| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 2bad88d164754f1f0694e9fea1051e71b3cb5347 |
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06-Jul-2015 |
pbos <pbos@webrtc.org> |
Prevent heap overflows for incorrect FEC packet lengths.
Bugs found by manual inspection of code, not by fuzzing or packet
replays. At least one of them confirmed by local fuzzing.
BUG=chromium:496094, webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1182793002
Cr-Commit-Position: refs/heads/master@{#9542}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| bd2522abf75891f34da6f83c247c47ca95641cee |
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01-Jul-2015 |
pbos <pbos@webrtc.org> |
Fail RTP parsing on excessive padding length.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1220863002
Cr-Commit-Position: refs/heads/master@{#9525}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 2e43b26c78f465d71dfd180d55d04be1b8d4f1fb |
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30-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent OOB reads in FEC packets without complete RED headers.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1220753003
Cr-Commit-Position: refs/heads/master@{#9518}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 70d5c475ddef7ed9f848df02446d222729ed04ec |
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29-Jun-2015 |
pbos <pbos@webrtc.org> |
Prevent out-of-bounds reads for short FEC packets.
BUG=webrtc:4771
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1219703002
Cr-Commit-Position: refs/heads/master@{#9514}
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_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/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 0800db74b991dec8ef750c428eb611360a1286f4 |
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15-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add percentage of fec packets and recovered media packets to histogram stats:
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 2ec560606be6519dc4e32a1e6855b0f362ca498d |
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31-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add H.264 packetization.
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| dc80bae2a62a1bdbe0d342b3260a7e5b2cb958df |
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08-Apr-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
Clean some logs and add asserts in the way.
BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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| 7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
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06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for combining RTX and FEC/RED.
This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX.
Enables retransmissions over RTX by default in the loopback test.
BUG=1811
TESTS=voe/vie_auto_test --automated and trybots.
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2154004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
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