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webrtc_m130/modules/audio_processing
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Emil Lundmark f591f2dcf1 Remove expired WebRTC-Aec3DelayEstimatorDetectPreEcho 
Bug: webrtc:14205
Change-Id: Ib817b043d9368ba003b2b40a7315da845910c2f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345481
Reviewed-by: Lionel Koenig Gélas <lionelk@google.com>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42039}
2024-04-11 09:57:16 +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
AGC2: Enable clipping predictor by default
2024-02-09 14:08:27 +00:00
capture_levels_adjuster
…
echo_detector
…
g3doc
Reland "Migrate WebRTC documentation to new renderer"
2023-01-31 09:30:04 +00:00
include
Remove dependency on rtc::TaskQueue in AudioProcessing module
2024-01-17 18:12:16 +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
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
…
audio_buffer.cc
AudioBuffer: Remove deprecated constructor
2022-04-11 10:06:07 +00:00
audio_buffer.h
APM: Signal error on unsupported sample rates
2022-11-17 12:12:04 +00:00
audio_frame_view_unittest.cc
…
audio_processing_builder_impl.cc
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +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
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
Migrate CallSimulator to the new perf metrics logging API
2022-09-26 19:37:51 +00:00
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
AEC dump Stream::level renamed
2022-09-09 14:39:35 +00:00
DEPS
…
echo_control_mobile_bit_exact_unittest.cc
…
echo_control_mobile_impl.cc
…
echo_control_mobile_impl.h
…
echo_control_mobile_unittest.cc
…
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
…
gain_controller2_unittest.cc
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +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
…
high_pass_filter.cc
…
high_pass_filter.h
…
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
…
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
Delete rtc_base/atomic_ops.h
2022-06-28 08:32:13 +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
…
splitting_filter.cc
…
splitting_filter.h
…
three_band_filter_bank.cc
…
three_band_filter_bank.h
…