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webrtc/modules/audio_processing
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Li-Yu Yu 758f26852d Fix downstream review comments for C++20 
This CL addresses the review comments for
https://webrtc-review.googlesource.com/c/src/+/261221
in the downstream cherry-pick: https://crrev.com/c/4660950.

*   Always use size_t{} for casting.
*   Remove unneeded size_t casts.
*   Avoid using __x as it is reserved for the compiler.

Bug: b:217226507
Change-Id: I13c57cb69d7db066ac9a6dbd15b7f6de54abb613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311360
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Li-Yu Yu <aaronyu@google.com>
Cr-Commit-Position: refs/heads/main@{#40395}
2023-07-04 09:06:07 +00:00
..
aec3 Changing the pre echo configuration default. 2023-06-20 14:46:44 +00:00
aec_dump APM: add AudioProcessingImpl::capture_::applied_input_volume(_changed) 2022-09-09 17:36:05 +00:00
aecm Format /modules 2023-04-20 02:02:45 +00:00
agc Format /modules 2023-04-20 02:02:45 +00:00
agc2 Use sinf instead of std::sinf to improve libstdc++ compatibility 2023-05-30 20:16:31 +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 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 AGC2 adaptive digital controller config clean-up 2022-12-09 13:07:34 +00:00
logging Adopt absl::string_view in modules/audio_processing/ 2022-08-16 13:49:14 +00:00
ns Format /modules 2023-04-20 02:02:45 +00:00
test Format /modules 2023-04-20 02:02:45 +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 AudioBuffer: Remove deprecated constructor 2022-04-11 10:06:07 +00:00
audio_buffer.h APM: Signal error on unsupported sample rates 2022-11-17 12:12:04 +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 Remove more traces of keyboard mic support from APM 2022-02-04 14:27:51 +00:00
audio_processing_builder_impl.cc Move rtc::make_ref_counted to api/ 2022-06-15 09:47:38 +00:00
audio_processing_impl.cc 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_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 Remove mentions of already deleted field trials 2023-03-01 15:53:37 +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 Remove more traces of keyboard mic support from APM 2022-02-04 14:27:51 +00:00
echo_control_mobile_impl.cc Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/ 2022-01-24 11:50:20 +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 Remove unused critical section includes. 2020-07-16 13:52:28 +00:00
gain_control_impl.cc Format /modules 2023-04-20 02:02:45 +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 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 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED 2021-11-15 21:44:59 +00:00
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 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 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 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 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