pbos@webrtc.org
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a048d7cb0a
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Include files from webrtc/.. paths in rtp_rtcp/
BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1557004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-29 14:27:38 +00:00 |
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solenberg@webrtc.org
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c0352d566a
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Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension.
BUG=
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1510004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-20 20:55:07 +00:00 |
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solenberg@webrtc.org
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7ebbea14a9
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Add handling of the absolute send time header extension to the rtp_rtcp module.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1480004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-16 11:10:31 +00:00 |
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pbos@webrtc.org
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3004c79c6a
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Fix clang errors in non-GYP_DEFINES=clang=1 build
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1368004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-05-07 12:36:21 +00:00 |
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andrew@webrtc.org
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14b43beb7c
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Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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2012-10-22 18:19:23 +00:00 |
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