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2014-05-06Update makefiles after merge of Chromium at 268379Android Chromium Automerger
2014-04-30Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-28Update makefiles after merge of Chromium at 266543Android Chromium Automerger
2014-04-25Replace scoped_array<T> with scoped_ptr<T[]>.andrew@webrtc.org
2014-04-25Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-24* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.wu@webrtc.org
2014-04-23Update makefiles after merge of Chromium at 265680Android Chromium Automerger
2014-04-23Update makefiles after merge of Chromium at 265607Android Chromium Automerger
2014-04-23Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-23Remove ACM1/ACM2 switching from VoiceEngine testshenrik.lundin@webrtc.org
2014-04-22Support arbitrary input/output rates and downmixing in AudioProcessing.andrew@webrtc.org
2014-04-22Reland "Stop using ACM factory in VoiceEngine"henrik.lundin@webrtc.org
2014-04-22Revert "Stop using ACM factory in VoiceEngine"henrik.lundin@webrtc.org
2014-04-22Stop using ACM factory in VoiceEnginehenrik.lundin@webrtc.org
2014-04-22Reland "Make VoiceEngine choose ACM2 by default""henrik.lundin@webrtc.org
2014-04-19Resampler modifications in preparation for arbitrary audioproc rates.andrew@webrtc.org
2014-04-17Removes parts of the VoEBase sub API as part of a clean-up operation where th...henrika@webrtc.org
2014-04-17Removes VoECodec sub API as part of a clean-up operation where the goal is to...henrika@webrtc.org
2014-04-17Revert "Make VoiceEngine choose ACM2 by default"henrik.lundin@webrtc.org
2014-04-17Make VoiceEngine choose ACM2 by defaulthenrik.lundin@webrtc.org
2014-04-16Re-enable AGC tests:aluebs@webrtc.org
2014-04-15iOS: baby steps to being able to include_tests=1fischman@webrtc.org
2014-04-15Moved voe_neteq_stats_unittest to audio_coding_module_unittesthenrik.lundin@webrtc.org
2014-04-14Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.fischman@webrtc.org
2014-04-14Removes VoECallReport sub API as part of a clean-up operation where the goal ...henrika@webrtc.org
2014-04-14Added a new OnMoreData() interface which will not feed the playout data to APM.xians@webrtc.org
2014-04-10Update makefiles after merge of Chromium at 263101Android Chromium Automerger
2014-04-10Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-09Update makefiles after merge of Chromium at 262754Android Chromium Automerger
2014-04-09(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome buildshenrika@webrtc.org
2014-04-08Move output_mixer_unittest.cc to utility_unittest.cc.andrew@webrtc.org
2014-04-08Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.andresp@webrtc.org
2014-04-07Update makefiles after merge of Chromium at 262110Android Chromium Automerger
2014-04-04Update makefiles after merge of Chromium at 261622Android Chromium Automerger
2014-04-03Consolidate audio conversion from Channel and TransmitMixer.andrew@webrtc.org
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
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