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2014-04-04Update makefiles after merge of Chromium at 261622Android Chromium Automerger
2014-04-02Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-02Remove AudioDevice::{Microphone,Speaker}IsAvailable.andrew@webrtc.org
2014-04-02Extends max sample rate from 96kHz to 192kHz on the input side.henrika@webrtc.org
2014-04-01Make ACM2 the default in voe_cmd_test.andrew@webrtc.org
2014-04-01Update makefiles after merge of Chromium at 260927Android Chromium Automerger
2014-04-01VoiceEngine(iOS & Android): removed NOT_SUPPORTEDfischman@webrtc.org
2014-03-31VoE Channel: Don't register codecs when stopping receiverhenrik.lundin@webrtc.org
2014-03-25Make RTPHeaderParser skip over unknown RTP header extensions rather than bail...solenberg@webrtc.org
2014-03-24VoE changes to allow forwarding of packets from VoE to ViE BWE.solenberg@webrtc.org
2014-03-21Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.pbos@webrtc.org
2014-03-21Adding operator== and != methods for CodecInst and VideoCodec structures.mallinath@webrtc.org
2014-03-20Prevent playout delay wrap-around in VoiceEnginehenrik.lundin@webrtc.org
2014-03-20Removes error printout in voe_cmd_test which was caused by attempts to transm...henrika@webrtc.org
2014-03-18Resolves TSan v2 warnings in voe_auto_test.henrika@webrtc.org
2014-03-11Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instea...braveyao@webrtc.org
2014-03-06Help to land 7969005 on behalf of solenberg. The review and try is done in 79...wu@webrtc.org
2014-02-19Removes VoERTP_RTCP::InsertExtraRTPPacket.henrika@webrtc.org
2014-02-18Move the volume quantization workaround from VoE to AGC.andrew@webrtc.org
2014-02-18Remove obsolete voe_unit_test.solenberg@webrtc.org
2014-02-18Remove external encryption API for VoE.solenberg@webrtc.org
2014-02-12Remove unused and not working voe_extended_test.solenberg@webrtc.org
2014-02-11Reduce mixing threshold in test to avoid flakiness.andrew@webrtc.org
2014-02-11Add an interface for accepting keypress signals to AudioProcessing.andrew@webrtc.org
2014-02-10Restore mixing integration tests.andrew@webrtc.org
2014-02-04Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelepho...pbos@webrtc.org
2014-02-04Fix locking in LoopBackTransport::StorePacket.pbos@webrtc.org
2014-02-03Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEventsmarpan@webrtc.org
2014-02-02Moved the new OnData interface to AudioTranport, and expose the AudioTranspor...xians@webrtc.org
2014-01-30Move out typing detection to its own class.henrikg@webrtc.org
2014-01-29Added new capture callback interface to pass the capture callback to a specif...xians@webrtc.org
2014-01-21Output logs to stderr from voe_cmd_test by default.andrew@webrtc.org
2014-01-14Temporarily disabling some more audio processing tests.aluebs@webrtc.org
2014-01-11Minor voice engine improvements around AGC.andrew@webrtc.org
2014-01-10Android: Fixes crash when exiting WebRTCDemo.henrike@webrtc.org
2014-01-07Remove the requirement to call set_sample_rate_hz and friends.andrew@webrtc.org
2013-12-24Fix the include guard in transmit_mixer.hbraveyao@webrtc.org
2013-12-24Fix the include guard in transmit_mixer.hbraveyao@webrtc.org
2013-12-19Add callbacks for receive channel RTCP statistics.sprang@webrtc.org
2013-12-13Disabled tests on Android. The issue 2723 is filed to investigate the reason ...turaj@webrtc.org
2013-12-13Fix jitter buffer delay estimate.turaj@webrtc.org
2013-12-13Update talk to 58174641 together with http://review.webrtc.org/4319005/.wu@webrtc.org
2013-12-11Enables mixing and matching Java and native audio. It is used for getting bes...henrike@webrtc.org
2013-12-11Roll chromium_revision 232627:238260kjellander@webrtc.org
2013-12-06Allow opening an AEC dump from an existing file handle.henrikg@webrtc.org
2013-12-04Remove the long disabled WEBRTC_SVNREVISION define.andrew@webrtc.org
2013-11-20Fixes a crash in VoE when unregistering JNI hooks.henrike@webrtc.org
2013-11-19Add experimental noise suppression dummy API.aluebs@webrtc.org
2013-11-13Inject config when creating channels to override the existing one.turaj@webrtc.org
2013-11-08Fix for making sure that the packet in order checks are done prior to updatin...stefan@webrtc.org
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