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fp2-dev
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platform
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chromium_org
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third_party
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webrtc
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499392ce105faca0d368e8512e287e81f9392950
499392c
Minor fix to avoid breakage
by henrik.lundin@webrtc.org
· 11 years ago
a8532a8
Disable Receiver unittests on Android.
by turaj@webrtc.org
· 11 years ago
85cdc39
ACM test are modified to run with both ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
22a2893
Fix include of isolate.gypi
by kjellander@webrtc.org
· 11 years ago
1b59234
Android OpenSL: Fixes faulty assertion in jni-code.
by henrike@webrtc.org
· 11 years ago
a6063fd
Remove ReturnTrace from DeregisterCallback().
by pbos@webrtc.org
· 11 years ago
59e1db1
Remove templatization of the AudioVector test
by henrik.lundin@webrtc.org
· 11 years ago
369da50
Workaround issue with stdin on Windows.
by kjellander@webrtc.org
· 11 years ago
6583dff
APK for opensl loopback.
by henrike@webrtc.org
· 11 years ago
b5d2d16
Implement TraceCallbacks in Call.
by pbos@webrtc.org
· 11 years ago
39079d1
Piping AutoMuter interface through to ViE API
by henrik.lundin@webrtc.org
· 11 years ago
d6da239
Added support for sending and receiving RTCP XR packets:
by asapersson@webrtc.org
· 11 years ago
69598c5
Stop timer in ~EventWindows().
by pbos@webrtc.org
· 11 years ago
053d45a
Update sampling rate and number of channels of NetEq4 if decoder is changed.
by turaj@webrtc.org
· 11 years ago
c5080a9
Test multiple send/receive streams in Call.
by pbos@webrtc.org
· 11 years ago
b9421ac
Remove include_dirs from utility.
by pbos@webrtc.org
· 11 years ago
b82f683
PeerConnection(Android): enable tracing to logcat.
by fischman@webrtc.org
· 11 years ago
2934af5
Reset audio bufer if codec changes, b/10835525.
by turaj@webrtc.org
· 11 years ago
37da9ab
Ensure adjusted "known delay" doesn't drop below zero.
by andrew@webrtc.org
· 11 years ago
0e9c399
NetEq4: Removing templatization for AudioMultiVector
by henrik.lundin@webrtc.org
· 11 years ago
24f0702
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
d4e1329
Remove include_dirs from video_render.
by pbos@webrtc.org
· 11 years ago
76a6ffb
Remove include_dirs from video_capture.
by pbos@webrtc.org
· 11 years ago
0d4d51b
Revert 4876 "Support for CELT in NetEq4."
by tina.legrand@webrtc.org
· 11 years ago
76238f6
Propagate AutoMuter interface out to VideoCodingModule
by henrik.lundin@webrtc.org
· 11 years ago
0de0049
1. adding request of ACM version in the manual mode of voe_auto_test
by minyue@webrtc.org
· 11 years ago
cd5c882
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
2b35b95
Change the parameters of calculating maximum decode time.
by wuchengli@chromium.org
· 11 years ago
a3a3a0f
Makes OpensSL default audio implementation/device on Android.
by henrike@webrtc.org
· 11 years ago
9e035d2
Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from
by wu@webrtc.org
· 11 years ago
b503d1e
Only use -lm on Linux in ISAC.
by andrew@webrtc.org
· 11 years ago
362e3e5
Remove test parameters from CallTest.
by pbos@webrtc.org
· 11 years ago
424e0e4
With ACM2 and NetEq4, VoE fuzz test very often fails.
by minyue@webrtc.org
· 11 years ago
cdc5e6a
Remove include_dirs from tools.
by pbos@webrtc.org
· 11 years ago
b655adf
Remove include_dirs from test.
by pbos@webrtc.org
· 11 years ago
44f030c
Implemented AutoMuter in MediaOptimization
by henrik.lundin@webrtc.org
· 11 years ago
dc1f7e9
Remove include_dirs from pacing.
by pbos@webrtc.org
· 11 years ago
ee817d3
Remove include_dirs from remote_bitrate_estimator.
by pbos@webrtc.org
· 11 years ago
fc75214
Remove include_dirs from bitrate_controller.
by pbos@webrtc.org
· 11 years ago
1fc4659
Remove include_dirs from video_coding.
by pbos@webrtc.org
· 11 years ago
85592ad
Remove include_dirs from video_processing.
by pbos@webrtc.org
· 11 years ago
1800406
Remove include_dirs from rtp_rtcp.
by pbos@webrtc.org
· 11 years ago
2f0a942
Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3.
by turaj@webrtc.org
· 11 years ago
d4f6789
Move the Config DelayCorrection struct to audio_processing.h.
by andrew@webrtc.org
· 11 years ago
8ddec2c
Add an extended filter mode to AEC.
by andrew@webrtc.org
· 11 years ago
53fdd3b
Fix WindowCapturerWin to capture window decorations after window size changes.
by sergeyu@chromium.org
· 11 years ago
605daf0
Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails.
by turaj@webrtc.org
· 11 years ago
72790c7
Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
by niklas.enbom@webrtc.org
· 11 years ago
f7d5a08
Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org
by elham@webrtc.org
· 11 years ago
7f35836
Re-enable verbose logging in NetEq4.
by turaj@webrtc.org
· 11 years ago
79c884c
Convert DeviceInfoImpl::_captureCapabilities from a map to a vector.
by fischman@webrtc.org
· 11 years ago
99b6d9e
Revert 4837 "Add an extended filter mode to AEC."
by asapersson@webrtc.org
· 11 years ago
83c5f62
Add an extended filter mode to AEC.
by andrew@webrtc.org
· 11 years ago
933267f
Small fixes to run ACM2 tests.
by turaj@webrtc.org
· 11 years ago
6ca9e7d
API add to set background noise mode.
by turaj@webrtc.org
· 11 years ago
08099e0
Fix window capturer not to leak HDC.
by sergeyu@chromium.org
· 11 years ago
82707bf
Fix window capturer to stop capturing when the target is minimized.
by sergeyu@chromium.org
· 11 years ago
4b067da
Disable some VP8 tests on Android.
by andrew@webrtc.org
· 11 years ago
8da2f65
Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
by henrika@webrtc.org
· 11 years ago
3bd659f
Add libjingle_peerconnection_objc_test to buildbot_tests.py
by kjellander@webrtc.org
· 11 years ago
a89f7e8
Revert r4823 "Reenable test and remove flaky expects."
by stefan@webrtc.org
· 11 years ago
890706b
Reenable test and remove flaky expects.
by stefan@webrtc.org
· 11 years ago
da6d2a2
MediaOptimization: Converting a few members to scoped_ptrs
by henrik.lundin@webrtc.org
· 11 years ago
b0382ea
Disable flaky RunsRtpRtcpTestWithoutErrors.
by andrew@webrtc.org
· 11 years ago
510ee1b
Remove deprecated AudioCodingModule::Destroy.
by andrew@webrtc.org
· 11 years ago
ae14504
- Reset capture deltas at resolution change.
by asapersson@webrtc.org
· 11 years ago
a6665e7
Reformatting media_optimization.cc and .h
by henrik.lundin@webrtc.org
· 11 years ago
36441e3
Re-enable VideoCaptureTest.CreateDelete
by fischman@webrtc.org
· 11 years ago
3b6d2d4
Updated WebRTC version to 3.42
by elham@webrtc.org
· 11 years ago
84afa19
Adding unit tests for default temporal layer strategy.
by andresp@webrtc.org
· 11 years ago
199555c
Revert test change in r4808.
by stefan@webrtc.org
· 11 years ago
d704640
Reduce flakiness in network down test.
by stefan@webrtc.org
· 11 years ago
2529558
Enable SetInitialPlayoutDelay on Android.
by dwkang@webrtc.org
· 11 years ago
d1fe828
Fix bugs in DesktopRegion::Subtract().
by sergeyu@chromium.org
· 11 years ago
717267a
VAD changes ported to ACM2.
by turaj@webrtc.org
· 11 years ago
045e45e
Address Windows 64-bits warnings.
by turaj@webrtc.org
· 11 years ago
0011252
Enable FEC for VideoSendStream.
by pbos@webrtc.org
· 11 years ago
54f0246
Disable flaky video capture test.
by stefan@webrtc.org
· 11 years ago
51d53aa
Avoid recursively taking critical section.
by stefan@webrtc.org
· 11 years ago
7ab577d
Use link_settings instead of all_dependent_settings to pacify xcode gyp generator
by fischman@webrtc.org
· 11 years ago
6876512
Roll webrtc's chromium_revision 217707:224141
by fischman@webrtc.org
· 11 years ago
28a1166
Rename EngineTest to CallTest.
by pbos@webrtc.org
· 11 years ago
f5013c0
Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct
by tina.legrand@webrtc.org
· 11 years ago
28631e7
Refactor frame generation code so it can be used by multiple modules.
by andresp@webrtc.org
· 11 years ago
a89566f
Disable NACK bandwidth statistics test due to being too flaky.
by stefan@webrtc.org
· 11 years ago
93b9912
Fixes a flake in network down tests.
by stefan@webrtc.org
· 11 years ago
032f731
Disable tests for TSan v2
by kjellander@webrtc.org
· 11 years ago
4d08199
Compile ACM2 and ACM1.
by turaj@webrtc.org
· 11 years ago
80142aa
Small refactoring of AudioProcessing use in channel.cc.
by andrew@webrtc.org
· 11 years ago
ab34f11
NetEq4: Making a few more members scoped_ptrs
by henrik.lundin@webrtc.org
· 11 years ago
05dd6c0
Dedicated speed test for NetEq3
by henrik.lundin@webrtc.org
· 11 years ago
c61a170
MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file.
by andrew@webrtc.org
· 11 years ago
ec09fcb
Revert r4772 "Compile ACM1 and ACM2."
by stefan@webrtc.org
· 11 years ago
671d90b
NetEq4: Make some DSP operation classes member variables
by henrik.lundin@webrtc.org
· 11 years ago
c2c8e6a
Fix races in vcm::Process().
by stefan@webrtc.org
· 11 years ago
1ddd57f
Break out glue for old->new Transport.
by pbos@webrtc.org
· 11 years ago
5b7878f
Changing 'frame' method to 'bounds' method.
by sjlee@webrtc.org
· 11 years ago
7556d2d
Compile ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
0c57671
Use the native sample rate for OpenSL recording.
by henrike@webrtc.org
· 11 years ago
39e22a1
Adds a new voice engine warning for the typing noise off state.
by jiayl@webrtc.org
· 11 years ago
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