webrtc/modules/audio_processing
2024-06-21 16:31:45 -07:00
..
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 Format /modules 2023-04-20 02:02:45 +00:00
agc Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +00:00
agc2 Merge remote-tracking branch 'google/branch-heads/6478' 2024-06-21 16:31:45 -07:00
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 Merge remote-tracking branch 'google/branch-heads/6478' 2024-06-21 16:31:45 -07:00
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 Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +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.cc Update to 5005 (M102) (#86) 2022-08-24 11:07:33 -04:00
audio_buffer.h Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +00:00
audio_buffer_unittest.cc Rename more death test to *DeathTest 2020-05-26 20:27:34 +00:00
audio_frame_view_unittest.cc Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
audio_processing_builder_impl.cc Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +00:00
audio_processing_impl.cc Merge m123/6312 2024-06-12 22:25:35 -07:00
audio_processing_impl.h Merge remote-tracking branch 'google/branch-heads/6478' 2024-06-21 16:31:45 -07:00
audio_processing_impl_locking_unittest.cc Finish resolving merge conflicts 2022-11-11 19:10:59 -05:00
audio_processing_impl_unittest.cc Move webrtc::AudioProcessing include to api/ folder 2024-04-20 07:02:50 +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 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 AEC dump Stream::level renamed 2022-09-09 14:39:35 +00:00
DEPS
echo_control_mobile_bit_exact_unittest.cc Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
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 Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
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
gain_controller2_unittest.cc Update GainController2 adaptive digital default parameters 2024-04-12 08:29:26 +00:00
high_pass_filter.cc Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
high_pass_filter.h Reduce for reallocations the pre-amplifier and high-pass filter 2020-01-03 14:10:21 +00:00
high_pass_filter_unittest.cc Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
optionally_built_submodule_creators.cc Update to 5005 (M102) (#86) 2022-08-24 11:07:33 -04:00
optionally_built_submodule_creators.h Update to 5005 (M102) (#86) 2022-08-24 11:07:33 -04:00
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 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
splitting_filter_unittest.cc Reland "Simplification and refactoring of the AudioBuffer code" 2019-08-22 10:34:05 +00:00
three_band_filter_bank.cc Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
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