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webrtc_m130/modules/audio_processing
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Hanna Silen 7000b99384 Update GainController2 adaptive digital default parameters 
Update the default parameters to match with the Chromium parameters in
https://chromium-review.googlesource.com/c/chromium/src/+/5272859.

Bug: webrtc:7494
Change-Id: Id405109cca4adaa6d21945ed89f84d7e27ad361b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345923
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42048}
2024-04-12 08:29:26 +00:00
..
aec3
Remove expired WebRTC-Aec3DelayEstimatorDetectPreEcho
2024-04-11 09:57:16 +00:00
aec_dump
Remove dependency on rtc::TaskQueue in AudioProcessing module
2024-01-17 18:12:16 +00:00
aecm
Format /modules
2023-04-20 02:02:45 +00:00
agc
Format /modules
2023-04-20 02:02:45 +00:00
agc2
Update GainController2 adaptive digital default parameters
2024-04-12 08:29:26 +00:00
capture_levels_adjuster
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echo_detector
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g3doc
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
include
Update GainController2 adaptive digital default parameters
2024-04-12 08:29:26 +00:00
logging
Fix improper buffer size in call to rtc::strcpyn
2023-09-12 11:40:07 +00:00
ns
Format /modules
2023-04-20 02:02:45 +00:00
test
In AecDump take raw pointer to TaskQueueBase instead of legacy rtc::TaskQueue
2024-01-08 12:17:06 +00:00
transient
Fix missing libc++ includes in webrtc
2023-03-02 10:14:51 +00:00
utility
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vad
Fix downstream review comments for C++20
2023-07-04 09:06:07 +00:00
audio_buffer_unittest.cc
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audio_buffer.cc
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audio_buffer.h
APM: Signal error on unsupported sample rates
2022-11-17 12:12:04 +00:00
audio_frame_view_unittest.cc
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audio_processing_builder_impl.cc
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audio_processing_impl_locking_unittest.cc
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audio_processing_impl_unittest.cc
Move some users to use webrtc::RefCountInterface
2023-11-02 14:45:57 +00:00
audio_processing_impl.cc
Use num_output_channels() in GainController2
2024-02-12 11:29:20 +00:00
audio_processing_impl.h
Change AudioProcessing interface to allow not to require rtc::TaskQueue
2024-01-10 13:48:44 +00:00
audio_processing_performance_unittest.cc
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audio_processing_unittest.cc
In AecDump take raw pointer to TaskQueueBase instead of legacy rtc::TaskQueue
2024-01-08 12:17:06 +00:00
BUILD.gn
Delete rtc::TaskQueue
2024-02-28 10:22:49 +00:00
debug.proto
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DEPS
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echo_control_mobile_bit_exact_unittest.cc
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echo_control_mobile_impl.cc
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echo_control_mobile_impl.h
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echo_control_mobile_unittest.cc
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gain_control_impl.cc
Format /modules
2023-04-20 02:02:45 +00:00
gain_control_impl.h
AGC1: remove unused field trial WebRTC-UseLegacyDigitalGainApplier
2022-11-18 21:58:04 +00:00
gain_control_unittest.cc
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gain_controller2_unittest.cc
Update GainController2 adaptive digital default parameters
2024-04-12 08:29:26 +00:00
gain_controller2.cc
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +00:00
gain_controller2.h
AGC2: Return the recommended volume from RecommendInputVolume()
2022-12-14 13:05:37 +00:00
high_pass_filter_unittest.cc
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high_pass_filter.cc
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high_pass_filter.h
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optionally_built_submodule_creators.cc
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optionally_built_submodule_creators.h
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OWNERS
Update some audio modules with new OWNERS
2022-12-01 14:55:38 +00:00
render_queue_item_verifier.h
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residual_echo_detector_unittest.cc
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residual_echo_detector.cc
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residual_echo_detector.h
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rms_level_unittest.cc
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rms_level.cc
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rms_level.h
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splitting_filter_unittest.cc
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splitting_filter.cc
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splitting_filter.h
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three_band_filter_bank.cc
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three_band_filter_bank.h
…