admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/audio_processing
History
Artem Titov 0f2ce5cc1c Revert "Migrate WebRTC documentation to new renderer" 
This reverts commit 3eceaf46695518f25bef43f155f82ed174827197.

Reason for revert: 

Original change's description:
> Migrate WebRTC documentation to new renderer
>
> Bug: b/258408932
> Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39205}

Bug: b/258408932
Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39209}
2023-01-26 20:19:12 +00:00
..
aec3
Version 3: Various changes on the pre-echo delay estimator:
2023-01-23 15:10:48 +00:00
aec_dump
…
aecm
…
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
…
echo_detector
…
g3doc
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
include
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
logging
…
ns
…
test
APM: remove kClippedLevelMin from audio_processing.h
2022-10-13 15:01:37 +00:00
transient
…
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_unittest.cc
…
audio_buffer.cc
…
audio_buffer.h
APM: Signal error on unsupported sample rates
2022-11-17 12:12:04 +00:00
audio_frame_view_unittest.cc
…
audio_processing_builder_impl.cc
…
audio_processing_impl_locking_unittest.cc
…
audio_processing_impl_unittest.cc
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +00:00
audio_processing_impl.cc
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +00:00
audio_processing_impl.h
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +00:00
audio_processing_performance_unittest.cc
…
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
…
DEPS
…
echo_control_mobile_bit_exact_unittest.cc
…
echo_control_mobile_impl.cc
…
echo_control_mobile_impl.h
…
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
…
gain_controller2_unittest.cc
APM: fix TS initialization bugs with WebRTC-Audio-GainController2
2023-01-16 20:30:12 +00:00
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
high_pass_filter_unittest.cc
…
high_pass_filter.cc
…
high_pass_filter.h
…
optionally_built_submodule_creators.cc
…
optionally_built_submodule_creators.h
…
OWNERS
Update some audio modules with new OWNERS
2022-12-01 14:55:38 +00:00
render_queue_item_verifier.h
…
residual_echo_detector_unittest.cc
…
residual_echo_detector.cc
…
residual_echo_detector.h
…
rms_level_unittest.cc
…
rms_level.cc
…
rms_level.h
…
splitting_filter_unittest.cc
…
splitting_filter.cc
…
splitting_filter.h
…
three_band_filter_bank.cc
…
three_band_filter_bank.h
…