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webrtc_m130/modules/audio_processing
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Tommi 7e59d264f1 Remove unused istream code in test_utils. 
Bug: webrtc:8982
Change-Id: I52cf9778581190399de8e2068e4a1cd03c97fb3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356140
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42569}
2024-07-02 10:22:12 +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
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agc
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
agc2
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +00:00
capture_levels_adjuster
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echo_detector
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g3doc
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include
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +00:00
logging
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ns
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
test
Remove unused istream code in test_utils.
2024-07-02 10:22:12 +00:00
transient
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +00:00
utility
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vad
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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
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +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
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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
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audio_processing_unittest.cc
Use audio views in Interleave() and Deinterleave()
2024-05-30 13:07:32 +00:00
BUILD.gn
Switch away from hz to samples per channel for FrameCombiner et al
2024-06-13 19:00:39 +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
Switch away from hz to samples per channel for FrameCombiner et al
2024-06-13 19:00:39 +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
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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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