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gerrit-public.fairphone.software
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platform
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external
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webrtc
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da7f6589aa42724de48a8da049b417188d80aca8
da7f658
Add svn:ignore to avoid re-download of resources
by kjellander@webrtc.org
· 11 years ago
b8cb85b
Fix broken build on x86 Android
by fischman@webrtc.org
· 11 years ago
7b273a5
PeerConnection iOS: update README instructions
by fischman@webrtc.org
· 11 years ago
07a6fbe
Update talk to 56092586.
by wu@webrtc.org
· 11 years ago
3779c1c
Fix invalid .sha1 files for audio_coding
by kjellander@webrtc.org
· 11 years ago
8017458
Replace old resources download script with depot_tools
by kjellander@webrtc.org
· 11 years ago
a452fc2
Remove resources/ svn:ignore to prepare for updated resource handling
by kjellander@webrtc.org
· 11 years ago
58bcdee
Roll chromium_revision 229708:231713
by kjellander@webrtc.org
· 11 years ago
766154a
Removed unused code.
by asapersson@webrtc.org
· 11 years ago
e2df8b7
Make video quality analysis unittests print to log instead of stdout.
by kjellander@webrtc.org
· 11 years ago
5dd2ecb
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
74e6e84
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
d705649
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
1a4ed0d
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
de30501
Update talk to 55906045.
by wu@webrtc.org
· 11 years ago
58cd316
Address Clag Analyzer issues.
by turaj@webrtc.org
· 11 years ago
7d6bd22
Propagate estimated RTT from receivers to rtt observer.
by asapersson@webrtc.org
· 11 years ago
da2c37b
Video bandwidth not reported correctly
by sprang@webrtc.org
· 11 years ago
773e727
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
by sergeyu@chromium.org
· 11 years ago
de748c8
Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build.
by wu@webrtc.org
· 11 years ago
dce70cc
Add delay limit to ChokeFilter.
by solenberg@webrtc.org
· 11 years ago
f424cb8
Update talk to 55863981.
by wu@webrtc.org
· 11 years ago
d6e4663
Logging for BWE test framework.
by solenberg@webrtc.org
· 11 years ago
cecfd18
Update talk to 55821645.
by wu@webrtc.org
· 11 years ago
ec4cccc
Update libyuv to 832.
by wu@webrtc.org
· 11 years ago
47ebbad
Make video/ only depend on video_engine_core.
by pbos@webrtc.org
· 11 years ago
def22b4
Stop DirectTransports in VideoSendStreamTests.
by pbos@webrtc.org
· 11 years ago
55e1723
Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN.
by turaj@webrtc.org
· 11 years ago
9ca93a8
Explicitly @synthesize ObjC @properties
by fischman@webrtc.org
· 11 years ago
0aeb22e
Adding tl0idx consideration for continuity
by mikhal@webrtc.org
· 11 years ago
0803c03
Fix build/isolate.gypi path in webrtc_tests.gypi.
by pbos@webrtc.org
· 11 years ago
b7a1718
Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038.
by fischman@webrtc.org
· 11 years ago
16e03b7
Separate Call API/build files from video_engine/.
by pbos@webrtc.org
· 11 years ago
850bcbe
Remove frame_callback.h include in webrtcvie.h.
by pbos@webrtc.org
· 11 years ago
1a3a6e5
Removing the threshold from the auto-mute APIs
by henrik.lundin@webrtc.org
· 11 years ago
fe5d36b
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
by sprang@webrtc.org
· 11 years ago
97077a3
Update libjingle to 55618622. Update libyuv to r826.
by wu@webrtc.org
· 11 years ago
728bc0f
Add qiang.lu@intel.com to WATCHLISTS.
by fischman@webrtc.org
· 11 years ago
c94abd3
Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
by xians@webrtc.org
· 11 years ago
e4e5683
Clean up tsan suppression file:
by wu@webrtc.org
· 11 years ago
0729460
Added a "interleaved_" flag to webrtc::AudioFrame.
by xians@webrtc.org
· 11 years ago
442c5e4
Update adapter.js to use TURN transport parameters for FF version 27 & above.
by vikasmarwaha@webrtc.org
· 11 years ago
d674a56
Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511.
by vikasmarwaha@webrtc.org
· 11 years ago
b3731da
Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_.
by andrew@webrtc.org
· 11 years ago
b56d0e3
Change the low-bitrate handling in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
37bb497
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
by fischman@webrtc.org
· 11 years ago
d371a29
Fix tsan failures for libjingle_unittest.
by wu@webrtc.org
· 11 years ago
d1bcf11
Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
by andrew@webrtc.org
· 11 years ago
22858d4
Add an extended filter option to audioproc.
by andrew@webrtc.org
· 11 years ago
042e91c
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
by asapersson@webrtc.org
· 11 years ago
ba975e2
Porting auto mute to new ViE API
by henrik.lundin@webrtc.org
· 11 years ago
886aef0
Fixing broken tests in voe_auto_test extended
by tina.legrand@webrtc.org
· 11 years ago
8804a29
Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread.
by wu@webrtc.org
· 11 years ago
4d7116b
Fix tsan failures on filevideocapturer.cc.
by wu@webrtc.org
· 11 years ago
90d8719
Radix should be specified when calling ParseInt function in adapter.js. Refer to issue 2490.
by vikasmarwaha@webrtc.org
· 11 years ago
8575980
Add default trybots for WebRTC try server.
by kjellander@webrtc.org
· 11 years ago
31628aa
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
06b60c0
Roll chromium_revision 228675:229708
by kjellander@webrtc.org
· 11 years ago
621df67
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
by andrew@webrtc.org
· 11 years ago
943e3b9
Add CurrentLayerId() to temporal layers.
by marpan@webrtc.org
· 11 years ago
50bc553
Reenable DTLS renegotiation unittest in libjingle.
by mallinath@webrtc.org
· 11 years ago
9c735c4
Updated WebRTC version to 3.45
by elham@webrtc.org
· 11 years ago
8215106
Framework for testing bandwidth estimation.
by solenberg@webrtc.org
· 11 years ago
29dd0de
Changing the bitrate clamping in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
0d19ed9
AutoMute: Adding channel_id parameter to callback.
by henrik.lundin@webrtc.org
· 11 years ago
fe1ef93
Implement I420FrameCallbacks in Call.
by pbos@webrtc.org
· 11 years ago
e053629
Make sure the first frame isn't dropped.
by pbos@webrtc.org
· 11 years ago
eb61a85
Move audio_e2e_harness into include_tests==1 condition.
by kjellander@webrtc.org
· 11 years ago
88a3108
Add audio_e2e_test target to tools.gyp
by kjellander@webrtc.org
· 11 years ago
fb648da
Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug.
by wu@webrtc.org
· 11 years ago
3c5d2b4
Thread::Stop() must be called before any subclass's destructor completes.
by wu@webrtc.org
· 11 years ago
3e00505
Have padding decay to zero if no frames are being captured.
by stefan@webrtc.org
· 11 years ago
893c07f
Disable the -Wno-unused-const-variable Clang warning on Mac
by kjellander@webrtc.org
· 11 years ago
89b1e68
Minor comment fix after clang reformat.
by andrew@webrtc.org
· 11 years ago
1c82037
AppRTCDemo(android): remove vestigial mentions of PowerManager
by fischman@webrtc.org
· 11 years ago
2df89c0
MouseCursorMonitor implementation for OSX and Windows.
by sergeyu@chromium.org
· 11 years ago
6b426ba
Final round of LSan suppressions (take 2)
by kjellander@webrtc.org
· 11 years ago
6342066
Fix tsan failures in channel.cc regarding to the volume settings.
by wu@webrtc.org
· 11 years ago
b22049b
Final round of LSan suppressions.
by kjellander@webrtc.org
· 11 years ago
8a7b89f
More libjingle LSan suppressions.
by kjellander@webrtc.org
· 11 years ago
675e260
Check the number of playout channels instead of the send channels in StopPlayout()
by xians@webrtc.org
· 11 years ago
e61da8c
Suppressions and PRESUBMIT.py for LSan
by kjellander@webrtc.org
· 11 years ago
c11148b
Compound/reduced-size RTCP in VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
603ed98
Suppress race condition warn in CallTest_ReceivesAndRetransmitsNack_Test
by sprang@webrtc.org
· 11 years ago
54e729b
Remove tsan suppression for the failure that's already fixed.
by wu@webrtc.org
· 11 years ago
853dd07
Add issue links to the tsanv2 suppressions.
by wu@webrtc.org
· 11 years ago
e7771e2
Add /webrtc/modules/audio_device/android/test/{bin,gen,libs} to .gitignore
by fischman@webrtc.org
· 11 years ago
d030972
Remove unused kPowTableFrac which causes anroid clang build failure.
by wu@webrtc.org
· 11 years ago
1d1ffc9
Update talk to 54898858.
by wu@webrtc.org
· 11 years ago
83e9c89
Exclude more tests for TSan on Windows.
by kjellander@webrtc.org
· 11 years ago
d1cfa71
TSan v2 suppressions and exclusions for libjingle tests.
by kjellander@webrtc.org
· 11 years ago
25fce9a
Fixed issue with how MTU is calculated.
by sprang@webrtc.org
· 11 years ago
b400aa7
Don't pad if only one stream is sent, except if auto muted.
by stefan@webrtc.org
· 11 years ago
e7009f3
Revert "Disable tests for TSan v2"
by kjellander@webrtc.org
· 11 years ago
5d957e2
Wired up max packet size and added simple test.
by sprang@webrtc.org
· 11 years ago
9401524
Run FullStack tests without render windows.
by pbos@webrtc.org
· 11 years ago
5ed4f46
Remove TSan v2 disabled test in condition_variable_unittest.cc
by kjellander@webrtc.org
· 11 years ago
662b1c5
Add suppressions for DrMemory and TSan on Windows.
by kjellander@webrtc.org
· 11 years ago
b44c2a3
Open file in binary in CreateFromYuvFile().
by pbos@webrtc.org
· 11 years ago
e6e749d
Add MouseCursorRenderer.
by sergeyu@chromium.org
· 11 years ago
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