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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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| 303       if (header.extension.hasTransmissionTimeOffset) | 303       if (header.extension.hasTransmissionTimeOffset) | 
| 304         ss << ", toffset: " << header.extension.transmissionTimeOffset; | 304         ss << ", toffset: " << header.extension.transmissionTimeOffset; | 
| 305       if (header.extension.hasAbsoluteSendTime) | 305       if (header.extension.hasAbsoluteSendTime) | 
| 306         ss << ", abs send time: " << header.extension.absoluteSendTime; | 306         ss << ", abs send time: " << header.extension.absoluteSendTime; | 
| 307       LOG(LS_INFO) << ss.str(); | 307       LOG(LS_INFO) << ss.str(); | 
| 308       last_packet_log_ms_ = now_ms; | 308       last_packet_log_ms_ = now_ms; | 
| 309     } | 309     } | 
| 310   } | 310   } | 
| 311 | 311 | 
| 312   remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length, | 312   remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length, | 
| 313                                             header, true); | 313                                             header); | 
| 314   header.payload_type_frequency = kVideoPayloadTypeFrequency; | 314   header.payload_type_frequency = kVideoPayloadTypeFrequency; | 
| 315 | 315 | 
| 316   bool in_order = IsPacketInOrder(header); | 316   bool in_order = IsPacketInOrder(header); | 
| 317   rtp_payload_registry_.SetIncomingPayloadType(header); | 317   rtp_payload_registry_.SetIncomingPayloadType(header); | 
| 318   bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); | 318   bool ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); | 
| 319   // Update receive statistics after ReceivePacket. | 319   // Update receive statistics after ReceivePacket. | 
| 320   // Receive statistics will be reset if the payload type changes (make sure | 320   // Receive statistics will be reset if the payload type changes (make sure | 
| 321   // that the first packet is included in the stats). | 321   // that the first packet is included in the stats). | 
| 322   rtp_receive_statistics_->IncomingPacket( | 322   rtp_receive_statistics_->IncomingPacket( | 
| 323       header, rtp_packet_length, IsPacketRetransmitted(header, in_order)); | 323       header, rtp_packet_length, IsPacketRetransmitted(header, in_order)); | 
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| 536     const std::string& extension, int id) { | 536     const std::string& extension, int id) { | 
| 537   // One-byte-extension local identifiers are in the range 1-14 inclusive. | 537   // One-byte-extension local identifiers are in the range 1-14 inclusive. | 
| 538   RTC_DCHECK_GE(id, 1); | 538   RTC_DCHECK_GE(id, 1); | 
| 539   RTC_DCHECK_LE(id, 14); | 539   RTC_DCHECK_LE(id, 14); | 
| 540   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 540   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); | 
| 541   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | 541   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( | 
| 542       StringToRtpExtensionType(extension), id)); | 542       StringToRtpExtensionType(extension), id)); | 
| 543 } | 543 } | 
| 544 | 544 | 
| 545 }  // namespace webrtc | 545 }  // namespace webrtc | 
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