Searched refs:rtp_parser (Results 1 – 8 of 8) sorted by relevance
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_unittest.cc | 200 webrtc::RtpUtility::RtpHeaderParser rtp_parser(data, len); in VerifyCVOPacket() local 211 ASSERT_TRUE(rtp_parser.Parse(&rtp_header, map)); in VerifyCVOPacket() 212 ASSERT_FALSE(rtp_parser.RTCP()); in VerifyCVOPacket() 335 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); in TEST_F() local 338 const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, nullptr); in TEST_F() 341 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 367 webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length); in TEST_F() local 373 const bool valid_rtp_header = rtp_parser.Parse(&rtp_header, &map); in TEST_F() 376 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 384 const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr); in TEST_F() [all …]
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D | rtp_header_parser.cc | 45 RtpUtility::RtpHeaderParser rtp_parser(packet, length); in IsRtcp() local 46 return rtp_parser.RTCP(); in IsRtcp() 52 RtpUtility::RtpHeaderParser rtp_parser(packet, length); in Parse() local 61 const bool valid_rtpheader = rtp_parser.Parse(header, &map); in Parse()
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D | rtp_sender.cc | 580 RtpUtility::RtpHeaderParser rtp_parser(buffer, length); in TrySendRedundantPayloads() local 582 rtp_parser.Parse(&rtp_header); in TrySendRedundantPayloads() 671 RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length); in SendPadData() local 673 rtp_parser.Parse(&rtp_header); in SendPadData() 723 RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length); in ReSendPacket() local 725 if (!rtp_parser.Parse(&header)) { in ReSendPacket() 913 RtpUtility::RtpHeaderParser rtp_parser(buffer, length); in PrepareAndSendPacket() local 915 rtp_parser.Parse(&rtp_header); in PrepareAndSendPacket() 1031 RtpUtility::RtpHeaderParser rtp_parser(buffer, in SendToNetwork() local 1034 rtp_parser.Parse(&rtp_header); in SendToNetwork() [all …]
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D | rtp_sender_audio.cc | 351 RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); in SendAudio() local 353 rtp_parser.Parse(&rtp_header); in SendAudio()
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D | rtp_sender_video.cc | 305 RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize); in SendVideo() local 307 rtp_parser.Parse(&rtp_header); in SendVideo()
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/external/webrtc/webrtc/test/ |
D | rtp_file_reader.cc | 456 RtpUtility::RtpHeaderParser rtp_parser(read_buffer_, marker.payload_length); in ReadPacket() local 457 if (rtp_parser.RTCP()) { in ReadPacket() 458 rtp_parser.ParseRtcp(&marker.rtp_header); in ReadPacket() 461 if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) { in ReadPacket()
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
D | rtp_to_text.cc | 30 rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); in main() local
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D | bwe_rtp_play.cc | 52 rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(parser); in main() local
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