aboutsummaryrefslogtreecommitdiff
path: root/webrtc/modules/audio_processing/audio_processing_impl.cc
AgeCommit message (Expand)Author
2016-01-13Convert channel counts to size_t.Peter Kasting
2016-01-12Remove additional channel constraints when Beamforming is enabled in AudioPro...aluebs
2016-01-12Make Beamforming dynamically settable for Android platform buildsaluebs
2015-12-21Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that th...asapersson
2015-12-19Revert of Reland "Added option to specify a maximum file size when recording ...ivoc
2015-12-19Reland "Added option to specify a maximum file size when recording an AEC dum...ivoc
2015-12-18Revert of Added option to specify a maximum file size when recording an AEC d...ivoc
2015-12-18Added option to specify a maximum file size when recording an AEC dump.ivoc
2015-12-17Adding trace events for the APM render and capture stream processing functions.peah
2015-12-16Make VoiceDetection not a ProcessingComponent (bit exact).solenberg
2015-12-16Bugfix that fixes the error where the audio processing module is calledpeah
2015-12-15Make LevelEstimation not a ProcessingComponent.solenberg
2015-12-08Make NoiseSuppression not a processing component (bit exact).solenberg
2015-12-08Make HighPassFilter not a ProcessingComponent anymore (bit exact).solenberg
2015-11-28Introduced the new locking schemepeah
2015-11-27Added a threadchecking scheme to APM that checks that the APM API calls are c...peah
2015-11-17Preparational work before introducing the locks in order to harmonize the code:peah
2015-11-17Applied the render queueing to the agc.peah
2015-11-17Introduced the render sample queue for the aec and aecm.peah
2015-11-04modules: more interface -> include renamesHenrik Kjellander
2015-10-30Make the nonlinear beamformer steerableAlejandro Luebs
2015-10-28system_wrappers: rename interface -> includeHenrik Kjellander
2015-10-02Adding APM configuration in AEC dump.Minyue
2015-10-02Removed unused API functions in AudioProcessing and AudioProcessingModulepeah
2015-09-23Fix the maximum native sample rate in AudioProcessingAlejandro Luebs
2015-08-24Update a ton of audio code to use size_t more correctly and in general reducePeter Kasting
2015-08-14Integrate Intelligibility with APMekmeyerson
2015-08-10Fix data race in AMP.Michael Graczyk
2015-07-27Clean up the Config to enable 48kHz support in AudioProcessingaluebs
2015-07-23Allow more than 2 input channels in AudioProcessing.Michael Graczyk
2015-07-23Revert of Allow more than 2 input channels in AudioProcessing. (patchset #13 ...magjed
2015-07-23Allow more than 2 input channels in AudioProcessing.Michael Graczyk
2015-07-07audio_processing: Adds two UMA histograms logging delay jumps in AECBjorn Volcker
2015-07-05audio_processing: Changed kMinDiffDelayMs from 50 to 60 msBjorn Volcker
2015-06-29Adds UMA histogram for system delay jumpsBjorn Volcker
2015-06-25Ensure transient suppression is never enabled on mobile.andrew
2015-05-20Replace assert() with static_assert() if the condition is evaluatable atAndré Susano Pinto
2015-04-15audio_processing/agc: Adds config to set minimum microphone volume at startupBjorn Volcker
2015-03-27audio_processing/agc: Put entire method set_output_will_be_muted() under lockBjorn Volcker
2015-03-25Reparent Nonlinear beamformer under beamforming interface.Michael Graczyk
2015-03-20Remove build-time beamformer flags.andrew@webrtc.org
2015-03-13Rename Beamformer to NonlinearBeamformer.mgraczyk@chromium.org
2015-03-12Clean up LappedTransform and Blocker.mgraczyk@chromium.org
2015-03-04Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macrokjellander@webrtc.org
2015-03-02Add Config option to enable 48kHz support in AudioProcessingaluebs@webrtc.org
2015-02-26Add 48kHz support to Beamformeraluebs@webrtc.org
2015-02-10Make ChannelBuffer aware of frequency bandsaluebs@webrtc.org
2015-02-06audio_processing: Now records mic volume level also when using new AGCbjornv@webrtc.org
2015-01-28Move channel_buffer.{h,cc} to common_audio.kjellander@webrtc.org
2015-01-15Only adapt AGC when the desired signal is presentaluebs@webrtc.org
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-27 05:59:20 +0000