Tommi f58ded7cf0 Use audio views in Interleave() and Deinterleave() ... Interleave and Deinterleave now accept two parameters, one for the interleaved buffer and another for the deinterleaved one. The previous versions of the functions still need to exist for test code that uses ChannelBuffer. Bug: chromium:335805780 Change-Id: I20371ab6408766d21e6901e6a04000afa05b3553 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351664 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42412}		2024-05-30 13:07:32 +00:00
..
aec3	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +00:00
aec_dump	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +00:00
aecm	Format /modules	2023-04-20 02:02:45 +00:00
agc	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +00:00
agc2	Add audio view classes	2024-05-24 18:08:37 +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	Move webrtc::AudioProcessing include to api/ folder	2024-04-20 07:02:50 +00:00
logging	Fix improper buffer size in call to rtc::strcpyn	2023-09-12 11:40:07 +00:00
ns	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +00:00
test	Use audio views in Interleave() and Deinterleave()	2024-05-30 13:07:32 +00:00
transient	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +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	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	Remove more traces of keyboard mic support from APM	2022-02-04 14:27:51 +00:00
audio_processing_builder_impl.cc	Move webrtc::AudioProcessing include to api/ folder	2024-04-20 07:02:50 +00:00
audio_processing_impl.cc	Use num_output_channels() in GainController2	2024-02-12 11:29:20 +00:00
audio_processing_impl.h	Move webrtc::AudioProcessing include to api/ folder	2024-04-20 07:02:50 +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	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	Use audio views in Interleave() and Deinterleave()	2024-05-30 13:07:32 +00:00
BUILD.gn	Mass removal of absl_deps in all BUILD.gn files	2024-05-23 15:09:46 +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	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	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	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	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	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	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