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webrtc_m130/modules/audio_processing
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Minyue Li 656d609a95 Add UTC time to init event in AEC debug dump. 
Bug: webrtc:9616
Change-Id: I1350212f0b8835fb64427483269da96d51670c01
Reviewed-on: https://webrtc-review.googlesource.com/92620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-11 20:29:07 +00:00
..
aec
Explicitly add -mfpu=neon to all targets that use NEON
2018-08-01 13:15:42 +00:00
aec3
AEC3: Allow the main and shadow filters to have different lengths
2018-08-10 19:59:50 +00:00
aec_dump
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
aecm
Explicitly add -mfpu=neon to all targets that use NEON
2018-08-01 13:15:42 +00:00
agc
Optionally disable digital gain control in ExperimentalAgc.
2018-08-09 13:37:30 +00:00
agc2
Reset Agc2 on analog gain changes.
2018-08-08 14:36:37 +00:00
audio_generator
Add stub draft of audio generator to APM
2018-03-05 09:28:52 +00:00
echo_detector
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
include
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
intelligibility
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
logging
Remove stringstream usages from the APM
2018-04-06 14:17:03 +00:00
ns
Move fft4g to proper third_party directory
2018-07-25 15:44:53 +00:00
test
Optionally disable digital gain control in ExperimentalAgc.
2018-08-09 13:37:30 +00:00
transient
Fix MovingMoments::CalculateMoments.
2018-07-31 15:08:12 +00:00
utility
Explicitly add -mfpu=neon to all targets that use NEON
2018-08-01 13:15:42 +00:00
vad
Move fft4g to proper third_party directory
2018-07-25 15:44:53 +00:00
audio_buffer_unittest.cc
…
audio_buffer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_buffer.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
audio_frame_view_unittest.cc
Add namespace 'webrtc' to AudioFrameView.
2018-05-14 12:33:49 +00:00
audio_processing_impl_locking_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_processing_impl_unittest.cc
APM: render pre-processor moved before echo detector queuing.
2018-08-09 14:40:31 +00:00
audio_processing_impl.cc
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
audio_processing_impl.h
Revert "Add one-stop-shop for built-in AEC toggling in APM"
2018-07-23 14:48:17 +00:00
audio_processing_performance_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_processing_unittest.cc
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
BUILD.gn
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
common.h
…
config_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
debug.proto
Add UTC time to init event in AEC debug dump.
2018-08-11 20:29:07 +00:00
DEPS
…
echo_cancellation_bit_exact_unittest.cc
…
echo_cancellation_impl_unittest.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +00:00
echo_cancellation_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
echo_cancellation_impl.h
Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS
2017-12-22 15:42:13 +00:00
echo_control_mobile_impl.cc
Turn off comfort noise generation by default in AECM
2018-07-24 08:52:36 +00:00
echo_control_mobile_impl.h
…
echo_control_mobile_unittest.cc
…
gain_control_for_experimental_agc.cc
Reset Agc2 on analog gain changes.
2018-08-08 14:36:37 +00:00
gain_control_for_experimental_agc.h
Reset Agc2 on analog gain changes.
2018-08-08 14:36:37 +00:00
gain_control_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
gain_control_impl.h
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
gain_control_unittest.cc
…
gain_controller2_unittest.cc
Set a positive initial gain in the Adaptive Digital GC.
2018-04-27 09:05:25 +00:00
gain_controller2.cc
Reset Agc2 on analog gain changes.
2018-08-08 14:36:37 +00:00
gain_controller2.h
Reset Agc2 on analog gain changes.
2018-08-08 14:36:37 +00:00
level_estimator_impl.cc
…
level_estimator_impl.h
…
level_estimator_unittest.cc
…
low_cut_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
low_cut_filter.cc
…
low_cut_filter.h
…
noise_suppression_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
noise_suppression_impl.h
…
noise_suppression_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Adding alessiob@ and minyue@ as owners of APM.
2018-07-02 07:45:31 +00:00
render_queue_item_verifier.h
…
residual_echo_detector_unittest.cc
Change echo detector to scoped_refptr
2018-06-14 09:51:41 +00:00
residual_echo_detector.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
residual_echo_detector.h
Add more parameters to the Initialize function of the echo detector.
2018-03-15 09:21:56 +00:00
rms_level_unittest.cc
…
rms_level.cc
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
rms_level.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
splitting_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
splitting_filter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
splitting_filter.h
…
three_band_filter_bank.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
three_band_filter_bank.h
…
typing_detection.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
typing_detection.h
Delete root header file typedef.h.
2018-07-25 14:59:26 +00:00
voice_detection_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
voice_detection_impl.h
…
voice_detection_unittest.cc
…