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webrtc/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 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 Updates to AudioFrameView and VectorFloatFrame 2024-06-17 12:13:40 +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
g3doc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
include Updates to AudioFrameView and VectorFloatFrame 2024-06-17 12:13:40 +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 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 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.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_buffer_unittest.cc
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.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_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_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 Switch away from hz to samples per channel for FrameCombiner et al 2024-06-13 19:00:39 +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 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.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
gain_controller2_unittest.cc Update GainController2 adaptive digital default parameters 2024-04-12 08:29:26 +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
high_pass_filter_unittest.cc Remove more traces of keyboard mic support from APM 2022-02-04 14:27:51 +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
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
residual_echo_detector_unittest.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +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
rms_level_unittest.cc Ensure that an RTP audio level of 127 represents digital silence. 2022-05-06 07:56:39 +00:00
splitting_filter.cc
splitting_filter.h
splitting_filter_unittest.cc
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