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webrtc_m130/webrtc/modules/video_coding/main/test
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pbos@webrtc.org 62bafae661 Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 12:10:51 +00:00
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codec_database_test.cc
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codec_database_test.h
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generic_codec_test.cc
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generic_codec_test.h
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jitter_estimate_test.h
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media_opt_test.cc
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media_opt_test.h
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mt_rx_tx_test.cc
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mt_test_common.cc
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mt_test_common.h
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normal_test.cc
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normal_test.h
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pcap_file_reader_unittest.cc
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pcap_file_reader.cc
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pcap_file_reader.h
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plotJitterEstimate.m
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plotReceiveTrace.m
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plotTimingTest.m
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quality_modes_test.cc
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quality_modes_test.h
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receiver_tests.h
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release_test.h
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rtp_file_reader_unittest.cc
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rtp_file_reader.cc
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rtp_file_reader.h
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rtp_player.cc
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rtp_player.h
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subfigure.m
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test_callbacks.cc
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test_callbacks.h
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test_macros.h
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test_util.cc
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test_util.h
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tester_main.cc
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vcm_payload_sink_factory.cc
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vcm_payload_sink_factory.h
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video_rtp_play_mt.cc
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video_rtp_play.cc
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video_source.cc
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video_source.h
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