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voice_engine
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Author
2014-05-21
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
mcasas@webrtc.org
2014-05-20
Calculate capture ntp timestamp in local timebase for decoded audio frame.
wu@webrtc.org
2014-05-20
Switch to using base/constructormagic.h and remove system_wrappers/interface/...
henrike@webrtc.org
2014-05-20
VoEVolumeTest: Enabled Linux flaky tests
bjornv@webrtc.org
2014-05-20
Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size
minyue@webrtc.org
2014-05-19
Add interface to propagate audio capture timestamp to the renderer.
wu@webrtc.org
2014-05-16
Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*.
solenberg@webrtc.org
2014-05-14
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
andrew@webrtc.org
2014-05-14
Add webrtc field trials API.
andresp@webrtc.org
2014-05-14
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation ...
henrika@webrtc.org
2014-05-14
Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-14
VoEVolumeTest: Adds error return tests.
bjornv@webrtc.org
2014-05-14
Make vie/voe_auto_test accept non-supported flags without error.
kjellander@webrtc.org
2014-05-13
Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255
bjornv@webrtc.org
2014-05-12
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation ...
henrika@webrtc.org
2014-05-12
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-12
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operat...
henrika@webrtc.org
2014-05-09
Removes parts of the webrtc::VoEHardware sub API (relanding)
henrika@webrtc.org
2014-05-09
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
henrika@webrtc.org
2014-05-09
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up o...
henrika@webrtc.org
2014-05-05
Allow the RTP level indicator computation to work at any sample rate.
andrew@webrtc.org
2014-05-02
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupporte...
henrika@webrtc.org
2014-04-30
Only clamp to 16 kHz when AECM is enabled.
andrew@webrtc.org
2014-04-25
Replace scoped_array<T> with scoped_ptr<T[]>.
andrew@webrtc.org
2014-04-24
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
wu@webrtc.org
2014-04-23
Remove ACM1/ACM2 switching from VoiceEngine tests
henrik.lundin@webrtc.org
2014-04-22
Support arbitrary input/output rates and downmixing in AudioProcessing.
andrew@webrtc.org
2014-04-22
Reland "Stop using ACM factory in VoiceEngine"
henrik.lundin@webrtc.org
2014-04-22
Revert "Stop using ACM factory in VoiceEngine"
henrik.lundin@webrtc.org
2014-04-22
Stop using ACM factory in VoiceEngine
henrik.lundin@webrtc.org
2014-04-22
Reland "Make VoiceEngine choose ACM2 by default""
henrik.lundin@webrtc.org
2014-04-19
Resampler modifications in preparation for arbitrary audioproc rates.
andrew@webrtc.org
2014-04-17
Removes parts of the VoEBase sub API as part of a clean-up operation where th...
henrika@webrtc.org
2014-04-17
Removes VoECodec sub API as part of a clean-up operation where the goal is to...
henrika@webrtc.org
2014-04-17
Revert "Make VoiceEngine choose ACM2 by default"
henrik.lundin@webrtc.org
2014-04-17
Make VoiceEngine choose ACM2 by default
henrik.lundin@webrtc.org
2014-04-16
Re-enable AGC tests:
aluebs@webrtc.org
2014-04-15
iOS: baby steps to being able to include_tests=1
fischman@webrtc.org
2014-04-15
Moved voe_neteq_stats_unittest to audio_coding_module_unittest
henrik.lundin@webrtc.org
2014-04-14
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
fischman@webrtc.org
2014-04-14
Removes VoECallReport sub API as part of a clean-up operation where the goal ...
henrika@webrtc.org
2014-04-14
Added a new OnMoreData() interface which will not feed the playout data to APM.
xians@webrtc.org
2014-04-09
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
henrika@webrtc.org
2014-04-08
Move output_mixer_unittest.cc to utility_unittest.cc.
andrew@webrtc.org
2014-04-08
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
andresp@webrtc.org
2014-04-03
Consolidate audio conversion from Channel and TransmitMixer.
andrew@webrtc.org
2014-04-02
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
andrew@webrtc.org
2014-04-02
Extends max sample rate from 96kHz to 192kHz on the input side.
henrika@webrtc.org
2014-04-01
Make ACM2 the default in voe_cmd_test.
andrew@webrtc.org
2014-04-01
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
fischman@webrtc.org
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