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Fixing WebRTC after moving from src/webrtc to src/

Browse Source 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:47:31 +02:00
committed by Commit Bot
parent bb547203bf
commit 92ea95e34a
3635 changed files with 19692 additions and 19645 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_processing/level_controller/biquad_filter.cc
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
#include "modules/audio_processing/level_controller/biquad_filter.h"
namespace webrtc {

12
modules/audio_processing/level_controller/biquad_filter.h
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/arraysize.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/array_view.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -55,4 +55,4 @@ class BiQuadFilter {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_

10
modules/audio_processing/level_controller/down_sampler.cc
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
#include "modules/audio_processing/level_controller/down_sampler.h"
#include <string.h>
#include <algorithm>
#include <vector>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/rtc_base/checks.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/level_controller/biquad_filter.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {

12
modules/audio_processing/level_controller/down_sampler.h
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/level_controller/biquad_filter.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/array_view.h"
#include "modules/audio_processing/level_controller/biquad_filter.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -37,4 +37,4 @@ class DownSampler {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_

10
modules/audio_processing/level_controller/gain_applier.cc
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
#include "modules/audio_processing/level_controller/gain_applier.h"
#include <algorithm>
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/checks.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
namespace {

8
modules/audio_processing/level_controller/gain_applier.h
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#include "webrtc/rtc_base/constructormagic.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -39,4 +39,4 @@ class GainApplier {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_

8
modules/audio_processing/level_controller/gain_selector.cc
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
#include "modules/audio_processing/level_controller/gain_selector.h"
#include <math.h>
#include <algorithm>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
#include "webrtc/rtc_base/checks.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/level_controller/level_controller_constants.h"
#include "rtc_base/checks.h"
namespace webrtc {

10
modules/audio_processing/level_controller/gain_selector.h
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
#include "webrtc/rtc_base/constructormagic.h"
#include "rtc_base/constructormagic.h"
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
namespace webrtc {
@ -37,4 +37,4 @@ class GainSelector {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_

28
modules/audio_processing/level_controller/level_controller.cc
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@ -8,25 +8,25 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
#include "modules/audio_processing/level_controller/level_controller.h"
#include <math.h>
#include <algorithm>
#include <numeric>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/rtc_base/arraysize.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/level_controller/gain_applier.h"
#include "modules/audio_processing/level_controller/gain_selector.h"
#include "modules/audio_processing/level_controller/noise_level_estimator.h"
#include "modules/audio_processing/level_controller/peak_level_estimator.h"
#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {

24
modules/audio_processing/level_controller/level_controller.h
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@ -8,21 +8,21 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
#include <memory>
#include <vector>
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/gain_applier.h"
#include "webrtc/modules/audio_processing/level_controller/gain_selector.h"
#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/optional.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/level_controller/gain_applier.h"
#include "modules/audio_processing/level_controller/gain_selector.h"
#include "modules/audio_processing/level_controller/noise_level_estimator.h"
#include "modules/audio_processing/level_controller/peak_level_estimator.h"
#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -92,4 +92,4 @@ class LevelController {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_

24
modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
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@ -11,18 +11,18 @@
#include <numeric>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
#include "webrtc/modules/audio_processing/test/performance_timer.h"
#include "webrtc/modules/audio_processing/test/simulator_buffers.h"
#include "webrtc/rtc_base/random.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/level_controller/level_controller.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "modules/audio_processing/test/performance_timer.h"
#include "modules/audio_processing/test/simulator_buffers.h"
#include "rtc_base/random.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace {

6
modules/audio_processing/level_controller/level_controller_constants.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
namespace webrtc {
@ -20,4 +20,4 @@ const float kTargetLcPeakLeveldBFS = -6.0206f;
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_

16
modules/audio_processing/level_controller/level_controller_unittest.cc
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@ -10,14 +10,14 @@
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
#include "webrtc/test/gtest.h"
#include "api/array_view.h"
#include "api/optional.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/level_controller/level_controller.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "test/gtest.h"
namespace webrtc {
namespace {

6
modules/audio_processing/level_controller/noise_level_estimator.cc
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/noise_level_estimator.h"
#include "modules/audio_processing/level_controller/noise_level_estimator.h"
#include <algorithm>
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {

10
modules/audio_processing/level_controller/noise_level_estimator.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -34,4 +34,4 @@ class NoiseLevelEstimator {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_

8
modules/audio_processing/level_controller/noise_spectrum_estimator.cc
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include <string.h>
#include <algorithm>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/rtc_base/arraysize.h"
#include "api/array_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/arraysize.h"
namespace webrtc {
namespace {

10
modules/audio_processing/level_controller/noise_spectrum_estimator.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/array_view.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -37,4 +37,4 @@ class NoiseSpectrumEstimator {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_

6
modules/audio_processing/level_controller/peak_level_estimator.cc
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/peak_level_estimator.h"
#include "modules/audio_processing/level_controller/peak_level_estimator.h"
#include <algorithm>
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
namespace {

12
modules/audio_processing/level_controller/peak_level_estimator.h
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
#include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "modules/audio_processing/level_controller/level_controller_constants.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -34,4 +34,4 @@ class PeakLevelEstimator {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_

6
modules/audio_processing/level_controller/saturating_gain_estimator.cc
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
#include <math.h>
#include <algorithm>
#include "webrtc/modules/audio_processing/level_controller/level_controller_constants.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/level_controller/level_controller_constants.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {

8
modules/audio_processing/level_controller/saturating_gain_estimator.h
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
#include "webrtc/rtc_base/constructormagic.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -34,4 +34,4 @@ class SaturatingGainEstimator {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_

14
modules/audio_processing/level_controller/signal_classifier.cc
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@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
#include "modules/audio_processing/level_controller/signal_classifier.h"
#include <algorithm>
#include <numeric>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
#include "webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/level_controller/down_sampler.h"
#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace {

16
modules/audio_processing/level_controller/signal_classifier.h
View File

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
#include <memory>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/level_controller/down_sampler.h"
#include "webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include "webrtc/modules/audio_processing/utility/ooura_fft.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/array_view.h"
#include "modules/audio_processing/level_controller/down_sampler.h"
#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
#include "modules/audio_processing/utility/ooura_fft.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -64,4 +64,4 @@ class SignalClassifier {
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_