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webrtc/modules/audio_processing
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Mirko Bonadei 2ea9af2275 Revert "Export symbols needed by the Chromium component build (part 1)." 
This reverts commit 9e24dcff16.

Reason for revert: Breaks chromium.webrtc.fyi bots.

Original change's description:
> Export symbols needed by the Chromium component build (part 1).
> 
> This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
> to mark WebRTC symbols as visible from a shared library, this doesn't
> mean these symbols are part of the public API (please continue to refer
> to [1] for info about what is considered public WebRTC API).
> 
> [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md
> 
> Bug: webrtc:9419
> Change-Id: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}

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

Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
2018-10-04 09:49:53 +00:00
..
aec Remove deprecated field_trial_default and metrics_default. 2018-09-28 07:21:07 +00:00
aec3 AEC3: Reseting the ERLE at pre-amplifier gain changes 2018-10-02 15:53:58 +00:00
aec_dump Revert "Export symbols needed by the Chromium component build (part 1)." 2018-10-04 09:49:53 +00:00
aecm Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
agc Remove deprecated field_trial_default and metrics_default. 2018-09-28 07:21:07 +00:00
agc2 AEC3: Reseting the ERLE at pre-amplifier gain changes 2018-10-02 15:53:58 +00:00
audio_generator Add stub draft of audio generator to APM 2018-03-05 09:28:52 +00:00
echo_detector Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
include Revert "Export symbols needed by the Chromium component build (part 1)." 2018-10-04 09:49:53 +00:00
logging Remove stringstream usages from the APM 2018-04-06 14:17:03 +00:00
ns Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
test Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
transient Remove simple stringstream usages. 2018-09-06 12:53:19 +00:00
utility Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
vad Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
audio_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_buffer.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_frame_view_unittest.cc Add namespace 'webrtc' to AudioFrameView. 2018-05-14 12:33:49 +00:00
audio_processing_impl.cc Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
audio_processing_impl.h Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
audio_processing_impl_locking_unittest.cc Reland "Remove APM internal usage of EchoCancellation" 2018-09-17 09:51:08 +00:00
audio_processing_impl_unittest.cc APM: render pre-processor moved before echo detector queuing. 2018-08-09 14:40:31 +00:00
audio_processing_performance_unittest.cc Reland "Remove APM internal usage of EchoCancellation" 2018-09-17 09:51:08 +00:00
audio_processing_unittest.cc Reland "Remove APM-internal usage of EchoControlMobile" 2018-10-03 07:45:33 +00:00
BUILD.gn Revert "Export symbols needed by the Chromium component build (part 1)." 2018-10-04 09:49:53 +00:00
common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
debug.proto Reland "Remove APM-internal usage of EchoControlMobile" 2018-10-03 07:45:33 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_bit_exact_unittest.cc Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
echo_cancellation_impl.cc Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
echo_cancellation_impl.h Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
echo_cancellation_impl_unittest.cc Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
echo_control_mobile_bit_exact_unittest.cc Reland "Remove APM-internal usage of EchoControlMobile" 2018-10-03 07:45:33 +00:00
echo_control_mobile_impl.cc Reland "Remove APM-internal usage of EchoControlMobile" 2018-10-03 07:45:33 +00:00
echo_control_mobile_impl.h Delete deprecated AEC interfaces 2018-10-04 09:20:10 +00:00
echo_control_mobile_unittest.cc Reland "Remove APM-internal usage of EchoControlMobile" 2018-10-03 07:45:33 +00:00
gain_control_for_experimental_agc.cc Atomically increment GainControl instance counter. 2018-08-15 07:44:00 +00:00
gain_control_for_experimental_agc.h Reset Agc2 on analog gain changes. 2018-08-08 14:36:37 +00:00
gain_control_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
gain_control_impl.h Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
gain_control_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_controller2.cc Reduce digital adaptive AGC2 gain in some situations. 2018-10-02 08:34:10 +00:00
gain_controller2.h Optionally disable digital adaptive AGC2. 2018-08-24 15:54:43 +00:00
gain_controller2_unittest.cc Set a positive initial gain in the Adaptive Digital GC. 2018-04-27 09:05:25 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Adding alessiob@ and minyue@ as owners of APM. 2018-07-02 07:45:31 +00:00
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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
residual_echo_detector.h Add more parameters to the Initialize function of the echo detector. 2018-03-15 09:21:56 +00:00
residual_echo_detector_unittest.cc Change echo detector to scoped_refptr 2018-06-14 09:51:41 +00:00
rms_level.cc Replace rtc::Optional with absl::optional in modules/audio processing 2018-06-19 10:38:56 +00:00
rms_level.h Delete root header file typedef.h. 2018-07-25 14:59:26 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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 Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
typing_detection.h Revert "Export symbols needed by the Chromium component build (part 1)." 2018-10-04 09:49:53 +00:00
voice_detection_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +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