);
479 } 481 }
480 } // namespace webrtc 482 } // namespace webrtc
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