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AgeCommit message (Expand)Author
2014-05-06Update makefiles after merge of Chromium at 268379Android Chromium Automerger
2014-04-30Update makefiles after merge of Chromium at 267293Android Chromium Automerger
2014-04-30Merge third_party/webrtc from https://chromium.googlesource.com/external/webr...Android Chromium Automerger
2014-04-29Update makefiles after merge of Chromium at 266947Android Chromium Automerger
2014-04-29Fix a data race in ACM1 when audio is pulled.turaj@webrtc.org
2014-04-29Added include of assert.h for files calling assert but missing the include.henrike@webrtc.org
2014-04-29Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()henrike@webrtc.org
2014-04-29Disable failing GoogleWifiTrace3Mbps.pbos@webrtc.org
2014-04-29Disable GoogleWifiTrace3Mbps.pbos@webrtc.org
2014-04-29Adding BweFeedbackTest which tracks BWE performance over a set of simulated s...stefan@webrtc.org
2014-04-29Fix iOS assembly compile error.kjellander@webrtc.org
2014-04-29Roll chromium_revision 260462:266514kjellander@webrtc.org
2014-04-29Remove Version method from ACM1henrik.lundin@webrtc.org
2014-04-29Remove ACM1 and NetEq3 related targets from modules.gyphenrik.lundin@webrtc.org
2014-04-29Remove AudioCodingModuleFactoryhenrik.lundin@webrtc.org
2014-04-29Add clock to ACM config structhenrik.lundin@webrtc.org
2014-04-29AEC: Startup phase only runs if reported_delay_enabledbjornv@webrtc.org
2014-04-28APM: limit native sample rate to 16kHz on mobile.fischman@webrtc.org
2014-04-28Using realpath instead of android_src in Android webviewmichaelbai@google.com
2014-04-28Update makefiles after merge of Chromium at 266543Android Chromium Automerger
2014-04-28AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.hbjornv@webrtc.org
2014-04-28Disable capture test for FrameRate on Windows.pbos@webrtc.org
2014-04-28Introduce a config struct for AudioCoding modulehenrik.lundin@webrtc.org
2014-04-28Fix the NetEq buildhenrik.lundin@webrtc.org
2014-04-28Include buffer size limits in NetEq config structhenrik.lundin@webrtc.org
2014-04-28Add henrik.lundin as owner in AudioCoding modulehenrik.lundin@webrtc.org
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
2014-04-25Fix leak in remote bitrate estimator tests introduced in r5980stefan@webrtc.org
2014-04-25Support for simulating multiple independent flows in a network.stefan@webrtc.org
2014-04-24Returns a NULL frame on all platforms if the captured window is closed.jiayl@webrtc.org
2014-04-24Calculate local/remote clock delta and capture ntp timestamp in receiver's ti...wu@webrtc.org
2014-04-24* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.wu@webrtc.org
2014-04-24Add an output capacity parameter to ACMResampler::Resample10Msec()henrik.lundin@webrtc.org
2014-04-24Add keyboard channel support to AudioBuffer.andrew@webrtc.org
2014-04-24Fix the Android compilation (better structure for NetEq test libs)henrik.lundin@webrtc.org
2014-04-24Fixing a bug in ACM2 where the output frame energy was incorrectly sethenrik.lundin@webrtc.org
2014-04-24Use unique filenames in AudioProcessingTests for parallelization.andrew@webrtc.org
2014-04-23Update makefiles after merge of Chromium at 265680Android Chromium Automerger
2014-04-23AEC: Adds a reported_delay_enabled_ flagbjornv@webrtc.org
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
2014-04-23Restore sample_rate_hz() until Chromium is updated to not use it.andrew@webrtc.org
2014-04-22Support arbitrary input/output rates and downmixing in AudioProcessing.andrew@webrtc.org
2014-04-22Remove 44.1 kHz workaround from the iOS AudioDevice.andrew@webrtc.org
2014-04-22Fix a bug in AcmReceiver::NetworkStatisticshenrik.lundin@webrtc.org
2014-04-22Create ACM2 instance when calling AudioCodingModule::Createhenrik.lundin@webrtc.org
2014-04-22audio_processing: DestroyHandle() now returns voidbjornv@webrtc.org
2014-04-22common_audio: VADFree() now returns voidbjornv@webrtc.org
2014-04-19Resampler modifications in preparation for arbitrary audioproc rates.andrew@webrtc.org
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