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webrtc_m130/modules/audio_processing
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Jesús de Vicente Peña eeff850106 Adding the option to experiment with the max_allowed_excess_render_blocks parameter. 
Bug: webrtc:337900458
Change-Id: I2108c7c67eb9aa460932efe881760924109b1915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349460
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42207}
2024-05-02 12:20:23 +00:00
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aec3
Adding the option to experiment with the max_allowed_excess_render_blocks parameter.
2024-05-02 12:20:23 +00:00
aec_dump
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
aecm
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agc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
agc2
Start using ArrayView in AudioFrame, update PushResampler
2024-04-30 15:33:08 +00:00
capture_levels_adjuster
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echo_detector
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g3doc
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include
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
logging
Fix improper buffer size in call to rtc::strcpyn
2023-09-12 11:40:07 +00:00
ns
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test
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
transient
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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
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_frame_view_unittest.cc
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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
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audio_processing_impl_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_processing_impl.cc
Use num_output_channels() in GainController2
2024-02-12 11:29:20 +00:00
audio_processing_impl.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_processing_performance_unittest.cc
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audio_processing_unittest.cc
Start using ArrayView in AudioFrame, update PushResampler
2024-04-30 15:33:08 +00:00
BUILD.gn
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +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
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
echo_control_mobile_impl.h
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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
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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
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gain_controller2.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +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
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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
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
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
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