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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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f9d3555ec384a4ed428a114f5fd33abaefce30c6
f9d3555
Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
by phoglund@webrtc.org
· 10 years ago
ce3ac53
Adding TRYSERVER_PROJECT to codereview.settings.
by kjellander@webrtc.org
· 10 years ago
018c087
Add /talk/examples/androidtests/{bin,gen} to .gitignore.
by kjellander@webrtc.org
· 10 years ago
a32d154
Disable tests failing on Android ARM64 (Nexus9).
by kjellander@webrtc.org
· 10 years ago
ff9462e
Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.
by sprang@webrtc.org
· 10 years ago
2624b1e
Remove unused private data member engine_id_
by tommi@webrtc.org
· 10 years ago
fe672e3
release the turn allocation by sending a refresh request with lifetime 0
by pthatcher@webrtc.org
· 10 years ago
d7de120
Re-enable the messagequeue unittests. These were commented out at one point but never reenabled.
by decurtis@webrtc.org
· 10 years ago
a1aea10
Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."
by stefan@webrtc.org
· 10 years ago
4ba1e44
Remove unnecessary remote bitrate estimator build rule which serves no purpose.
by andresp@webrtc.org
· 10 years ago
487a444
Add stats collection for the data channel.
by decurtis@webrtc.org
· 10 years ago
357469d
Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels.
by decurtis@webrtc.org
· 10 years ago
ef2a5dd
Update AppRTCDemo UI.
by tkchin@webrtc.org
· 10 years ago
64d3c4b
Support 48kHz in AEC
by aluebs@webrtc.org
· 10 years ago
89aa276
Fix a case where empty candidate id is used
by guoweis@webrtc.org
· 10 years ago
d82f55d
Only adapt AGC when the desired signal is present
by aluebs@webrtc.org
· 10 years ago
3e42a8a
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.
by stefan@webrtc.org
· 10 years ago
32e8528
Log configs when creating video streams in Call.
by pbos@webrtc.org
· 10 years ago
1f67b53
Remove dual stream functionality in ACM
by henrik.lundin@webrtc.org
· 10 years ago
9ce01e6
Clean unnecessary workaround for chromium import.
by andresp@webrtc.org
· 10 years ago
0800db7
Add percentage of fec packets and recovered media packets to histogram stats:
by asapersson@webrtc.org
· 10 years ago
61c1247
Fix a case where empty candidate id is used
by guoweis@webrtc.org
· 10 years ago
6c38552
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
by andrew@webrtc.org
· 10 years ago
5a92b78
Add beamforming to audioproc_float utility.
by mgraczyk@chromium.org
· 10 years ago
6b63015
Move ring_buffer to common_audio.
by andrew@webrtc.org
· 10 years ago
fd630a5
Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior.
by pthatcher@webrtc.org
· 10 years ago
693e01c
Fix searching for DirectX SDK during GN build.
by kjellander@webrtc.org
· 10 years ago
f1c8b90
Remove WebRtcVideoEncoderFactory2.
by pbos@webrtc.org
· 10 years ago
e5a31e1
Revert removing of compile_assert.h.
by turaj@webrtc.org
· 10 years ago
85fa94d
Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
by kjellander@webrtc.org
· 10 years ago
387841a
Improved fairness simulation by starting the flows 20 seconds apart.
by stefan@webrtc.org
· 10 years ago
f18fba2
Implement SimulcastEncoderAdapter support.
by pbos@webrtc.org
· 10 years ago
8315d7d
Remove dual stream functionality in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
b4e5d1b
Remove RTX SSRC when deleting the default receive stream.
by mflodman@webrtc.org
· 10 years ago
2ebfac5
Remove COMPILE_ASSERT and use static_assert everywhere
by kwiberg@webrtc.org
· 10 years ago
86e1e48
Move system_wrappers.gyp files to the proper directory.
by andresp@webrtc.org
· 10 years ago
a35f741
Add .classpath + talk/app/webrtc/androidtests to .gitignore
by kjellander@webrtc.org
· 10 years ago
f7a5893
Combine RegKeyTests to prevent parallel execution.
by pbos@webrtc.org
· 10 years ago
ef09092
No longer asserting in mocks, split first test case in two methods.
by phoglund@webrtc.org
· 10 years ago
69f4738
Roll chromium_revision 3dd2edf..a6eafec (310717:311223)
by kjellander@webrtc.org
· 10 years ago
d6e84d9
Always copy processed audio to output buffer in ProcessStream.
by mgraczyk@chromium.org
· 10 years ago
c0da63c
Optimize minimum delay in blocker
by aluebs@webrtc.org
· 10 years ago
af9d56f
Unify the two copies of template_util.h
by kwiberg@webrtc.org
· 10 years ago
0b0c241
Only return Rtx mode in RTXSendStatus().
by pbos@webrtc.org
· 10 years ago
3df38b4
Unify the two copies of compile_assert.h
by kwiberg@webrtc.org
· 10 years ago
58a1ba6
Roll chromium_revision 271c6cc..3dd2edf (309333:310717)
by kjellander@webrtc.org
· 10 years ago
46323b3
Remove useless AudioProcessing::Create() overload.
by andrew@webrtc.org
· 10 years ago
16825b1
Use int64_t more consistently for times, in particular for RTT values.
by pkasting@chromium.org
· 10 years ago
a7add19
audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter
by bjornv@webrtc.org
· 10 years ago
2a26734
Partial revert of r7396
by henrik.lundin@webrtc.org
· 10 years ago
be40eb0
Allow 720x1280 frames encoding on Android.
by glaznev@webrtc.org
· 10 years ago
a525c98
Fix parallelizability in ApmTests.
by pbos@webrtc.org
· 10 years ago
45db7ee
Use Java based audio as default for WebRTC.
by henrika@webrtc.org
· 10 years ago
81134d0
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory.
by perkj@webrtc.org
· 10 years ago
88a4298
common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin()
by bjornv@webrtc.org
· 10 years ago
c14e357
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
by bjornv@webrtc.org
· 10 years ago
19e4e8d
Add support for trying alternate server (STUN 300 error message) on TCP
by guoweis@webrtc.org
· 10 years ago
0ba1533
Added support for an Origin header in STUN messages.
by pthatcher@webrtc.org
· 10 years ago
2693a54
Add WEBRTC_BEAMFORMER define to BUILD.gn
by aluebs@webrtc.org
· 10 years ago
8f27fcc
Revert 8028 "Support associated payload type when registering Rt..."
by andrew@webrtc.org
· 10 years ago
80452d7
Sync Android AppRTCDemo with internal repo.
by glaznev@webrtc.org
· 10 years ago
9657265
Revert "Accept incoming pings before remote answer is set to reduce connection latency."
by pthatcher@webrtc.org
· 10 years ago
f3fd8e7
Add NEON intrinsics version for transform_neon
by andrew@webrtc.org
· 10 years ago
1592df7
PRESUBMIT: Add GN trybots for Windows and Mac.
by kjellander@webrtc.org
· 10 years ago
2a16964
Support associated payload type when registering Rtx payload type.
by pbos@webrtc.org
· 10 years ago
8649fed
GN: Fix Windows build.
by kjellander@webrtc.org
· 10 years ago
2ead571
Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
by decurtis@webrtc.org
· 10 years ago
758d6d4
audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16
by bjornv@webrtc.org
· 10 years ago
dec649c
audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with *
by bjornv@webrtc.org
· 10 years ago
5e5b327
audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc
by bjornv@webrtc.org
· 10 years ago
124b9c7
Suppress races in event tracing code.
by pbos@webrtc.org
· 10 years ago
5f09564
Suppress AsyncHttpRequestTest.TestCancel leak for LSan
by kjellander@webrtc.org
· 10 years ago
823c9b8
Add histograms stats for sent/received fraction loss for a stream:
by asapersson@webrtc.org
· 10 years ago
d730b28
Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon
by andrew@webrtc.org
· 10 years ago
59062d5
Rename SendAndReceiveH264SvcQqvga to VP8 instead.
by pbos@webrtc.org
· 10 years ago
8af1104
Avoid reading past end of string in GetLine.
by decurtis@webrtc.org
· 10 years ago
3663fb0
Reenable dlclose() for InternalUnloadDll on TSan.
by pbos@webrtc.org
· 10 years ago
bab7995
Convert FileMediaEngineTest to use more expects.
by pbos@webrtc.org
· 10 years ago
69472e7
Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS.
by pthatcher@webrtc.org
· 10 years ago
c10ecea
Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const.
by henrike@webrtc.org
· 10 years ago
dfef028
Ignore virtual box interfaces.
by pthatcher@webrtc.org
· 10 years ago
25dd754
Excluding a flaky test from DrMemory
by tina.legrand@webrtc.org
· 10 years ago
7fbf278
Suppress memcheck error in video_engine_tests
by kjellander@webrtc.org
· 10 years ago
1777880
Roll gtest-parallel.
by pbos@webrtc.org
· 10 years ago
07c83a1
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2)
by kjellander@webrtc.org
· 10 years ago
4e5115a
RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
by tkchin@webrtc.org
· 10 years ago
f6a9714
Remove peer connection and signaling calls from UI thread.
by glaznev@webrtc.org
· 10 years ago
2ec50f2
Memcheck suppression for uninitalized memory in WebRtcIsac_Decode
by kjellander@webrtc.org
· 10 years ago
d95435c
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win
by kjellander@webrtc.org
· 10 years ago
cbe7ca8
Roll chromium_revision 8e72e1d..271c6cc (307131:309333)
by kjellander@webrtc.org
· 10 years ago
3a63a3c
iOS AppRTC: First unit test.
by tkchin@webrtc.org
· 10 years ago
4796cb9
Disable flaky RelayServerTest.TestExpiration on all platforms.
by andrew@webrtc.org
· 10 years ago
fb7a039
Use array geometry in Beamformer
by aluebs@webrtc.org
· 10 years ago
a37bf2c
Hack clock_unittest fix for parallel execution.
by andrew@webrtc.org
· 10 years ago
c37e72e
Make setting identical RTP extensions a no-op.
by pbos@webrtc.org
· 10 years ago
e5a921a
Use tmp files in file_utils_unittests
by aluebs@webrtc.org
· 10 years ago
76bc981
Use a temp file in FileLockTest.
by pbos@webrtc.org
· 10 years ago
433006a
Fixed style issues from lint and got rid of unused fields.
by wzh@webrtc.org
· 10 years ago
c4ad157
Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9.
by marpan@webrtc.org
· 10 years ago
215bbbd
Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec.
by mflodman@webrtc.org
· 10 years ago
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