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webrtc
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86ce46d4ffec5d1556888309abaadeeab32744d1
86ce46d
Cleaned up and completed current dashboard milestone.
by phoglund@webrtc.org
· 13 years ago
c80d9d9
Removed default cases causing clang errors, -Wcovered-switch-default.
by mflodman@webrtc.org
· 13 years ago
5eeaa38
Improved readability of tests in master.cfg and enabling some tests
by kjellander@webrtc.org
· 13 years ago
4942832
Fix "may be used uninitialized" warning.
by andrew@webrtc.org
· 13 years ago
b783a55
Unit test for forward_error_correction.
by marpan@webrtc.org
· 13 years ago
307c1ff
Fix for issue #254: windows crash of test_fec.
by marpan@webrtc.org
· 13 years ago
dde977e
AudioFrame payload shouldn't be mutable.
by andrew@webrtc.org
· 13 years ago
ce0a6ff
Restoring previous vie_auto_test.gypi structure due to problems on Mac
by kjellander@webrtc.org
· 13 years ago
918a8bf
External transport is modified to never drop packets from the first frame.
by kjellander@webrtc.org
· 13 years ago
6838334
Fix for warning in GCC 4.6
by henrik.lundin@webrtc.org
· 13 years ago
82e1c8d
Fix for issue 253
by henrik.lundin@webrtc.org
· 13 years ago
fdf21c8
Removed dead version code.
by pwestin@webrtc.org
· 13 years ago
4ea57e5
Changed VP8 to follow the style guide a little bit more.
by pwestin@webrtc.org
· 13 years ago
9b3474a
Disable the unused API interfaces for VoE chromium build.
by xians@webrtc.org
· 13 years ago
07b45a5
Added API for getting the send-side estimated bandwidth.
by stefan@webrtc.org
· 13 years ago
ac7e89f
Correct and update LICENSE
by leozwang@webrtc.org
· 13 years ago
de66b91
In voice engine, added member audioFrame to classes AudioCodingModuleImpl and VoEBaseImpl,
by kma@webrtc.org
· 13 years ago
2b87891
Implemented build status tracking.
by phoglund@webrtc.org
· 13 years ago
fede80c
Updated test web page info for PeerConnection v2.
by henrikg@webrtc.org
· 13 years ago
d4f0a0e
Refactored the dashboard in order to add new functionality and added some new functionality.
by phoglund@webrtc.org
· 13 years ago
7fe219f
Add some additional checks for corrupt payload.
by andrew@webrtc.org
· 13 years ago
1f238fd
Update libsrtp revision in DEPS to chrome/deps/../libsrtp@119742.
by mallinath@webrtc.org
· 13 years ago
727a0a0
Fixed a bug in assembly code in aecm_core.c (hasn't caused a problem yet).
by kma@webrtc.org
· 13 years ago
d8f58a4
Cross platform build fix for SSIM (part 2)
by frkoenig@google.com
· 13 years ago
26e8a58
VAD refactor: Create() and Free()
by bjornv@webrtc.org
· 13 years ago
dd478e2
Fix for warning in GCC 4.6
by henrik.lundin@webrtc.org
· 13 years ago
79af734
This patch fixes the converity warnings in voice engine.
by xians@webrtc.org
· 13 years ago
91c6308
Fix potential VCMReceiver crash.
by stefan@webrtc.org
· 13 years ago
2919e95
Resolves Coverty issue #10347.
by henrika@webrtc.org
· 13 years ago
cdba1a8
test_fec: Reduce execution time of test, and use testsupport/fileutils.h for path of randomSeedLog file.
by marpan@webrtc.org
· 13 years ago
293d22b
Add a new macro for bit-exact audioproc tests.
by andrew@webrtc.org
· 13 years ago
72fe244
Fix neteq watchlist.
by andrew@webrtc.org
· 13 years ago
4065403
Use pointer-based CriticalSectionScoped().
by andrew@webrtc.org
· 13 years ago
7ca9925
Addding myself to video codec watch list.
by pwestin@webrtc.org
· 13 years ago
89a1000
A minor change in function WebRtcNetEQ_PacketBufferFindLowestTimestamp for
by kma@webrtc.org
· 13 years ago
7627843
Added NULL check in external transport test code.
by mflodman@webrtc.org
· 13 years ago
5dad00b
Coverty fix: FEC unintended signed extension and resource leaks.
by pwestin@webrtc.org
· 13 years ago
d3b22c9
Resolved X11 shared memiory leak.
by mflodman@webrtc.org
· 13 years ago
0c6f931
Removed versions in module/audio_processing and common_audio/vad.
by bjornv@webrtc.org
· 13 years ago
2fd1e1e
Add unittests for ReceiverFec.
by stefan@webrtc.org
· 13 years ago
31ba407
Enabled GCC 4.6 bot.
by phoglund@webrtc.org
· 13 years ago
04cf69a
Coverty: cleanup CheckCSRC.
by pwestin@webrtc.org
· 13 years ago
2f77409
Fixed C errors from GCC 4.6.
by phoglund@webrtc.org
· 13 years ago
1f99280
Fixed frame scaler bugs.
by mflodman@webrtc.org
· 13 years ago
048eb7c
Finished rewriting the audio processing test.
by phoglund@webrtc.org
· 13 years ago
832adeb
Removed MapWrapper from ViEFrameProviderBase.
by mflodman@webrtc.org
· 13 years ago
194a93a
Adding ViE NULL checks.
by mflodman@webrtc.org
· 13 years ago
cbe1de9
This CL solves three remaining Coverity warnings.
by tina.legrand@webrtc.org
· 13 years ago
4bcd177
New libsrtp roll @ 119285
by mallinath@webrtc.org
· 13 years ago
53df136
Add upload and commit checks to a common function.
by andrew@webrtc.org
· 13 years ago
a8c568f
Fix external codec erase in destructor.
by mallinath@webrtc.org
· 13 years ago
d1a860b
Fixed GCC 4.6 errors (mostly 'unused variable' errors and incorrect usage of EXPECT_EQ with booleans.
by phoglund@webrtc.org
· 13 years ago
42ae41e
Fix enumeral comparison error.
by andrew@webrtc.org
· 13 years ago
b9d7d93
Rename interface/ to include/ in audio_processing.
by andrew@webrtc.org
· 13 years ago
24bd58e
Properly count anonymous mixing participants.
by andrew@webrtc.org
· 13 years ago
96bc9c4
Enable audioproc_unittest on Windows.
by andrew@webrtc.org
· 13 years ago
7adda5c
Roll libsrtp 115467:118928.
by andrew@webrtc.org
· 13 years ago
dcf0064
Fix typo in a comment
by henrik.lundin@webrtc.org
· 13 years ago
4679652
Implemented a fix for Issue 88.
by henrik.lundin@webrtc.org
· 13 years ago
9b0a820
Fixed double erase in ViEChannelManager channel map.
by mflodman@webrtc.org
· 13 years ago
b11424b
Remove ViEShared inheritance for interface impl.
by mflodman@webrtc.org
· 13 years ago
f4b77fd
VAD refactor: Mode changed to "int".
by bjornv@webrtc.org
· 13 years ago
2a4dcd7
VAD refactor: WebRtcVad_InitCore().
by bjornv@webrtc.org
· 13 years ago
567b99b
Coverity report: fixes an issue where the returnvalue of a function is not checked.
by henrike@webrtc.org
· 13 years ago
f5d8c3b
Fix audioproc_unittest on Windows.
by andrew@webrtc.org
· 13 years ago
76fa8c9
disable failing win tests.
by ivinnichenko@webrtc.org
· 13 years ago
d224306
enable video_processing_unittests for Win
by ivinnichenko@webrtc.org
· 13 years ago
2fc722c
Enable building library and test app
by leozwang@webrtc.org
· 13 years ago
24f1c90
Enable audioproc_unittest on more platforms.
by andrew@webrtc.org
· 13 years ago
f6bb77a
Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace.
by pwestin@webrtc.org
· 13 years ago
218db3d
Iterator was invalid while removing entries from codec db maps.
by mallinath@webrtc.org
· 13 years ago
9e332ab
Make sure we check the return value from shmat().
by stefan@webrtc.org
· 13 years ago
b73c3d1
Bugfix android build. Review URL: https://webrtc-codereview.appspot.com/374003
by pwestin@webrtc.org
· 13 years ago
96c39d1
Completed implementation of oauth in coverage scripts.
by phoglund@webrtc.org
· 13 years ago
28a5cb2
Bugfix receive side only packet loss estimate with NACK.
by pwestin@webrtc.org
· 13 years ago
40d3c08
Changed max number of vie channels to 32.
by perkj@webrtc.org
· 13 years ago
52c9d47
Android, Chrome, cleanup, etc.
by ivinnichenko@webrtc.org
· 13 years ago
ba09cf1
Correcting uninitialized members.
by mflodman@webrtc.org
· 13 years ago
a5a5cbb
Switched from WebRTC wrappers to stl in ChannelManager.
by mflodman@webrtc.org
· 13 years ago
eeaf3d1
Merge /branches/3.2:r1380 to /trunk
by andrew@webrtc.org
· 13 years ago
6cf529d
Changed REMB return value to int instead of bool.
by mflodman@webrtc.org
· 13 years ago
d3a0c1c
Merge /branches/3.2:r1378 to /trunk
by andrew@webrtc.org
· 13 years ago
4bc24c4
Optimized function WebRtcSpl_FilterARFastQ12 for ARM platform.
by kma@webrtc.org
· 13 years ago
6da8eeb
Removing an assert for a case that can occur
by punyabrata@webrtc.org
· 13 years ago
f5cacdc
Fix line aligement Review URL: https://webrtc-codereview.appspot.com/373002
by leozwang@webrtc.org
· 13 years ago
2442de1
Clean up PRESUBMIT.py, and enable license check.
by andrew@webrtc.org
· 13 years ago
f9cd693
Enable vp8 and videoengine on android
by leozwang@webrtc.org
· 13 years ago
a45d05a
Add brighten.cc to makefile
by leozwang@webrtc.org
· 13 years ago
376be6c
Fix compilation error Review URL: https://webrtc-codereview.appspot.com/358005
by leozwang@webrtc.org
· 13 years ago
b30f0ed
Bugfix buffer usage out of scope.
by pwestin@webrtc.org
· 13 years ago
12dbc23
Rewrote volume test.
by phoglund@webrtc.org
· 13 years ago
175fecd
Fix clang build error.
by stefan@webrtc.org
· 13 years ago
8fe03af
Refactor to use std::list in the video rtp play tools.
by stefan@webrtc.org
· 13 years ago
152c34c
VAD-refactor. Changed to int as return value for WebRtcVad_set_mode().
by bjornv@webrtc.org
· 13 years ago
3b57ee0
Rewrote DTMF test.
by phoglund@webrtc.org
· 13 years ago
31627fe
Add vie_remb.cc to makefile
by leozwang@webrtc.org
· 13 years ago
e2ed5ba
Enable audioproc_unittest on all platforms.
by andrew@webrtc.org
· 13 years ago
2638577
Add an argument in ANDROID_NOT_SUPPORT macro
by leozwang@webrtc.org
· 13 years ago
f27916a
Remove use of MapWrapper in video_coding.
by stefan@webrtc.org
· 13 years ago
d798953
NetEqRTPplay modification
by henrik.lundin@webrtc.org
· 13 years ago
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