| f6975f46131981f83e0c88d276dee6b6c5753180 |
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28-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint errors cleaned from rtp_utility
R=åsapersson
BUG=webrtc:5277
Review URL: https://codereview.webrtc.org/1539423003
Cr-Commit-Position: refs/heads/master@{#11131}
/external/webrtc/webrtc/test/rtp_file_reader_unittest.cc
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| 00b8f6b3643332cce1ee711715f7fbb824d793ca |
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26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36229004
Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/rtp_file_reader_unittest.cc
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| 91d928e737732f7ad71c335da9a1c8b58f3a7701 |
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26-Nov-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/rtp_file_reader_unittest.cc
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| 38c121c484e12f677c2cb6afb882cd024bd469c1 |
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30-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Minor modifications to test::RtpFileReader
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.
Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/rtp_file_reader_unittest.cc
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| 4b5625e5acc4022fd2b9e01f7746497e569103ea |
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06-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTP video playback tool using Call APIs.
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/test/rtp_file_reader_unittest.cc
|