webrtc/modules/audio_processing
Per Kjellander a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
This reverts commit c73e1f4378.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00
..
aec Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
aec3 Use the filter delay to use the proper render block in the AEC3 AecState 2018-01-10 15:53:02 +00:00
aec_dump Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
aecm Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
agc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
agc2 Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
beamformer Fix macro clash with _USE_MATH_DEFINES. 2017-12-13 09:39:20 +00:00
echo_detector Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
include Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
intelligibility Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
level_controller Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
logging Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ns Forgotten 'memset' in NoiseSuppression. 2017-10-23 12:11:47 +00:00
test Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
transient Fix circular deps in common_audio. 2017-11-17 11:20:17 +00:00
utility Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
vad audio_processing VAD annotations in APM-qa. 2017-11-07 10:37:00 +00:00
audio_buffer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_processing_impl.cc Added a builder class for the AudioProcessingModule. 2017-12-22 12:19:03 +00:00
audio_processing_impl.h Render-side pre-processing in APM. 2017-12-18 16:11:03 +00:00
audio_processing_impl_locking_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
audio_processing_impl_unittest.cc Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
audio_processing_performance_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
audio_processing_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
BUILD.gn Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug.proto
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_bit_exact_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_impl.cc Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS 2017-12-22 15:42:13 +00:00
echo_cancellation_impl.h Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS 2017-12-22 15:42:13 +00:00
echo_cancellation_impl_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
echo_control_mobile_impl.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
echo_control_mobile_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_control_mobile_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_impl.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
gain_control_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS
render_queue_item_verifier.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
residual_echo_detector.cc Stop using LOG macros in favor of RTC_ prefixed macros. 2017-11-09 11:56:32 +00:00
residual_echo_detector.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
residual_echo_detector_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rms_level.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
rms_level.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
rms_level_unittest.cc Move some more numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 12:39:39 +00:00
splitting_filter.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
three_band_filter_bank.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
three_band_filter_bank.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
typing_detection.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
typing_detection.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
voice_detection_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
voice_detection_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
voice_detection_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00