| .. |
|
aec3
|
Penalization of large delays on the initial phase.
|
2023-02-22 07:11:58 +00:00 |
|
aec_dump
|
Clean up diff
|
2022-11-18 15:39:11 -05:00 |
|
aecm
|
Finish resolving merge conflicts
|
2022-11-11 19:10:59 -05:00 |
|
agc
|
Add generic input volume controller test for both AGC1 and AGC2
|
2022-12-20 14:41:31 +00:00 |
|
agc2
|
Retune AGC2 input volume controller speech ratio threshold
|
2023-01-20 14:03:58 +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
|
Merge branch 'm110' into 5481
|
2023-02-17 11:35:29 -05:00 |
|
logging
|
Adopt absl::string_view in modules/audio_processing/
|
2022-08-16 13:49:14 +00:00 |
|
ns
|
Update to 5005 (M102) (#86)
|
2022-08-24 11:07:33 -04:00 |
|
test
|
Finish resolving merge conflicts
|
2022-11-11 19:10:59 -05:00 |
|
transient
|
Finish resolving merge conflicts
|
2022-11-11 19:10:59 -05:00 |
|
utility
|
Fix math involving enums in C++20
|
2022-09-27 06:55:31 +00:00 |
|
vad
|
Make header files self contained.
|
2022-10-08 08:38:36 +00:00 |
|
audio_buffer.cc
|
Update to 5005 (M102) (#86)
|
2022-08-24 11:07:33 -04:00 |
|
audio_buffer.h
|
APM: Signal error on unsupported sample rates
|
2022-11-17 12:12:04 +00:00 |
|
audio_buffer_unittest.cc
|
|
|
|
audio_frame_view_unittest.cc
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
audio_processing_builder_impl.cc
|
Move rtc::make_ref_counted to api/
|
2022-06-15 09:47:38 +00:00 |
|
audio_processing_impl.cc
|
Merge branch 'm112' into 5615
|
2023-04-27 12:45:13 -04:00 |
|
audio_processing_impl.h
|
Merge branch 'm112' into 5615
|
2023-04-27 12:45:13 -04:00 |
|
audio_processing_impl_locking_unittest.cc
|
Finish resolving merge conflicts
|
2022-11-11 19:10:59 -05:00 |
|
audio_processing_impl_unittest.cc
|
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
|
2023-01-16 20:30:12 +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
|
AGC2 adaptive digital controller config clean-up
|
2022-12-09 13:07:34 +00:00 |
|
BUILD.gn
|
Remove all usage of //rtc_base target
|
2023-01-16 14:36:06 +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
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
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
|
|
|
|
gain_control_impl.cc
|
AGC1: remove unused field trial WebRTC-UseLegacyDigitalGainApplier
|
2022-11-18 21:58:04 +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
|
AGC2: Return the recommended volume from RecommendInputVolume()
|
2022-12-14 13:05:37 +00:00 |
|
gain_controller2_unittest.cc
|
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
|
2023-01-16 20:30:12 +00:00 |
|
high_pass_filter.cc
|
Update to 4896 (M100) (#72)
|
2022-04-15 17:13:23 -06:00 |
|
high_pass_filter.h
|
|
|
|
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
|
Delete rtc_base/atomic_ops.h
|
2022-06-28 08:32:13 +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
|
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 |
