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voice_engine
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Author
2014-02-26
Update makefiles after merge of Chromium at 253634
Android Chromium Automerger
2014-02-25
Update makefiles after merge of Chromium at 253112
Android Chromium Automerger
2014-02-12
Remove unused and not working voe_extended_test.
solenberg@webrtc.org
2014-02-11
Reduce mixing threshold in test to avoid flakiness.
andrew@webrtc.org
2014-02-11
Add an interface for accepting keypress signals to AudioProcessing.
andrew@webrtc.org
2014-02-10
Restore mixing integration tests.
andrew@webrtc.org
2014-02-04
Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelepho...
pbos@webrtc.org
2014-02-04
Fix locking in LoopBackTransport::StorePacket.
pbos@webrtc.org
2014-02-03
Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents
marpan@webrtc.org
2014-02-02
Moved the new OnData interface to AudioTranport, and expose the AudioTranspor...
xians@webrtc.org
2014-01-30
Move out typing detection to its own class.
henrikg@webrtc.org
2014-01-29
Added new capture callback interface to pass the capture callback to a specif...
xians@webrtc.org
2014-01-21
Output logs to stderr from voe_cmd_test by default.
andrew@webrtc.org
2014-01-14
Temporarily disabling some more audio processing tests.
aluebs@webrtc.org
2014-01-11
Minor voice engine improvements around AGC.
andrew@webrtc.org
2014-01-10
Android: Fixes crash when exiting WebRTCDemo.
henrike@webrtc.org
2014-01-07
Remove the requirement to call set_sample_rate_hz and friends.
andrew@webrtc.org
2013-12-24
Fix the include guard in transmit_mixer.h
braveyao@webrtc.org
2013-12-24
Fix the include guard in transmit_mixer.h
braveyao@webrtc.org
2013-12-19
Add callbacks for receive channel RTCP statistics.
sprang@webrtc.org
2013-12-13
Disabled tests on Android. The issue 2723 is filed to investigate the reason ...
turaj@webrtc.org
2013-12-13
Fix jitter buffer delay estimate.
turaj@webrtc.org
2013-12-13
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
wu@webrtc.org
2013-12-11
Enables mixing and matching Java and native audio. It is used for getting bes...
henrike@webrtc.org
2013-12-11
Roll chromium_revision 232627:238260
kjellander@webrtc.org
2013-12-06
Allow opening an AEC dump from an existing file handle.
henrikg@webrtc.org
2013-12-04
Remove the long disabled WEBRTC_SVNREVISION define.
andrew@webrtc.org
2013-11-20
Fixes a crash in VoE when unregistering JNI hooks.
henrike@webrtc.org
2013-11-19
Add experimental noise suppression dummy API.
aluebs@webrtc.org
2013-11-13
Inject config when creating channels to override the existing one.
turaj@webrtc.org
2013-11-08
Fix for making sure that the packet in order checks are done prior to updatin...
stefan@webrtc.org
2013-10-23
Fixing broken tests in voe_auto_test extended
tina.legrand@webrtc.org
2013-10-22
Upgrade scoped_ptr to Chromium's latest version.
andrew@webrtc.org
2013-10-18
Protect _transportPtr, which can be accessed by different threads in the case...
wu@webrtc.org
2013-10-17
Fix tsan failures in channel.cc regarding to the volume settings.
wu@webrtc.org
2013-10-17
Check the number of playout channels instead of the send channels in StopPlay...
xians@webrtc.org
2013-10-15
Roll chromium_revision 226126:228675 and fix clang warnings
kjellander@webrtc.org
2013-10-04
Add APK and isolate target for video_engine_tests
kjellander@webrtc.org
2013-10-04
Clean up AudioProcessing defaults and errors.
andrew@webrtc.org
2013-10-02
Fix include of isolate.gypi
kjellander@webrtc.org
2013-09-30
1. adding request of ACM version in the manual mode of voe_auto_test
minyue@webrtc.org
2013-09-24
Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket
henrika@webrtc.org
2013-09-23
Remove deprecated AudioCodingModule::Destroy.
andrew@webrtc.org
2013-09-23
Enable SetInitialPlayoutDelay on Android.
dwkang@webrtc.org
2013-09-18
Small refactoring of AudioProcessing use in channel.cc.
andrew@webrtc.org
2013-09-17
Adds a new voice engine warning for the typing noise off state.
jiayl@webrtc.org
2013-09-12
This issue is related to
minyue@webrtc.org
2013-09-10
OpenSL (not default): Enables low latency audio on Android.
henrike@webrtc.org
2013-09-10
Remove include_dirs from voice_engine.gyp.
pbos@webrtc.org
2013-09-09
Remove redundant STR_CASE_CMP macro definitions.
andrew@webrtc.org
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