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webrtc_m130/modules/audio_processing
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Hanna Silen 7ee37cf839 Deprecate WebRTC-Audio-GainController2 fieldtrial 
Bug: webrtc:7494
Change-Id: I315a6e5d203a7f7f86e27d5b1b1f7dd72ccf1b08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354100
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42468}
2024-06-12 12:37:49 +00:00
..
aec3
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
aec_dump
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
aecm
Format /modules
2023-04-20 02:02:45 +00:00
agc
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
agc2
Update FrameCombiner to use audio view methods for interleaved buffers
2024-06-12 09:44:40 +00:00
capture_levels_adjuster
Add refined handling of the internal scaling of the audio in APM
2021-03-15 19:12:02 +00:00
echo_detector
Format almost everything.
2019-07-08 13:45:15 +00:00
g3doc
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
include
Use MonoView for deinterleaved channels in AudioFrameView
2024-06-05 12:39:27 +00:00
logging
Fix improper buffer size in call to rtc::strcpyn
2023-09-12 11:40:07 +00:00
ns
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
test
Add AGC2 input volume controller mode in audioproc_f
2024-06-11 08:44:10 +00:00
transient
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
utility
Fix math involving enums in C++20
2022-09-27 06:55:31 +00:00
vad
Fix downstream review comments for C++20
2023-07-04 09:06:07 +00:00
audio_buffer_unittest.cc
Rename more death test to *DeathTest
2020-05-26 20:27:34 +00:00
audio_buffer.cc
AudioBuffer: Remove deprecated constructor
2022-04-11 10:06:07 +00:00
audio_buffer.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_frame_view_unittest.cc
Remove more traces of keyboard mic support from APM
2022-02-04 14:27:51 +00:00
audio_processing_builder_impl.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_processing_impl_locking_unittest.cc
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +00:00
audio_processing_impl_unittest.cc
Deprecate WebRTC-Audio-GainController2 fieldtrial
2024-06-12 12:37:49 +00:00
audio_processing_impl.cc
Deprecate WebRTC-Audio-GainController2 fieldtrial
2024-06-12 12:37:49 +00:00
audio_processing_impl.h
Deprecate WebRTC-Audio-GainController2 fieldtrial
2024-06-12 12:37:49 +00:00
audio_processing_performance_unittest.cc
Migrate CallSimulator to the new perf metrics logging API
2022-09-26 19:37:51 +00:00
audio_processing_unittest.cc
Use audio views in Interleave() and Deinterleave()
2024-05-30 13:07:32 +00:00
BUILD.gn
Use MonoView for deinterleaved channels in AudioFrameView
2024-06-05 12:39:27 +00:00
debug.proto
AEC dump Stream::level renamed
2022-09-09 14:39:35 +00:00
DEPS
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
echo_control_mobile_bit_exact_unittest.cc
Remove more traces of keyboard mic support from APM
2022-02-04 14:27:51 +00:00
echo_control_mobile_impl.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
echo_control_mobile_impl.h
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
2021-08-09 21:49:02 +00:00
echo_control_mobile_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
gain_control_impl.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +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
Remove more traces of keyboard mic support from APM
2022-02-04 14:27:51 +00:00
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
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
high_pass_filter_unittest.cc
Remove more traces of keyboard mic support from APM
2022-02-04 14:27:51 +00:00
high_pass_filter.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
high_pass_filter.h
Reduce for reallocations the pre-amplifier and high-pass filter
2020-01-03 14:10:21 +00:00
optionally_built_submodule_creators.cc
APM Transient Suppressor (TS): initialization params in ctor
2022-04-08 09:41:44 +00:00
optionally_built_submodule_creators.h
APM Transient Suppressor (TS): initialization params in ctor
2022-04-08 09:41:44 +00:00
OWNERS
Update some audio modules with new OWNERS
2022-12-01 14:55:38 +00:00
render_queue_item_verifier.h
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
residual_echo_detector_unittest.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
residual_echo_detector.cc
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +00:00
residual_echo_detector.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
rms_level_unittest.cc
Ensure that an RTP audio level of 127 represents digital silence.
2022-05-06 07:56:39 +00:00
rms_level.cc
Ensure that an RTP audio level of 127 represents digital silence.
2022-05-06 07:56:39 +00:00
rms_level.h
Ensure that an RTP audio level of 127 represents digital silence.
2022-05-06 07:56:39 +00:00
splitting_filter_unittest.cc
Reland "Simplification and refactoring of the AudioBuffer code"
2019-08-22 10:34:05 +00:00
splitting_filter.cc
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
splitting_filter.h
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
three_band_filter_bank.cc
Optimize the three band filter bank.
2021-12-16 13:37:30 +00:00
three_band_filter_bank.h
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
2021-08-09 21:49:02 +00:00