Format almost everything.
This CL was generated by running
git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format
Most of these changes are clang-format grouping and reordering includes
differently.
Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
diff --git a/modules/audio_processing/aec/aec_core.cc b/modules/audio_processing/aec/aec_core.cc
index f0deddc..1e0f63f 100644
--- a/modules/audio_processing/aec/aec_core.cc
+++ b/modules/audio_processing/aec/aec_core.cc
@@ -18,6 +18,7 @@
#include <stddef.h> // size_t
#include <stdlib.h>
#include <string.h>
+
#include <algorithm>
#include <cmath>
diff --git a/modules/audio_processing/aec/aec_core_mips.cc b/modules/audio_processing/aec/aec_core_mips.cc
index bf89cfa..2b388a7 100644
--- a/modules/audio_processing/aec/aec_core_mips.cc
+++ b/modules/audio_processing/aec/aec_core_mips.cc
@@ -12,10 +12,10 @@
* The core AEC algorithm, which is presented with time-aligned signals.
*/
-#include "modules/audio_processing/aec/aec_core.h"
-
#include <math.h>
+#include "modules/audio_processing/aec/aec_core.h"
+
extern "C" {
#include "common_audio/signal_processing/include/signal_processing_library.h"
}
diff --git a/modules/audio_processing/aec/echo_cancellation.h b/modules/audio_processing/aec/echo_cancellation.h
index 2039347..62dc0f0 100644
--- a/modules/audio_processing/aec/echo_cancellation.h
+++ b/modules/audio_processing/aec/echo_cancellation.h
@@ -11,10 +11,10 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
#define MODULES_AUDIO_PROCESSING_AEC_ECHO_CANCELLATION_H_
-#include <memory>
-
#include <stddef.h>
+#include <memory>
+
extern "C" {
#include "common_audio/ring_buffer.h"
}
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter.h b/modules/audio_processing/aec3/adaptive_fir_filter.h
index 5afb80e..a7418b0 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter.h
+++ b/modules/audio_processing/aec3/adaptive_fir_filter.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_ADAPTIVE_FIR_FILTER_H_
#include <stddef.h>
+
#include <array>
#include <vector>
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index 3c4f5a5..4e13bd6 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -11,12 +11,13 @@
#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
// Defines WEBRTC_ARCH_X86_FAMILY, used below.
-#include "rtc_base/system/arch.h"
-
#include <math.h>
+
#include <algorithm>
#include <numeric>
#include <string>
+
+#include "rtc_base/system/arch.h"
#if defined(WEBRTC_ARCH_X86_FAMILY)
#include <emmintrin.h>
#endif
diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index 179d98fe..e4ec9f8 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/aec_state.h"
#include <math.h>
+
#include <algorithm>
#include <numeric>
#include <vector>
diff --git a/modules/audio_processing/aec3/aec_state.h b/modules/audio_processing/aec3/aec_state.h
index 51a8ec0..713fa7e 100644
--- a/modules/audio_processing/aec3/aec_state.h
+++ b/modules/audio_processing/aec3/aec_state.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
#include <stddef.h>
+
#include <array>
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc b/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
index 86608aa..b902487 100644
--- a/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/api_call_jitter_metrics_unittest.cc
@@ -9,8 +9,8 @@
*/
#include "modules/audio_processing/aec3/api_call_jitter_metrics.h"
-#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/block_delay_buffer.h b/modules/audio_processing/aec3/block_delay_buffer.h
index 624e913..dd57759 100644
--- a/modules/audio_processing/aec3/block_delay_buffer.h
+++ b/modules/audio_processing/aec3/block_delay_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "modules/audio_processing/audio_buffer.h"
diff --git a/modules/audio_processing/aec3/block_processor.cc b/modules/audio_processing/aec3/block_processor.cc
index 0997b1a..0c31a2e 100644
--- a/modules/audio_processing/aec3/block_processor.cc
+++ b/modules/audio_processing/aec3/block_processor.cc
@@ -7,7 +7,10 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/aec3/block_processor.h"
+
#include <stddef.h>
+
#include <memory>
#include <utility>
#include <vector>
@@ -16,7 +19,6 @@
#include "api/audio/echo_canceller3_config.h"
#include "api/audio/echo_control.h"
#include "modules/audio_processing/aec3/aec3_common.h"
-#include "modules/audio_processing/aec3/block_processor.h"
#include "modules/audio_processing/aec3/block_processor_metrics.h"
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/echo_path_variability.h"
diff --git a/modules/audio_processing/aec3/block_processor.h b/modules/audio_processing/aec3/block_processor.h
index bcee3b7..8b1bb90 100644
--- a/modules/audio_processing/aec3/block_processor.h
+++ b/modules/audio_processing/aec3/block_processor.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_H_
#include <stddef.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/aec3/block_processor_metrics_unittest.cc b/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
index 73f7689..3e23c24 100644
--- a/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
@@ -9,8 +9,8 @@
*/
#include "modules/audio_processing/aec3/block_processor_metrics.h"
-#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/cascaded_biquad_filter.h b/modules/audio_processing/aec3/cascaded_biquad_filter.h
index 3d9b14b..34085f1 100644
--- a/modules/audio_processing/aec3/cascaded_biquad_filter.h
+++ b/modules/audio_processing/aec3/cascaded_biquad_filter.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_CASCADED_BIQUAD_FILTER_H_
#include <stddef.h>
+
#include <complex>
#include <vector>
diff --git a/modules/audio_processing/aec3/comfort_noise_generator.h b/modules/audio_processing/aec3/comfort_noise_generator.h
index f78fda2..79bf623 100644
--- a/modules/audio_processing/aec3/comfort_noise_generator.h
+++ b/modules/audio_processing/aec3/comfort_noise_generator.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_COMFORT_NOISE_GENERATOR_H_
#include <stdint.h>
+
#include <array>
#include <memory>
diff --git a/modules/audio_processing/aec3/decimator_unittest.cc b/modules/audio_processing/aec3/decimator_unittest.cc
index 79e7440..cf8de84 100644
--- a/modules/audio_processing/aec3/decimator_unittest.cc
+++ b/modules/audio_processing/aec3/decimator_unittest.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/decimator.h"
#include <math.h>
+
#include <algorithm>
#include <array>
#include <cmath>
diff --git a/modules/audio_processing/aec3/downsampled_render_buffer.h b/modules/audio_processing/aec3/downsampled_render_buffer.h
index c91ea3b..fbdc9b4 100644
--- a/modules/audio_processing/aec3/downsampled_render_buffer.h
+++ b/modules/audio_processing/aec3/downsampled_render_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aec3/echo_canceller3.h b/modules/audio_processing/aec3/echo_canceller3.h
index c1298d2..2782687 100644
--- a/modules/audio_processing/aec3/echo_canceller3.h
+++ b/modules/audio_processing/aec3/echo_canceller3.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
#include <stddef.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/aec3/echo_path_variability_unittest.cc b/modules/audio_processing/aec3/echo_path_variability_unittest.cc
index b1795ed..0f10f95 100644
--- a/modules/audio_processing/aec3/echo_path_variability_unittest.cc
+++ b/modules/audio_processing/aec3/echo_path_variability_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/aec3/echo_path_variability.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 493916c..f93288c 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -11,6 +11,7 @@
#include <math.h>
#include <stddef.h>
+
#include <algorithm>
#include <array>
#include <cmath>
diff --git a/modules/audio_processing/aec3/echo_remover_metrics.cc b/modules/audio_processing/aec3/echo_remover_metrics.cc
index 71d149e..4590f85 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics.cc
@@ -12,6 +12,7 @@
#include <math.h>
#include <stddef.h>
+
#include <algorithm>
#include <cmath>
#include <numeric>
diff --git a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
index 00ce1ea..c16c7ea 100644
--- a/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/echo_remover_metrics.h"
#include <math.h>
+
#include <cmath>
#include "modules/audio_processing/aec3/aec3_fft.h"
diff --git a/modules/audio_processing/aec3/erl_estimator.h b/modules/audio_processing/aec3/erl_estimator.h
index 060fb91..2ca21df 100644
--- a/modules/audio_processing/aec3/erl_estimator.h
+++ b/modules/audio_processing/aec3/erl_estimator.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_ERL_ESTIMATOR_H_
#include <stddef.h>
+
#include <array>
#include "api/array_view.h"
diff --git a/modules/audio_processing/aec3/erle_estimator.h b/modules/audio_processing/aec3/erle_estimator.h
index 8036c21..126774d 100644
--- a/modules/audio_processing/aec3/erle_estimator.h
+++ b/modules/audio_processing/aec3/erle_estimator.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_ERLE_ESTIMATOR_H_
#include <stddef.h>
+
#include <array>
#include <memory>
diff --git a/modules/audio_processing/aec3/erle_estimator_unittest.cc b/modules/audio_processing/aec3/erle_estimator_unittest.cc
index 5ef4f24..ac681b3 100644
--- a/modules/audio_processing/aec3/erle_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/erle_estimator_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/aec3/erle_estimator.h"
+
#include <cmath>
#include "api/array_view.h"
-#include "modules/audio_processing/aec3/erle_estimator.h"
#include "modules/audio_processing/aec3/render_delay_buffer.h"
#include "modules/audio_processing/aec3/vector_buffer.h"
#include "rtc_base/random.h"
diff --git a/modules/audio_processing/aec3/fft_buffer.h b/modules/audio_processing/aec3/fft_buffer.h
index 9f81a91..a367f9e 100644
--- a/modules/audio_processing/aec3/fft_buffer.h
+++ b/modules/audio_processing/aec3/fft_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "modules/audio_processing/aec3/fft_data.h"
diff --git a/modules/audio_processing/aec3/filter_analyzer.cc b/modules/audio_processing/aec3/filter_analyzer.cc
index 6bbeb6e..06bd4b7 100644
--- a/modules/audio_processing/aec3/filter_analyzer.cc
+++ b/modules/audio_processing/aec3/filter_analyzer.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/aec3/filter_analyzer.h"
+
#include <math.h>
#include <algorithm>
diff --git a/modules/audio_processing/aec3/filter_analyzer.h b/modules/audio_processing/aec3/filter_analyzer.h
index 0e1798c..bcce528 100644
--- a/modules/audio_processing/aec3/filter_analyzer.h
+++ b/modules/audio_processing/aec3/filter_analyzer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
#include <stddef.h>
+
#include <array>
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/aec3/frame_blocker.h b/modules/audio_processing/aec3/frame_blocker.h
index 68cee97..759f431 100644
--- a/modules/audio_processing/aec3/frame_blocker.h
+++ b/modules/audio_processing/aec3/frame_blocker.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_FRAME_BLOCKER_H_
#include <stddef.h>
+
#include <vector>
#include "api/array_view.h"
diff --git a/modules/audio_processing/aec3/main_filter_update_gain.h b/modules/audio_processing/aec3/main_filter_update_gain.h
index 5c817cd..dca0ff87 100644
--- a/modules/audio_processing/aec3/main_filter_update_gain.h
+++ b/modules/audio_processing/aec3/main_filter_update_gain.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_MAIN_FILTER_UPDATE_GAIN_H_
#include <stddef.h>
+
#include <array>
#include <memory>
diff --git a/modules/audio_processing/aec3/matched_filter.h b/modules/audio_processing/aec3/matched_filter.h
index 084267f..df92453 100644
--- a/modules/audio_processing/aec3/matched_filter.h
+++ b/modules/audio_processing/aec3/matched_filter.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_MATCHED_FILTER_H_
#include <stddef.h>
+
#include <vector>
#include "api/array_view.h"
@@ -66,7 +67,6 @@
} // namespace aec3
-
// Produces recursively updated cross-correlation estimates for several signal
// shifts where the intra-shift spacing is uniform.
class MatchedFilter {
diff --git a/modules/audio_processing/aec3/matrix_buffer.h b/modules/audio_processing/aec3/matrix_buffer.h
index cae3759..8fb96d21 100644
--- a/modules/audio_processing/aec3/matrix_buffer.h
+++ b/modules/audio_processing/aec3/matrix_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_MATRIX_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aec3/moving_average.h b/modules/audio_processing/aec3/moving_average.h
index 0f855be..913d785 100644
--- a/modules/audio_processing/aec3/moving_average.h
+++ b/modules/audio_processing/aec3/moving_average.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
#include <stddef.h>
+
#include <vector>
#include "api/array_view.h"
diff --git a/modules/audio_processing/aec3/moving_average_unittest.cc b/modules/audio_processing/aec3/moving_average_unittest.cc
index 05542d1..84ba9cb 100644
--- a/modules/audio_processing/aec3/moving_average_unittest.cc
+++ b/modules/audio_processing/aec3/moving_average_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/aec3/moving_average.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/render_buffer.h b/modules/audio_processing/aec3/render_buffer.h
index cc6cd1c..762eab8 100644
--- a/modules/audio_processing/aec3/render_buffer.h
+++ b/modules/audio_processing/aec3/render_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_BUFFER_H_
#include <stddef.h>
+
#include <array>
#include <vector>
diff --git a/modules/audio_processing/aec3/render_delay_buffer.cc b/modules/audio_processing/aec3/render_delay_buffer.cc
index 0b2e979..1a48f15 100644
--- a/modules/audio_processing/aec3/render_delay_buffer.cc
+++ b/modules/audio_processing/aec3/render_delay_buffer.cc
@@ -8,7 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/aec3/render_delay_buffer.h"
+
#include <string.h>
+
#include <algorithm>
#include <memory>
#include <numeric>
@@ -25,7 +28,6 @@
#include "modules/audio_processing/aec3/fft_data.h"
#include "modules/audio_processing/aec3/matrix_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
-#include "modules/audio_processing/aec3/render_delay_buffer.h"
#include "modules/audio_processing/aec3/vector_buffer.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
diff --git a/modules/audio_processing/aec3/render_delay_buffer.h b/modules/audio_processing/aec3/render_delay_buffer.h
index 89b3a2a..970cf91 100644
--- a/modules/audio_processing/aec3/render_delay_buffer.h
+++ b/modules/audio_processing/aec3/render_delay_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "api/audio/echo_canceller3_config.h"
diff --git a/modules/audio_processing/aec3/render_delay_controller.cc b/modules/audio_processing/aec3/render_delay_controller.cc
index e8423cb..ceafa21 100644
--- a/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/modules/audio_processing/aec3/render_delay_controller.cc
@@ -7,7 +7,10 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/aec3/render_delay_controller.h"
+
#include <stddef.h>
+
#include <algorithm>
#include <memory>
@@ -18,7 +21,6 @@
#include "modules/audio_processing/aec3/delay_estimate.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
-#include "modules/audio_processing/aec3/render_delay_controller.h"
#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomic_ops.h"
diff --git a/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
index 216b0e2..e7d7703 100644
--- a/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
+++ b/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
@@ -9,9 +9,9 @@
*/
#include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
+
#include "absl/types/optional.h"
#include "modules/audio_processing/aec3/aec3_common.h"
-
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/render_signal_analyzer.cc b/modules/audio_processing/aec3/render_signal_analyzer.cc
index 33b04bf..e3e41a7 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/render_signal_analyzer.h"
#include <math.h>
+
#include <algorithm>
#include <utility>
#include <vector>
diff --git a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
index ffd7fe2..1adfbfb 100644
--- a/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
+++ b/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/render_signal_analyzer.h"
#include <math.h>
+
#include <array>
#include <cmath>
#include <vector>
diff --git a/modules/audio_processing/aec3/residual_echo_estimator.cc b/modules/audio_processing/aec3/residual_echo_estimator.cc
index eaeaf49..a6fd2ff 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/residual_echo_estimator.h"
#include <stddef.h>
+
#include <algorithm>
#include <vector>
diff --git a/modules/audio_processing/aec3/reverb_decay_estimator.cc b/modules/audio_processing/aec3/reverb_decay_estimator.cc
index cdcbee5..2415931 100644
--- a/modules/audio_processing/aec3/reverb_decay_estimator.cc
+++ b/modules/audio_processing/aec3/reverb_decay_estimator.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/reverb_decay_estimator.h"
#include <stddef.h>
+
#include <algorithm>
#include <cmath>
#include <numeric>
diff --git a/modules/audio_processing/aec3/reverb_frequency_response.cc b/modules/audio_processing/aec3/reverb_frequency_response.cc
index 98eeca6..f4bd91f 100644
--- a/modules/audio_processing/aec3/reverb_frequency_response.cc
+++ b/modules/audio_processing/aec3/reverb_frequency_response.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/reverb_frequency_response.h"
#include <stddef.h>
+
#include <algorithm>
#include <array>
#include <numeric>
@@ -59,7 +60,6 @@
int filter_delay_blocks,
const absl::optional<float>& linear_filter_quality,
bool stationary_block) {
-
if (stationary_block || !linear_filter_quality) {
return;
}
diff --git a/modules/audio_processing/aec3/reverb_model.cc b/modules/audio_processing/aec3/reverb_model.cc
index f0a24c0..ca65960 100644
--- a/modules/audio_processing/aec3/reverb_model.cc
+++ b/modules/audio_processing/aec3/reverb_model.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/reverb_model.h"
#include <stddef.h>
+
#include <algorithm>
#include <functional>
diff --git a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
index 9947ed7..8fce9d2 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
@@ -21,7 +21,6 @@
#include "modules/audio_processing/aec3/aec3_fft.h"
#include "modules/audio_processing/aec3/fft_data.h"
#include "rtc_base/checks.h"
-
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/aec3/reverb_model_fallback.h b/modules/audio_processing/aec3/reverb_model_fallback.h
index 1bd2b59..83ad233 100644
--- a/modules/audio_processing/aec3/reverb_model_fallback.h
+++ b/modules/audio_processing/aec3/reverb_model_fallback.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
#include <stddef.h>
+
#include <array>
#include <vector>
diff --git a/modules/audio_processing/aec3/shadow_filter_update_gain.h b/modules/audio_processing/aec3/shadow_filter_update_gain.h
index 05e632f..9d14807 100644
--- a/modules/audio_processing/aec3/shadow_filter_update_gain.h
+++ b/modules/audio_processing/aec3/shadow_filter_update_gain.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_SHADOW_FILTER_UPDATE_GAIN_H_
#include <stddef.h>
+
#include <array>
#include "api/audio/echo_canceller3_config.h"
diff --git a/modules/audio_processing/aec3/stationarity_estimator.h b/modules/audio_processing/aec3/stationarity_estimator.h
index 704859a..023043b 100644
--- a/modules/audio_processing/aec3/stationarity_estimator.h
+++ b/modules/audio_processing/aec3/stationarity_estimator.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
#include <stddef.h>
+
#include <array>
#include <memory>
diff --git a/modules/audio_processing/aec3/subband_erle_estimator.h b/modules/audio_processing/aec3/subband_erle_estimator.h
index 903c629..0a22d61 100644
--- a/modules/audio_processing/aec3/subband_erle_estimator.h
+++ b/modules/audio_processing/aec3/subband_erle_estimator.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_SUBBAND_ERLE_ESTIMATOR_H_
#include <stddef.h>
+
#include <array>
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/aec3/subtractor.h b/modules/audio_processing/aec3/subtractor.h
index 910be18..ccff7c1 100644
--- a/modules/audio_processing/aec3/subtractor.h
+++ b/modules/audio_processing/aec3/subtractor.h
@@ -13,6 +13,7 @@
#include <math.h>
#include <stddef.h>
+
#include <array>
#include <vector>
diff --git a/modules/audio_processing/aec3/suppression_filter_unittest.cc b/modules/audio_processing/aec3/suppression_filter_unittest.cc
index 2c745ad..80d96ec 100644
--- a/modules/audio_processing/aec3/suppression_filter_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_filter_unittest.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/aec3/suppression_filter.h"
#include <math.h>
+
#include <algorithm>
#include <cmath>
#include <numeric>
diff --git a/modules/audio_processing/aec3/suppression_gain.cc b/modules/audio_processing/aec3/suppression_gain.cc
index b741a71..4831b71 100644
--- a/modules/audio_processing/aec3/suppression_gain.cc
+++ b/modules/audio_processing/aec3/suppression_gain.cc
@@ -12,6 +12,7 @@
#include <math.h>
#include <stddef.h>
+
#include <algorithm>
#include <numeric>
@@ -264,9 +265,9 @@
std::array<float, kFftLengthBy2Plus1> max_gain;
GetMaxGain(max_gain);
- GainToNoAudibleEcho(nearend, weighted_residual_echo, comfort_noise,
- min_gain, max_gain, gain);
- AdjustForExternalFilters(gain);
+ GainToNoAudibleEcho(nearend, weighted_residual_echo, comfort_noise, min_gain,
+ max_gain, gain);
+ AdjustForExternalFilters(gain);
// Adjust the gain for frequencies which have not yet converged.
AdjustNonConvergedFrequencies(gain);
diff --git a/modules/audio_processing/aec3/vector_buffer.h b/modules/audio_processing/aec3/vector_buffer.h
index 4c0257c..9d1539f 100644
--- a/modules/audio_processing/aec3/vector_buffer.h
+++ b/modules/audio_processing/aec3/vector_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AEC3_VECTOR_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aec3/vector_math.h b/modules/audio_processing/aec3/vector_math.h
index 255331b..883cd95 100644
--- a/modules/audio_processing/aec3/vector_math.h
+++ b/modules/audio_processing/aec3/vector_math.h
@@ -21,6 +21,7 @@
#include <emmintrin.h>
#endif
#include <math.h>
+
#include <algorithm>
#include <array>
#include <functional>
diff --git a/modules/audio_processing/aec_dump/aec_dump_impl.cc b/modules/audio_processing/aec_dump/aec_dump_impl.cc
index ba15336..904033a 100644
--- a/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <utility>
-
#include "modules/audio_processing/aec_dump/aec_dump_impl.h"
+#include <utility>
+
#include "absl/memory/memory.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
index 561fa62..3624bfc 100644
--- a/modules/audio_processing/aec_dump/aec_dump_unittest.cc
+++ b/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -11,7 +11,6 @@
#include <utility>
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
-
#include "rtc_base/task_queue_for_test.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
diff --git a/modules/audio_processing/aecm/aecm_core.cc b/modules/audio_processing/aecm/aecm_core.cc
index 67b70bf..78d8dfd 100644
--- a/modules/audio_processing/aecm/aecm_core.cc
+++ b/modules/audio_processing/aecm/aecm_core.cc
@@ -21,7 +21,6 @@
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aecm/echo_control_mobile.h"
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
-
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -440,9 +439,8 @@
aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
aecm->farEnergyMaxMin = 0;
- aecm->farEnergyVAD =
- FAR_ENERGY_MIN; // This prevents false speech detection at the
- // beginning.
+ aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection
+ // at the beginning.
aecm->farEnergyMSE = 0;
aecm->currentVADValue = 0;
aecm->vadUpdateCount = 0;
diff --git a/modules/audio_processing/aecm/aecm_core_c.cc b/modules/audio_processing/aecm/aecm_core_c.cc
index 905274f..2727182 100644
--- a/modules/audio_processing/aecm/aecm_core_c.cc
+++ b/modules/audio_processing/aecm/aecm_core_c.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/aecm/aecm_core.h"
-
#include <stddef.h>
#include <stdlib.h>
+#include "modules/audio_processing/aecm/aecm_core.h"
+
extern "C" {
#include "common_audio/ring_buffer.h"
#include "common_audio/signal_processing/include/real_fft.h"
@@ -198,11 +198,11 @@
} else if (freq_signal[i].imag == 0) {
freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
} else {
-// Approximation for magnitude of complex fft output
-// magn = sqrt(real^2 + imag^2)
-// magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
-//
-// The parameters alpha and beta are stored in Q15
+ // Approximation for magnitude of complex fft output
+ // magn = sqrt(real^2 + imag^2)
+ // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
+ //
+ // The parameters alpha and beta are stored in Q15
#ifdef AECM_WITH_ABS_APPROX
tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
diff --git a/modules/audio_processing/aecm/aecm_core_mips.cc b/modules/audio_processing/aecm/aecm_core_mips.cc
index 11e4095..75aee91 100644
--- a/modules/audio_processing/aecm/aecm_core_mips.cc
+++ b/modules/audio_processing/aecm/aecm_core_mips.cc
@@ -9,7 +9,6 @@
*/
#include "modules/audio_processing/aecm/aecm_core.h"
-
#include "modules/audio_processing/aecm/echo_control_mobile.h"
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/aecm/aecm_core_neon.cc b/modules/audio_processing/aecm/aecm_core_neon.cc
index a2153a2..94a318b 100644
--- a/modules/audio_processing/aecm/aecm_core_neon.cc
+++ b/modules/audio_processing/aecm/aecm_core_neon.cc
@@ -8,11 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/aecm/aecm_core.h"
-
#include <arm_neon.h>
#include "common_audio/signal_processing/include/real_fft.h"
+#include "modules/audio_processing/aecm/aecm_core.h"
#include "rtc_base/checks.h"
// TODO(kma): Re-write the corresponding assembly file, the offset
diff --git a/modules/audio_processing/agc/loudness_histogram.cc b/modules/audio_processing/agc/loudness_histogram.cc
index cd57b82..4775ff7 100644
--- a/modules/audio_processing/agc/loudness_histogram.cc
+++ b/modules/audio_processing/agc/loudness_histogram.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc/loudness_histogram.h"
#include <string.h>
+
#include <cmath>
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/agc/loudness_histogram.h b/modules/audio_processing/agc/loudness_histogram.h
index b210be9..badd443 100644
--- a/modules/audio_processing/agc/loudness_histogram.h
+++ b/modules/audio_processing/agc/loudness_histogram.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AGC_LOUDNESS_HISTOGRAM_H_
#include <stdint.h>
+
#include <memory>
namespace webrtc {
diff --git a/modules/audio_processing/agc/loudness_histogram_unittest.cc b/modules/audio_processing/agc/loudness_histogram_unittest.cc
index 0c291d8..30ea5d3 100644
--- a/modules/audio_processing/agc/loudness_histogram_unittest.cc
+++ b/modules/audio_processing/agc/loudness_histogram_unittest.cc
@@ -13,6 +13,7 @@
#include "modules/audio_processing/agc/loudness_histogram.h"
#include <stdio.h>
+
#include <algorithm>
#include <cmath>
#include <memory>
diff --git a/modules/audio_processing/agc/mock_agc.h b/modules/audio_processing/agc/mock_agc.h
index 4297e2a..d31c265 100644
--- a/modules/audio_processing/agc/mock_agc.h
+++ b/modules/audio_processing/agc/mock_agc.h
@@ -12,7 +12,6 @@
#define MODULES_AUDIO_PROCESSING_AGC_MOCK_AGC_H_
#include "modules/audio_processing/agc/agc.h"
-
#include "test/gmock.h"
namespace webrtc {
diff --git a/modules/audio_processing/agc2/agc2_common.cc b/modules/audio_processing/agc2/agc2_common.cc
index 1107885..3f697d1 100644
--- a/modules/audio_processing/agc2/agc2_common.cc
+++ b/modules/audio_processing/agc2/agc2_common.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc2/agc2_common.h"
#include <stdio.h>
+
#include <string>
#include "system_wrappers/include/field_trial.h"
diff --git a/modules/audio_processing/agc2/agc2_testing_common_unittest.cc b/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
index b9f7126..f52ea3c 100644
--- a/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
+++ b/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/agc2/agc2_testing_common.h"
+
#include "rtc_base/gunit.h"
namespace webrtc {
diff --git a/modules/audio_processing/agc2/down_sampler.cc b/modules/audio_processing/agc2/down_sampler.cc
index 50486e0..654ed4b 100644
--- a/modules/audio_processing/agc2/down_sampler.cc
+++ b/modules/audio_processing/agc2/down_sampler.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc2/down_sampler.h"
#include <string.h>
+
#include <algorithm>
#include "modules/audio_processing/agc2/biquad_filter.h"
diff --git a/modules/audio_processing/agc2/interpolated_gain_curve.cc b/modules/audio_processing/agc2/interpolated_gain_curve.cc
index f5d6b47..502e702 100644
--- a/modules/audio_processing/agc2/interpolated_gain_curve.cc
+++ b/modules/audio_processing/agc2/interpolated_gain_curve.cc
@@ -113,7 +113,9 @@
}
break;
}
- default: { RTC_NOTREACHED(); }
+ default: {
+ RTC_NOTREACHED();
+ }
}
}
diff --git a/modules/audio_processing/agc2/interpolated_gain_curve.h b/modules/audio_processing/agc2/interpolated_gain_curve.h
index 1ecb94e..ef1c027 100644
--- a/modules/audio_processing/agc2/interpolated_gain_curve.h
+++ b/modules/audio_processing/agc2/interpolated_gain_curve.h
@@ -15,7 +15,6 @@
#include <string>
#include "modules/audio_processing/agc2/agc2_common.h"
-
#include "rtc_base/constructor_magic.h"
#include "rtc_base/gtest_prod_util.h"
#include "system_wrappers/include/metrics.h"
diff --git a/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc b/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
index a8e0f23..67d34e5 100644
--- a/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
+++ b/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
+
#include <array>
#include <type_traits>
#include <vector>
@@ -16,7 +18,6 @@
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
-#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/agc2/limiter_db_gain_curve.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/agc2/noise_level_estimator.cc b/modules/audio_processing/agc2/noise_level_estimator.cc
index 6e43672..2ca5034 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc2/noise_level_estimator.h"
#include <stddef.h>
+
#include <algorithm>
#include <cmath>
#include <numeric>
diff --git a/modules/audio_processing/agc2/noise_spectrum_estimator.cc b/modules/audio_processing/agc2/noise_spectrum_estimator.cc
index 5735faf..31438b1 100644
--- a/modules/audio_processing/agc2/noise_spectrum_estimator.cc
+++ b/modules/audio_processing/agc2/noise_spectrum_estimator.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
#include <string.h>
+
#include <algorithm>
#include "api/array_view.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
index 0561c37..af3619b 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
#include <stdlib.h>
+
#include <algorithm>
#include <cmath>
#include <cstddef>
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
index 6ccd165..2cc5ce6 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
#include <stddef.h>
+
#include <array>
#include "api/array_view.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc b/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
index 494dfe7..99c9dfa 100644
--- a/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
@@ -9,12 +9,12 @@
*/
#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
-#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
-#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
#include <algorithm>
#include <vector>
+#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
// #include "test/fpe_observer.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.cc b/modules/audio_processing/agc2/rnn_vad/rnn.cc
index 2b36034..a5b34c4 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.cc
@@ -25,21 +25,21 @@
using rnnoise::kInputLayerInputSize;
static_assert(kFeatureVectorSize == kInputLayerInputSize, "");
-using rnnoise::kInputDenseWeights;
using rnnoise::kInputDenseBias;
+using rnnoise::kInputDenseWeights;
using rnnoise::kInputLayerOutputSize;
static_assert(kInputLayerOutputSize <= kFullyConnectedLayersMaxUnits,
"Increase kFullyConnectedLayersMaxUnits.");
+using rnnoise::kHiddenGruBias;
using rnnoise::kHiddenGruRecurrentWeights;
using rnnoise::kHiddenGruWeights;
-using rnnoise::kHiddenGruBias;
using rnnoise::kHiddenLayerOutputSize;
static_assert(kHiddenLayerOutputSize <= kRecurrentLayersMaxUnits,
"Increase kRecurrentLayersMaxUnits.");
-using rnnoise::kOutputDenseWeights;
using rnnoise::kOutputDenseBias;
+using rnnoise::kOutputDenseWeights;
using rnnoise::kOutputLayerOutputSize;
static_assert(kOutputLayerOutputSize <= kFullyConnectedLayersMaxUnits,
"Increase kFullyConnectedLayersMaxUnits.");
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn.h b/modules/audio_processing/agc2/rnn_vad/rnn.h
index a7d057d..1129464 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn.h
+++ b/modules/audio_processing/agc2/rnn_vad/rnn.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <sys/types.h>
+
#include <array>
#include "api/array_view.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc b/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
index 933b555..40ac70b 100644
--- a/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
+++ b/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
+
#include <array>
-#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
#include "rtc_base/checks.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
index 24b0219..ed4caad 100644
--- a/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
+++ b/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
#include <stddef.h>
+
#include <array>
#include <vector>
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 0c38a4f..1a99463 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/audio_buffer.h"
#include <string.h>
+
#include <cstdint>
#include "common_audio/channel_buffer.h"
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index a85144b..8fba9f9 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/audio_buffer_unittest.cc b/modules/audio_processing/audio_buffer_unittest.cc
index 4cbb98e..5c23159 100644
--- a/modules/audio_processing/audio_buffer_unittest.cc
+++ b/modules/audio_processing/audio_buffer_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/audio_buffer.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index a700038..9b4ae81 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1492,7 +1492,9 @@
TRACE_EVENT0("webrtc", "AudioProcessing::AnalyzeReverseStream_ChannelLayout");
rtc::CritScope cs(&crit_render_);
const StreamConfig reverse_config = {
- sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
+ sample_rate_hz,
+ ChannelsFromLayout(layout),
+ LayoutHasKeyboard(layout),
};
if (samples_per_channel != reverse_config.num_frames()) {
return kBadDataLengthError;
diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index 9063980..9182d2c 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -8,13 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/audio_processing_impl.h"
-
#include <algorithm>
#include <memory>
#include <vector>
#include "api/array_view.h"
+#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/event.h"
diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
index 993b8b6..4e297a5 100644
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ b/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -7,8 +7,6 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/audio_processing_impl.h"
-
#include <math.h>
#include <algorithm>
@@ -16,6 +14,7 @@
#include <vector>
#include "api/array_view.h"
+#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/event.h"
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 2c23cb3..831799f 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -7,6 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/include/audio_processing.h"
+
#include <math.h>
#include <stdio.h>
@@ -23,7 +25,6 @@
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/common.h"
-#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "modules/audio_processing/test/protobuf_utils.h"
#include "modules/audio_processing/test/test_utils.h"
@@ -78,16 +79,11 @@
enum StreamDirection { kForward = 0, kReverse };
void ConvertToFloat(const int16_t* int_data, ChannelBuffer<float>* cb) {
- ChannelBuffer<int16_t> cb_int(cb->num_frames(),
- cb->num_channels());
- Deinterleave(int_data,
- cb->num_frames(),
- cb->num_channels(),
+ ChannelBuffer<int16_t> cb_int(cb->num_frames(), cb->num_channels());
+ Deinterleave(int_data, cb->num_frames(), cb->num_channels(),
cb_int.channels());
for (size_t i = 0; i < cb->num_channels(); ++i) {
- S16ToFloat(cb_int.channels()[i],
- cb->num_frames(),
- cb->channels()[i]);
+ S16ToFloat(cb_int.channels()[i], cb->num_frames(), cb->channels()[i]);
}
}
@@ -110,13 +106,15 @@
return 0;
}
-void MixStereoToMono(const float* stereo, float* mono,
+void MixStereoToMono(const float* stereo,
+ float* mono,
size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; ++i)
mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) / 2;
}
-void MixStereoToMono(const int16_t* stereo, int16_t* mono,
+void MixStereoToMono(const int16_t* stereo,
+ int16_t* mono,
size_t samples_per_channel) {
for (size_t i = 0; i < samples_per_channel; ++i)
mono[i] = (stereo[i * 2] + stereo[i * 2 + 1]) >> 1;
@@ -206,7 +204,7 @@
// These functions are only used by ApmTest.Process.
template <class T>
T AbsValue(T a) {
- return a > 0 ? a: -a;
+ return a > 0 ? a : -a;
}
int16_t MaxAudioFrame(const AudioFrame& frame) {
@@ -232,7 +230,7 @@
ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
ASSERT_EQ(static_cast<size_t>(size),
- fwrite(array.get(), sizeof(array[0]), size, file));
+ fwrite(array.get(), sizeof(array[0]), size, file));
fclose(file);
}
@@ -317,7 +315,9 @@
//
// |int_data| and |float_data| are just temporary space that must be
// sufficiently large to hold the 10 ms chunk.
-bool ReadChunk(FILE* file, int16_t* int_data, float* float_data,
+bool ReadChunk(FILE* file,
+ int16_t* int_data,
+ float* float_data,
ChannelBuffer<float>* cb) {
// The files always contain stereo audio.
size_t frame_size = cb->num_frames() * 2;
@@ -332,8 +332,7 @@
if (cb->num_channels() == 1) {
MixStereoToMono(float_data, cb->channels()[0], cb->num_frames());
} else {
- Deinterleave(float_data, cb->num_frames(), 2,
- cb->channels());
+ Deinterleave(float_data, cb->num_frames(), 2, cb->channels());
}
return true;
@@ -350,10 +349,7 @@
static void TearDownTestSuite() { ClearTempFiles(); }
// Used to select between int and float interface tests.
- enum Format {
- kIntFormat,
- kFloatFormat
- };
+ enum Format { kIntFormat, kFloatFormat };
void Init(int sample_rate_hz,
int output_sample_rate_hz,
@@ -367,11 +363,14 @@
bool ReadFrame(FILE* file, AudioFrame* frame);
bool ReadFrame(FILE* file, AudioFrame* frame, ChannelBuffer<float>* cb);
void ReadFrameWithRewind(FILE* file, AudioFrame* frame);
- void ReadFrameWithRewind(FILE* file, AudioFrame* frame,
+ void ReadFrameWithRewind(FILE* file,
+ AudioFrame* frame,
ChannelBuffer<float>* cb);
void ProcessWithDefaultStreamParameters(AudioFrame* frame);
- void ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
- int delay_min, int delay_max);
+ void ProcessDelayVerificationTest(int delay_ms,
+ int system_delay_ms,
+ int delay_min,
+ int delay_max);
void TestChangingChannelsInt16Interface(
size_t num_channels,
AudioProcessing::Error expected_return);
@@ -408,11 +407,11 @@
ApmTest::ApmTest()
: output_path_(test::OutputPath()),
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- ref_filename_(test::ResourcePath("audio_processing/output_data_fixed",
- "pb")),
+ ref_filename_(
+ test::ResourcePath("audio_processing/output_data_fixed", "pb")),
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- ref_filename_(test::ResourcePath("audio_processing/output_data_float",
- "pb")),
+ ref_filename_(
+ test::ResourcePath("audio_processing/output_data_float", "pb")),
#endif
frame_(NULL),
revframe_(NULL),
@@ -491,16 +490,14 @@
}
std::string filename = ResourceFilePath("far", sample_rate_hz);
far_file_ = fopen(filename.c_str(), "rb");
- ASSERT_TRUE(far_file_ != NULL) << "Could not open file " <<
- filename << "\n";
+ ASSERT_TRUE(far_file_ != NULL) << "Could not open file " << filename << "\n";
if (near_file_) {
ASSERT_EQ(0, fclose(near_file_));
}
filename = ResourceFilePath("near", sample_rate_hz);
near_file_ = fopen(filename.c_str(), "rb");
- ASSERT_TRUE(near_file_ != NULL) << "Could not open file " <<
- filename << "\n";
+ ASSERT_TRUE(near_file_ != NULL) << "Could not open file " << filename << "\n";
if (open_output_file) {
if (out_file_) {
@@ -511,8 +508,8 @@
reverse_sample_rate_hz, num_input_channels, num_output_channels,
num_reverse_channels, num_reverse_channels, kForward);
out_file_ = fopen(filename.c_str(), "wb");
- ASSERT_TRUE(out_file_ != NULL) << "Could not open file " <<
- filename << "\n";
+ ASSERT_TRUE(out_file_ != NULL)
+ << "Could not open file " << filename << "\n";
}
}
@@ -520,14 +517,13 @@
EnableAllAPComponents(apm_.get());
}
-bool ApmTest::ReadFrame(FILE* file, AudioFrame* frame,
+bool ApmTest::ReadFrame(FILE* file,
+ AudioFrame* frame,
ChannelBuffer<float>* cb) {
// The files always contain stereo audio.
size_t frame_size = frame->samples_per_channel_ * 2;
- size_t read_count = fread(frame->mutable_data(),
- sizeof(int16_t),
- frame_size,
- file);
+ size_t read_count =
+ fread(frame->mutable_data(), sizeof(int16_t), frame_size, file);
if (read_count != frame_size) {
// Check that the file really ended.
EXPECT_NE(0, feof(file));
@@ -551,7 +547,8 @@
// If the end of the file has been reached, rewind it and attempt to read the
// frame again.
-void ApmTest::ReadFrameWithRewind(FILE* file, AudioFrame* frame,
+void ApmTest::ReadFrameWithRewind(FILE* file,
+ AudioFrame* frame,
ChannelBuffer<float>* cb) {
if (!ReadFrame(near_file_, frame_, cb)) {
rewind(near_file_);
@@ -565,8 +562,7 @@
void ApmTest::ProcessWithDefaultStreamParameters(AudioFrame* frame) {
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame));
}
@@ -574,13 +570,11 @@
if (format == kIntFormat) {
return apm_->ProcessStream(frame_);
}
- return apm_->ProcessStream(float_cb_->channels(),
- frame_->samples_per_channel_,
- frame_->sample_rate_hz_,
- LayoutFromChannels(frame_->num_channels_),
- output_sample_rate_hz_,
- LayoutFromChannels(num_output_channels_),
- float_cb_->channels());
+ return apm_->ProcessStream(
+ float_cb_->channels(), frame_->samples_per_channel_,
+ frame_->sample_rate_hz_, LayoutFromChannels(frame_->num_channels_),
+ output_sample_rate_hz_, LayoutFromChannels(num_output_channels_),
+ float_cb_->channels());
}
int ApmTest::AnalyzeReverseStreamChooser(Format format) {
@@ -588,14 +582,14 @@
return apm_->ProcessReverseStream(revframe_);
}
return apm_->AnalyzeReverseStream(
- revfloat_cb_->channels(),
- revframe_->samples_per_channel_,
- revframe_->sample_rate_hz_,
- LayoutFromChannels(revframe_->num_channels_));
+ revfloat_cb_->channels(), revframe_->samples_per_channel_,
+ revframe_->sample_rate_hz_, LayoutFromChannels(revframe_->num_channels_));
}
-void ApmTest::ProcessDelayVerificationTest(int delay_ms, int system_delay_ms,
- int delay_min, int delay_max) {
+void ApmTest::ProcessDelayVerificationTest(int delay_ms,
+ int system_delay_ms,
+ int delay_min,
+ int delay_max) {
// The |revframe_| and |frame_| should include the proper frame information,
// hence can be used for extracting information.
AudioFrame tmp_frame;
@@ -687,15 +681,12 @@
// -- Missing AGC level --
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- ProcessStreamChooser(format));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Resets after successful ProcessStream().
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- ProcessStreamChooser(format));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
// Other stream parameters set correctly.
AudioProcessing::Config apm_config = apm_->GetConfig();
@@ -703,8 +694,7 @@
apm_config.echo_canceller.mobile_mode = false;
apm_->ApplyConfig(apm_config);
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- ProcessStreamChooser(format));
+ EXPECT_EQ(apm_->kStreamParameterNotSetError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// -- Missing delay --
@@ -718,20 +708,17 @@
// Other stream parameters set correctly.
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
// -- No stream parameters --
- EXPECT_EQ(apm_->kNoError,
- AnalyzeReverseStreamChooser(format));
+ EXPECT_EQ(apm_->kNoError, AnalyzeReverseStreamChooser(format));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
// -- All there --
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_stream_analog_level(127));
EXPECT_EQ(apm_->kNoError, ProcessStreamChooser(format));
}
@@ -856,40 +843,34 @@
TEST_F(ApmTest, GainControl) {
// Testing gain modes
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(
- apm_->gain_control()->mode()));
+ apm_->gain_control()->set_mode(apm_->gain_control()->mode()));
- GainControl::Mode mode[] = {
- GainControl::kAdaptiveAnalog,
- GainControl::kAdaptiveDigital,
- GainControl::kFixedDigital
- };
+ GainControl::Mode mode[] = {GainControl::kAdaptiveAnalog,
+ GainControl::kAdaptiveDigital,
+ GainControl::kFixedDigital};
for (size_t i = 0; i < arraysize(mode); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(mode[i]));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_mode(mode[i]));
EXPECT_EQ(mode[i], apm_->gain_control()->mode());
}
// Testing target levels
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(
- apm_->gain_control()->target_level_dbfs()));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_target_level_dbfs(
+ apm_->gain_control()->target_level_dbfs()));
int level_dbfs[] = {0, 6, 31};
for (size_t i = 0; i < arraysize(level_dbfs); i++) {
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
+ apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
}
// Testing compression gains
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(
- apm_->gain_control()->compression_gain_db()));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_compression_gain_db(
+ apm_->gain_control()->compression_gain_db()));
int gain_db[] = {0, 10, 90};
for (size_t i = 0; i < arraysize(gain_db); i++) {
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(gain_db[i]));
+ apm_->gain_control()->set_compression_gain_db(gain_db[i]));
ProcessStreamChooser(kFloatFormat);
EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
}
@@ -901,22 +882,21 @@
EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
// Testing level limits
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(
- apm_->gain_control()->analog_level_minimum(),
- apm_->gain_control()->analog_level_maximum()));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(
+ apm_->gain_control()->analog_level_minimum(),
+ apm_->gain_control()->analog_level_maximum()));
int min_level[] = {0, 255, 1024};
for (size_t i = 0; i < arraysize(min_level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
+ EXPECT_EQ(apm_->kNoError, apm_->gain_control()->set_analog_level_limits(
+ min_level[i], 1024));
EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
}
int max_level[] = {0, 1024, 65535};
for (size_t i = 0; i < arraysize(min_level); i++) {
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
+ apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
}
@@ -981,7 +961,7 @@
// Always pass in the same volume.
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(100));
+ apm_->gain_control()->set_stream_analog_level(100));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->stream_analog_level();
}
@@ -1011,7 +991,7 @@
ScaleFrame(frame_, 0.25);
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(out_analog_level));
+ apm_->gain_control()->set_stream_analog_level(out_analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->stream_analog_level();
}
@@ -1027,7 +1007,7 @@
ScaleFrame(frame_, 0.25);
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(out_analog_level));
+ apm_->gain_control()->set_stream_analog_level(out_analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
out_analog_level = apm_->gain_control()->stream_analog_level();
// Check that AGC respected the manually adjusted volume.
@@ -1046,14 +1026,10 @@
TEST_F(ApmTest, NoiseSuppression) {
// Test valid suppression levels.
NoiseSuppression::Level level[] = {
- NoiseSuppression::kLow,
- NoiseSuppression::kModerate,
- NoiseSuppression::kHigh,
- NoiseSuppression::kVeryHigh
- };
+ NoiseSuppression::kLow, NoiseSuppression::kModerate,
+ NoiseSuppression::kHigh, NoiseSuppression::kVeryHigh};
for (size_t i = 0; i < arraysize(level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->noise_suppression()->set_level(level[i]));
+ EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->set_level(level[i]));
EXPECT_EQ(level[i], apm_->noise_suppression()->level());
}
@@ -1149,11 +1125,8 @@
// Test valid likelihoods
VoiceDetection::Likelihood likelihood[] = {
- VoiceDetection::kVeryLowLikelihood,
- VoiceDetection::kLowLikelihood,
- VoiceDetection::kModerateLikelihood,
- VoiceDetection::kHighLikelihood
- };
+ VoiceDetection::kVeryLowLikelihood, VoiceDetection::kLowLikelihood,
+ VoiceDetection::kModerateLikelihood, VoiceDetection::kHighLikelihood};
for (size_t i = 0; i < arraysize(likelihood); i++) {
EXPECT_EQ(apm_->kNoError,
apm_->voice_detection()->set_likelihood(likelihood[i]));
@@ -1182,10 +1155,7 @@
// Test that AudioFrame activity is maintained when VAD is disabled.
EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
AudioFrame::VADActivity activity[] = {
- AudioFrame::kVadActive,
- AudioFrame::kVadPassive,
- AudioFrame::kVadUnknown
- };
+ AudioFrame::kVadActive, AudioFrame::kVadPassive, AudioFrame::kVadUnknown};
for (size_t i = 0; i < arraysize(activity); i++) {
frame_->vad_activity_ = activity[i];
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@@ -1232,18 +1202,16 @@
// Test that ProcessStream copies input to output even with no processing.
const size_t kSamples = 80;
const int sample_rate = 8000;
- const float src[kSamples] = {
- -1.0f, 0.0f, 1.0f
- };
+ const float src[kSamples] = {-1.0f, 0.0f, 1.0f};
float dest[kSamples] = {};
auto src_channels = &src[0];
auto dest_channels = &dest[0];
apm_.reset(AudioProcessingBuilder().Create());
- EXPECT_NOERR(apm_->ProcessStream(
- &src_channels, kSamples, sample_rate, LayoutFromChannels(1),
- sample_rate, LayoutFromChannels(1), &dest_channels));
+ EXPECT_NOERR(apm_->ProcessStream(&src_channels, kSamples, sample_rate,
+ LayoutFromChannels(1), sample_rate,
+ LayoutFromChannels(1), &dest_channels));
for (size_t i = 0; i < kSamples; ++i) {
EXPECT_EQ(src[i], dest[i]);
@@ -1267,13 +1235,8 @@
EnableAllComponents();
for (size_t i = 0; i < arraysize(kProcessSampleRates); i++) {
- Init(kProcessSampleRates[i],
- kProcessSampleRates[i],
- kProcessSampleRates[i],
- 2,
- 2,
- 2,
- false);
+ Init(kProcessSampleRates[i], kProcessSampleRates[i], kProcessSampleRates[i],
+ 2, 2, 2, false);
int analog_level = 127;
ASSERT_EQ(0, feof(far_file_));
ASSERT_EQ(0, feof(near_file_));
@@ -1289,7 +1252,7 @@
ASSERT_EQ(kNoErr, apm_->set_stream_delay_ms(0));
ASSERT_EQ(kNoErr,
- apm_->gain_control()->set_stream_analog_level(analog_level));
+ apm_->gain_control()->set_stream_analog_level(analog_level));
ASSERT_EQ(kNoErr, apm_->ProcessStream(frame_));
analog_level = apm_->gain_control()->stream_analog_level();
@@ -1393,13 +1356,9 @@
output_sample_rate = msg.output_sample_rate();
}
- Init(msg.sample_rate(),
- output_sample_rate,
- reverse_sample_rate,
- msg.num_input_channels(),
- msg.num_output_channels(),
- msg.num_reverse_channels(),
- false);
+ Init(msg.sample_rate(), output_sample_rate, reverse_sample_rate,
+ msg.num_input_channels(), msg.num_output_channels(),
+ msg.num_reverse_channels(), false);
if (first_init) {
// AttachAecDump() writes an additional init message. Don't start
// recording until after the first init to avoid the extra message.
@@ -1417,9 +1376,8 @@
ASSERT_EQ(revframe_->num_channels_,
static_cast<size_t>(msg.channel_size()));
for (int i = 0; i < msg.channel_size(); ++i) {
- memcpy(revfloat_cb_->channels()[i],
- msg.channel(i).data(),
- msg.channel(i).size());
+ memcpy(revfloat_cb_->channels()[i], msg.channel(i).data(),
+ msg.channel(i).size());
}
} else {
memcpy(revframe_->mutable_data(), msg.data().data(), msg.data().size());
@@ -1447,9 +1405,8 @@
ASSERT_EQ(frame_->num_channels_,
static_cast<size_t>(msg.input_channel_size()));
for (int i = 0; i < msg.input_channel_size(); ++i) {
- memcpy(float_cb_->channels()[i],
- msg.input_channel(i).data(),
- msg.input_channel(i).size());
+ memcpy(float_cb_->channels()[i], msg.input_channel(i).data(),
+ msg.input_channel(i).size());
}
} else {
memcpy(frame_->mutable_data(), msg.input_data().data(),
@@ -1656,13 +1613,10 @@
EnableAllComponents();
- Init(test->sample_rate(),
- test->sample_rate(),
- test->sample_rate(),
+ Init(test->sample_rate(), test->sample_rate(), test->sample_rate(),
static_cast<size_t>(test->num_input_channels()),
static_cast<size_t>(test->num_output_channels()),
- static_cast<size_t>(test->num_reverse_channels()),
- true);
+ static_cast<size_t>(test->num_reverse_channels()), true);
int frame_count = 0;
int has_voice_count = 0;
@@ -1673,7 +1627,7 @@
float ns_speech_prob_average = 0.0f;
float rms_dbfs_average = 0.0f;
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- int stats_index = 0;
+ int stats_index = 0;
#endif
while (ReadFrame(far_file_, revframe_) && ReadFrame(near_file_, frame_)) {
@@ -1683,7 +1637,7 @@
EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(analog_level));
+ apm_->gain_control()->set_stream_analog_level(analog_level));
EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
@@ -1711,10 +1665,8 @@
rms_dbfs_average += *stats.output_rms_dbfs;
size_t frame_size = frame_->samples_per_channel_ * frame_->num_channels_;
- size_t write_count = fwrite(frame_->data(),
- sizeof(int16_t),
- frame_size,
- out_file_);
+ size_t write_count =
+ fwrite(frame_->data(), sizeof(int16_t), frame_size, out_file_);
ASSERT_EQ(frame_size, write_count);
// Reset in case of downmixing.
@@ -1787,8 +1739,7 @@
const int kMaxOutputAverageNear = kIntNear;
#endif
EXPECT_NEAR(test->has_voice_count(),
- has_voice_count - kHasVoiceCountOffset,
- kHasVoiceCountNear);
+ has_voice_count - kHasVoiceCountOffset, kHasVoiceCountNear);
EXPECT_NEAR(test->is_saturated_count(), is_saturated_count, kIntNear);
EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear);
@@ -1797,8 +1748,7 @@
kMaxOutputAverageNear);
#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const double kFloatNear = 0.0005;
- EXPECT_NEAR(test->ns_speech_probability_average(),
- ns_speech_prob_average,
+ EXPECT_NEAR(test->ns_speech_probability_average(), ns_speech_prob_average,
kFloatNear);
EXPECT_NEAR(test->rms_dbfs_average(), rms_dbfs_average, kFloatNear);
#endif
@@ -1832,9 +1782,9 @@
AudioProcessing::ChannelLayout out_layout;
};
ChannelFormat cf[] = {
- {AudioProcessing::kMonoAndKeyboard, AudioProcessing::kMono},
- {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
- {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
+ {AudioProcessing::kMonoAndKeyboard, AudioProcessing::kMono},
+ {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
+ {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
};
std::unique_ptr<AudioProcessing> ap(AudioProcessingBuilder().Create());
@@ -1850,14 +1800,9 @@
// Run over a few chunks.
for (int j = 0; j < 10; ++j) {
- EXPECT_NOERR(ap->ProcessStream(
- in_cb.channels(),
- in_cb.num_frames(),
- in_rate,
- cf[i].in_layout,
- out_rate,
- cf[i].out_layout,
- out_cb.channels()));
+ EXPECT_NOERR(ap->ProcessStream(in_cb.channels(), in_cb.num_frames(),
+ in_rate, cf[i].in_layout, out_rate,
+ cf[i].out_layout, out_cb.channels()));
}
}
}
@@ -1978,20 +1923,20 @@
FILE* far_file =
fopen(ResourceFilePath("far", reverse_input_rate).c_str(), "rb");
FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb");
- FILE* out_file =
- fopen(OutputFilePath(output_file_prefix, input_rate, output_rate,
- reverse_input_rate, reverse_output_rate,
- num_input_channels, num_output_channels,
- num_reverse_input_channels,
- num_reverse_output_channels, kForward).c_str(),
- "wb");
- FILE* rev_out_file =
- fopen(OutputFilePath(output_file_prefix, input_rate, output_rate,
- reverse_input_rate, reverse_output_rate,
- num_input_channels, num_output_channels,
- num_reverse_input_channels,
- num_reverse_output_channels, kReverse).c_str(),
- "wb");
+ FILE* out_file = fopen(
+ OutputFilePath(
+ output_file_prefix, input_rate, output_rate, reverse_input_rate,
+ reverse_output_rate, num_input_channels, num_output_channels,
+ num_reverse_input_channels, num_reverse_output_channels, kForward)
+ .c_str(),
+ "wb");
+ FILE* rev_out_file = fopen(
+ OutputFilePath(
+ output_file_prefix, input_rate, output_rate, reverse_input_rate,
+ reverse_output_rate, num_input_channels, num_output_channels,
+ num_reverse_input_channels, num_reverse_output_channels, kReverse)
+ .c_str(),
+ "wb");
ASSERT_TRUE(far_file != NULL);
ASSERT_TRUE(near_file != NULL);
ASSERT_TRUE(out_file != NULL);
@@ -2024,22 +1969,17 @@
EXPECT_NOERR(ap->gain_control()->set_stream_analog_level(analog_level));
EXPECT_NOERR(ap->ProcessStream(
- fwd_cb.channels(),
- fwd_cb.num_frames(),
- input_rate,
- LayoutFromChannels(num_input_channels),
- output_rate,
- LayoutFromChannels(num_output_channels),
- out_cb.channels()));
+ fwd_cb.channels(), fwd_cb.num_frames(), input_rate,
+ LayoutFromChannels(num_input_channels), output_rate,
+ LayoutFromChannels(num_output_channels), out_cb.channels()));
// Dump forward output to file.
Interleave(out_cb.channels(), out_cb.num_frames(), out_cb.num_channels(),
float_data.get());
size_t out_length = out_cb.num_channels() * out_cb.num_frames();
- ASSERT_EQ(out_length,
- fwrite(float_data.get(), sizeof(float_data[0]),
- out_length, out_file));
+ ASSERT_EQ(out_length, fwrite(float_data.get(), sizeof(float_data[0]),
+ out_length, out_file));
// Dump reverse output to file.
Interleave(rev_out_cb.channels(), rev_out_cb.num_frames(),
@@ -2047,9 +1987,8 @@
size_t rev_out_length =
rev_out_cb.num_channels() * rev_out_cb.num_frames();
- ASSERT_EQ(rev_out_length,
- fwrite(float_data.get(), sizeof(float_data[0]), rev_out_length,
- rev_out_file));
+ ASSERT_EQ(rev_out_length, fwrite(float_data.get(), sizeof(float_data[0]),
+ rev_out_length, rev_out_file));
analog_level = ap->gain_control()->stream_analog_level();
}
@@ -2076,12 +2015,8 @@
int num_reverse_output;
};
ChannelFormat cf[] = {
- {1, 1, 1, 1},
- {1, 1, 2, 1},
- {2, 1, 1, 1},
- {2, 1, 2, 1},
- {2, 2, 1, 1},
- {2, 2, 2, 2},
+ {1, 1, 1, 1}, {1, 1, 2, 1}, {2, 1, 1, 1},
+ {2, 1, 2, 1}, {2, 2, 1, 1}, {2, 2, 2, 2},
};
for (size_t i = 0; i < arraysize(cf); ++i) {
@@ -2122,15 +2057,17 @@
OutputFilePath("out", input_rate_, output_rate_, reverse_input_rate_,
reverse_output_rate_, cf[i].num_input,
cf[i].num_output, cf[i].num_reverse_input,
- cf[i].num_reverse_output, file_direction).c_str(),
+ cf[i].num_reverse_output, file_direction)
+ .c_str(),
"rb");
// The reference files always have matching input and output channels.
- FILE* ref_file = fopen(
- OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate,
- cf[i].num_output, cf[i].num_output,
- cf[i].num_reverse_output, cf[i].num_reverse_output,
- file_direction).c_str(),
- "rb");
+ FILE* ref_file =
+ fopen(OutputFilePath("ref", ref_rate, ref_rate, ref_rate, ref_rate,
+ cf[i].num_output, cf[i].num_output,
+ cf[i].num_reverse_output,
+ cf[i].num_reverse_output, file_direction)
+ .c_str(),
+ "rb");
ASSERT_TRUE(out_file != NULL);
ASSERT_TRUE(ref_file != NULL);
diff --git a/modules/audio_processing/echo_cancellation_impl.h b/modules/audio_processing/echo_cancellation_impl.h
index a80d139..1df41a7 100644
--- a/modules/audio_processing/echo_cancellation_impl.h
+++ b/modules/audio_processing/echo_cancellation_impl.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
#include <stddef.h>
+
#include <memory>
#include <string>
#include <vector>
@@ -35,7 +36,6 @@
void ProcessRenderAudio(rtc::ArrayView<const float> packed_render_audio);
int ProcessCaptureAudio(AudioBuffer* audio, int stream_delay_ms);
-
// Differences in clock speed on the primary and reverse streams can impact
// the AEC performance. On the client-side, this could be seen when different
// render and capture devices are used, particularly with webcams.
diff --git a/modules/audio_processing/echo_cancellation_impl_unittest.cc b/modules/audio_processing/echo_cancellation_impl_unittest.cc
index 1107564..a970a4e 100644
--- a/modules/audio_processing/echo_cancellation_impl_unittest.cc
+++ b/modules/audio_processing/echo_cancellation_impl_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/echo_cancellation_impl.h"
+
#include <memory>
#include "modules/audio_processing/aec/aec_core.h"
-#include "modules/audio_processing/echo_cancellation_impl.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/critical_section.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/echo_control_mobile_impl.cc b/modules/audio_processing/echo_control_mobile_impl.cc
index 0495b39..69dfafe 100644
--- a/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/modules/audio_processing/echo_control_mobile_impl.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/echo_control_mobile_impl.h"
#include <string.h>
+
#include <cstdint>
#include "modules/audio_processing/aecm/echo_control_mobile.h"
@@ -198,7 +199,7 @@
if (MapSetting(mode) == -1) {
return AudioProcessing::kBadParameterError;
}
- routing_mode_ = mode;
+ routing_mode_ = mode;
return Configure();
}
@@ -207,7 +208,7 @@
}
int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
- comfort_noise_enabled_ = enable;
+ comfort_noise_enabled_ = enable;
return Configure();
}
diff --git a/modules/audio_processing/echo_control_mobile_impl.h b/modules/audio_processing/echo_control_mobile_impl.h
index e443797..d84a15e 100644
--- a/modules/audio_processing/echo_control_mobile_impl.h
+++ b/modules/audio_processing/echo_control_mobile_impl.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/echo_detector/circular_buffer.h b/modules/audio_processing/echo_detector/circular_buffer.h
index c52311f..db1aeae 100644
--- a/modules/audio_processing/echo_detector/circular_buffer.h
+++ b/modules/audio_processing/echo_detector/circular_buffer.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
#include <stddef.h>
+
#include <vector>
#include "absl/types/optional.h"
diff --git a/modules/audio_processing/echo_detector/circular_buffer_unittest.cc b/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
index 0fa2a2b..7a234d4 100644
--- a/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
+++ b/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/echo_detector/circular_buffer.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc b/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc
index f8efc3a..8327d23 100644
--- a/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc
+++ b/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc
@@ -10,6 +10,7 @@
*/
#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/echo_detector/moving_max_unittest.cc b/modules/audio_processing/echo_detector/moving_max_unittest.cc
index b67b86f..9429127 100644
--- a/modules/audio_processing/echo_detector/moving_max_unittest.cc
+++ b/modules/audio_processing/echo_detector/moving_max_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/echo_detector/moving_max.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc b/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc
index 7e0512e..89fb938 100644
--- a/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc
+++ b/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc
@@ -10,6 +10,7 @@
*/
#include "modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/gain_control_config_proxy_unittest.cc b/modules/audio_processing/gain_control_config_proxy_unittest.cc
index 931c99f..5bd341f 100644
--- a/modules/audio_processing/gain_control_config_proxy_unittest.cc
+++ b/modules/audio_processing/gain_control_config_proxy_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/gain_control_config_proxy.h"
+
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/mock_audio_processing.h"
#include "rtc_base/critical_section.h"
diff --git a/modules/audio_processing/gain_control_impl.cc b/modules/audio_processing/gain_control_impl.cc
index 47cbe52..2ca522c 100644
--- a/modules/audio_processing/gain_control_impl.cc
+++ b/modules/audio_processing/gain_control_impl.cc
@@ -301,7 +301,6 @@
size_t num_proc_channels_local = 0u;
int sample_rate_hz_local = 0;
{
-
minimum_capture_level_ = minimum;
maximum_capture_level_ = maximum;
diff --git a/modules/audio_processing/gain_control_impl.h b/modules/audio_processing/gain_control_impl.h
index 36b84ee..99b43b5 100644
--- a/modules/audio_processing/gain_control_impl.h
+++ b/modules/audio_processing/gain_control_impl.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/gain_controller2_unittest.cc b/modules/audio_processing/gain_controller2_unittest.cc
index 46256d8..99749cc 100644
--- a/modules/audio_processing/gain_controller2_unittest.cc
+++ b/modules/audio_processing/gain_controller2_unittest.cc
@@ -8,13 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/gain_controller2.h"
+
#include <algorithm>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc2/agc2_testing_common.h"
#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/gain_controller2.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h
index b734adf..b64bf0b 100644
--- a/modules/audio_processing/include/aec_dump.h
+++ b/modules/audio_processing/include/aec_dump.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_INCLUDE_AEC_DUMP_H_
#include <stdint.h>
+
#include <string>
#include "api/audio/audio_frame.h"
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index a652dc9..4bcace2 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -20,6 +20,7 @@
#include <stddef.h> // size_t
#include <stdio.h> // FILE
#include <string.h>
+
#include <vector>
#include "absl/types/optional.h"
diff --git a/modules/audio_processing/low_cut_filter.cc b/modules/audio_processing/low_cut_filter.cc
index 12a6e73..1ee955d 100644
--- a/modules/audio_processing/low_cut_filter.cc
+++ b/modules/audio_processing/low_cut_filter.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/low_cut_filter.h"
#include <stdint.h>
+
#include <cstring>
#include "common_audio/signal_processing/include/signal_processing_library.h"
diff --git a/modules/audio_processing/low_cut_filter_unittest.cc b/modules/audio_processing/low_cut_filter_unittest.cc
index d7b3cb9..ea4fb67 100644
--- a/modules/audio_processing/low_cut_filter_unittest.cc
+++ b/modules/audio_processing/low_cut_filter_unittest.cc
@@ -7,11 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/low_cut_filter.h"
+
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
-#include "modules/audio_processing/low_cut_filter.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "modules/audio_processing/test/bitexactness_tools.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/ns/defines.h b/modules/audio_processing/ns/defines.h
index d6abfea..2935f25 100644
--- a/modules/audio_processing/ns/defines.h
+++ b/modules/audio_processing/ns/defines.h
@@ -46,7 +46,8 @@
(float)0.5 // default threshold for Spectral Flatness feature
#define SD_FEATURE_THR \
(float)0.5 // default threshold for Spectral Difference feature
-#define PROB_RANGE (float)0.20 // probability threshold for noise state in
+#define PROB_RANGE \
+ (float)0.20 // probability threshold for noise state in
// speech/noise likelihood
#define HIST_PAR_EST 1000 // histogram size for estimation of parameters
#define GAMMA_PAUSE (float)0.05 // update for conservative noise estimate
diff --git a/modules/audio_processing/residual_echo_detector_unittest.cc b/modules/audio_processing/residual_echo_detector_unittest.cc
index 6658999..84065cd 100644
--- a/modules/audio_processing/residual_echo_detector_unittest.cc
+++ b/modules/audio_processing/residual_echo_detector_unittest.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/residual_echo_detector.h"
+
#include <vector>
-#include "modules/audio_processing/residual_echo_detector.h"
#include "rtc_base/ref_counted_object.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/rms_level_unittest.cc b/modules/audio_processing/rms_level_unittest.cc
index 67489de..a1ceaad 100644
--- a/modules/audio_processing/rms_level_unittest.cc
+++ b/modules/audio_processing/rms_level_unittest.cc
@@ -9,12 +9,13 @@
*/
// MSVC++ requires this to be set before any other includes to get M_PI.
#define _USE_MATH_DEFINES
+#include "modules/audio_processing/rms_level.h"
+
#include <cmath>
#include <memory>
#include <vector>
#include "api/array_view.h"
-#include "modules/audio_processing/rms_level.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/splitting_filter_unittest.cc b/modules/audio_processing/splitting_filter_unittest.cc
index 1caee64..40f0c82 100644
--- a/modules/audio_processing/splitting_filter_unittest.cc
+++ b/modules/audio_processing/splitting_filter_unittest.cc
@@ -11,10 +11,11 @@
// MSVC++ requires this to be set before any other includes to get M_PI.
#define _USE_MATH_DEFINES
+#include "modules/audio_processing/splitting_filter.h"
+
#include <cmath>
#include "common_audio/channel_buffer.h"
-#include "modules/audio_processing/splitting_filter.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/test/aec_dump_based_simulator.h b/modules/audio_processing/test/aec_dump_based_simulator.h
index f15aa27..1181979 100644
--- a/modules/audio_processing/test/aec_dump_based_simulator.h
+++ b/modules/audio_processing/test/aec_dump_based_simulator.h
@@ -15,7 +15,6 @@
#include <string>
#include "modules/audio_processing/test/audio_processing_simulator.h"
-
#include "rtc_base/constructor_magic.h"
#include "rtc_base/ignore_wundef.h"
diff --git a/modules/audio_processing/test/audio_buffer_tools.h b/modules/audio_processing/test/audio_buffer_tools.h
index dc53e4f..9ee34e7 100644
--- a/modules/audio_processing/test/audio_buffer_tools.h
+++ b/modules/audio_processing/test/audio_buffer_tools.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
#include <vector>
+
#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
diff --git a/modules/audio_processing/test/bitexactness_tools.cc b/modules/audio_processing/test/bitexactness_tools.cc
index 7bf2b01..f245c2c 100644
--- a/modules/audio_processing/test/bitexactness_tools.cc
+++ b/modules/audio_processing/test/bitexactness_tools.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/test/bitexactness_tools.h"
#include <math.h>
+
#include <algorithm>
#include <string>
#include <vector>
diff --git a/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/modules/audio_processing/test/conversational_speech/generator_unittest.cc
index cad2656..c7a459c 100644
--- a/modules/audio_processing/test/conversational_speech/generator_unittest.cc
+++ b/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -37,6 +37,7 @@
#define _USE_MATH_DEFINES
#include <stdio.h>
+
#include <cmath>
#include <map>
#include <memory>
@@ -60,9 +61,9 @@
namespace {
using conversational_speech::LoadTiming;
-using conversational_speech::SaveTiming;
using conversational_speech::MockWavReaderFactory;
using conversational_speech::MultiEndCall;
+using conversational_speech::SaveTiming;
using conversational_speech::Turn;
using conversational_speech::WavReaderFactory;
@@ -81,12 +82,12 @@
constexpr int kDefaultSampleRate = 48000;
const std::map<std::string, const MockWavReaderFactory::Params>
kDefaultMockWavReaderFactoryParamsMap = {
- {"t300", {kDefaultSampleRate, 1u, 14400u}}, // Mono, 0.3 seconds.
- {"t500", {kDefaultSampleRate, 1u, 24000u}}, // Mono, 0.5 seconds.
- {"t1000", {kDefaultSampleRate, 1u, 48000u}}, // Mono, 1.0 seconds.
- {"sr8000", {8000, 1u, 8000u}}, // 8kHz sample rate, mono, 1 second.
- {"sr16000", {16000, 1u, 16000u}}, // 16kHz sample rate, mono, 1 second.
- {"sr16000_stereo", {16000, 2u, 16000u}}, // Like sr16000, but stereo.
+ {"t300", {kDefaultSampleRate, 1u, 14400u}}, // Mono, 0.3 seconds.
+ {"t500", {kDefaultSampleRate, 1u, 24000u}}, // Mono, 0.5 seconds.
+ {"t1000", {kDefaultSampleRate, 1u, 48000u}}, // Mono, 1.0 seconds.
+ {"sr8000", {8000, 1u, 8000u}}, // 8kHz sample rate, mono, 1 second.
+ {"sr16000", {16000, 1u, 16000u}}, // 16kHz sample rate, mono, 1 second.
+ {"sr16000_stereo", {16000, 2u, 16000u}}, // Like sr16000, but stereo.
};
const MockWavReaderFactory::Params& kDefaultMockWavReaderFactoryParams =
kDefaultMockWavReaderFactoryParamsMap.at("t500");
@@ -105,8 +106,8 @@
std::vector<int16_t> samples(params.num_samples);
for (std::size_t i = 0; i < params.num_samples; ++i) {
// TODO(alessiob): the produced tone is not pure, improve.
- samples[i] = std::lround(32767.0f * std::sin(
- two_pi * i * frequency / params.sample_rate));
+ samples[i] = std::lround(
+ 32767.0f * std::sin(two_pi * i * frequency / params.sample_rate));
}
// Write samples.
@@ -131,8 +132,7 @@
// Create sine tracks.
for (const auto& it : sine_tracks_params) {
const std::string temp_filepath = JoinFilename(temp_directory, it.first);
- CreateSineWavFile(
- temp_filepath, it.second.params, it.second.frequency);
+ CreateSineWavFile(temp_filepath, it.second.params, it.second.frequency);
}
return temp_directory;
@@ -148,7 +148,9 @@
}
void DeleteFolderAndContents(const std::string& dir) {
- if (!DirExists(dir)) { return; }
+ if (!DirExists(dir)) {
+ return;
+ }
absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
EXPECT_TRUE(dir_content);
for (const auto& path : *dir_content) {
@@ -170,8 +172,8 @@
using ::testing::_;
TEST(ConversationalSpeechTest, Settings) {
- const conversational_speech::Config config(
- audiotracks_path, timing_filepath, output_path);
+ const conversational_speech::Config config(audiotracks_path, timing_filepath,
+ output_path);
// Test getters.
EXPECT_EQ(audiotracks_path, config.audiotracks_path());
@@ -181,8 +183,8 @@
TEST(ConversationalSpeechTest, TimingSaveLoad) {
// Save test timing.
- const std::string temporary_filepath = TempFilename(
- OutputPath(), "TempTimingTestFile");
+ const std::string temporary_filepath =
+ TempFilename(OutputPath(), "TempTimingTestFile");
SaveTiming(temporary_filepath, expected_timing);
// Create a std::vector<Turn> instance by loading from file.
@@ -218,50 +220,54 @@
TEST(ConversationalSpeechTest, MultiEndCallSetupDifferentSampleRates) {
const std::vector<Turn> timing = {
- {"A", "sr8000", 0, 0}, {"B", "sr16000", 0, 0},
+ {"A", "sr8000", 0, 0},
+ {"B", "sr16000", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
// There are two unique audio tracks to read.
EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(2);
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
+ MultiEndCall multiend_call(timing, audiotracks_path,
+ std::move(mock_wavreader_factory));
EXPECT_FALSE(multiend_call.valid());
}
TEST(ConversationalSpeechTest, MultiEndCallSetupMultipleChannels) {
const std::vector<Turn> timing = {
- {"A", "sr16000_stereo", 0, 0}, {"B", "sr16000_stereo", 0, 0},
+ {"A", "sr16000_stereo", 0, 0},
+ {"B", "sr16000_stereo", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
// There is one unique audio track to read.
EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(1);
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
+ MultiEndCall multiend_call(timing, audiotracks_path,
+ std::move(mock_wavreader_factory));
EXPECT_FALSE(multiend_call.valid());
}
TEST(ConversationalSpeechTest,
- MultiEndCallSetupDifferentSampleRatesAndMultipleNumChannels) {
+ MultiEndCallSetupDifferentSampleRatesAndMultipleNumChannels) {
const std::vector<Turn> timing = {
- {"A", "sr8000", 0, 0}, {"B", "sr16000_stereo", 0, 0},
+ {"A", "sr8000", 0, 0},
+ {"B", "sr16000_stereo", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
// There are two unique audio tracks to read.
EXPECT_CALL(*mock_wavreader_factory, Create(::testing::_)).Times(2);
- MultiEndCall multiend_call(
- timing, audiotracks_path, std::move(mock_wavreader_factory));
+ MultiEndCall multiend_call(timing, audiotracks_path,
+ std::move(mock_wavreader_factory));
EXPECT_FALSE(multiend_call.valid());
}
TEST(ConversationalSpeechTest, MultiEndCallSetupFirstOffsetNegative) {
const std::vector<Turn> timing = {
- {"A", "t500", -100, 0}, {"B", "t500", 0, 0},
+ {"A", "t500", -100, 0},
+ {"B", "t500", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -279,7 +285,8 @@
// B .....1****
constexpr std::size_t expected_duration = kDefaultSampleRate;
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"B", "t500", 0, 0},
+ {"A", "t500", 0, 0},
+ {"B", "t500", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -303,7 +310,8 @@
// B .......1****
constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"B", "t500", 200, 0},
+ {"A", "t500", 0, 0},
+ {"B", "t500", 200, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -327,7 +335,8 @@
// B ....1****
constexpr std::size_t expected_duration = kDefaultSampleRate * 0.9;
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"B", "t500", -100, 0},
+ {"A", "t500", 0, 0},
+ {"B", "t500", -100, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -350,7 +359,8 @@
// A ..0****
// B .1****. The n-th turn cannot start before the (n-1)-th one.
const std::vector<Turn> timing = {
- {"A", "t500", 200, 0}, {"B", "t500", -600, 0},
+ {"A", "t500", 200, 0},
+ {"B", "t500", -600, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -368,7 +378,9 @@
// B ...1*********
constexpr std::size_t expected_duration = kDefaultSampleRate * 1.3;
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"B", "t1000", -200, 0}, {"A", "t500", -800, 0},
+ {"A", "t500", 0, 0},
+ {"B", "t1000", -200, 0},
+ {"A", "t500", -800, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -393,7 +405,9 @@
// B ......2****
// ^ Turn #1 overlaps with #0 which is from the same speaker.
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"A", "t500", -200, 0}, {"B", "t500", -200, 0},
+ {"A", "t500", 0, 0},
+ {"A", "t500", -200, 0},
+ {"B", "t500", -200, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -435,7 +449,9 @@
// C .......2****
constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
const std::vector<Turn> timing = {
- {"A", "t1000", 0, 0}, {"B", "t500", -800, 0}, {"C", "t500", 0, 0},
+ {"A", "t1000", 0, 0},
+ {"B", "t500", -800, 0},
+ {"C", "t500", 0, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -461,7 +477,9 @@
// ^ Turn #2 overlaps both with #0 and #1 (cross-talk with 3+ speakers
// not permitted).
const std::vector<Turn> timing = {
- {"A", "t1000", 0, 0}, {"B", "t500", -800, 0}, {"C", "t500", -300, 0},
+ {"A", "t1000", 0, 0},
+ {"B", "t500", -800, 0},
+ {"C", "t500", -300, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -480,7 +498,9 @@
// C .......3****
constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
const std::vector<Turn> timing = {
- {"A", "t1000", 0, 0}, {"B", "t500", -900, 0}, {"C", "t500", 100, 0},
+ {"A", "t1000", 0, 0},
+ {"B", "t500", -900, 0},
+ {"C", "t500", 100, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -503,7 +523,8 @@
// A 0****
// B 1****
const std::vector<Turn> timing = {
- {"A", "t500", 0, 0}, {"B", "t500", -500, 0},
+ {"A", "t500", 0, 0},
+ {"B", "t500", -500, 0},
};
auto mock_wavreader_factory = CreateMockWavReaderFactory();
@@ -579,9 +600,8 @@
const int sample_rates[] = {8000, 11025, 16000, 22050, 32000, 44100, 48000};
for (int sample_rate : sample_rates) {
- const std::string temp_filename =
- OutputPath() + "TempSineWavFile_" +
- std::to_string(sample_rate) + ".wav";
+ const std::string temp_filename = OutputPath() + "TempSineWavFile_" +
+ std::to_string(sample_rate) + ".wav";
// Write wav file.
const std::size_t num_samples = duration_seconds * sample_rate;
@@ -590,10 +610,9 @@
// Load wav file and check if params match.
WavReaderFactory wav_reader_factory;
- MockWavReaderFactory::Params expeted_params = {
- sample_rate, 1u, num_samples};
- CheckAudioTrackParams(
- wav_reader_factory, temp_filename, expeted_params);
+ MockWavReaderFactory::Params expeted_params = {sample_rate, 1u,
+ num_samples};
+ CheckAudioTrackParams(wav_reader_factory, temp_filename, expeted_params);
// Clean up.
RemoveFile(temp_filename);
@@ -618,21 +637,21 @@
{"t5000_440.wav", {{sample_rate, 1u, sample_rate * 5}, 440.0}},
{"t5000_880.wav", {{sample_rate, 1u, sample_rate * 5}, 880.0}},
};
- const std::string audiotracks_path = CreateTemporarySineAudioTracks(
- sine_tracks_params);
+ const std::string audiotracks_path =
+ CreateTemporarySineAudioTracks(sine_tracks_params);
// Set up the multi-end call.
- auto wavreader_factory = std::unique_ptr<WavReaderFactory>(
- new WavReaderFactory());
- MultiEndCall multiend_call(
- expected_timing, audiotracks_path, std::move(wavreader_factory));
+ auto wavreader_factory =
+ std::unique_ptr<WavReaderFactory>(new WavReaderFactory());
+ MultiEndCall multiend_call(expected_timing, audiotracks_path,
+ std::move(wavreader_factory));
// Simulate the call.
std::string output_path = JoinFilename(audiotracks_path, "output");
CreateDir(output_path);
RTC_LOG(LS_VERBOSE) << "simulator output path: " << output_path;
- auto generated_audiotrak_pairs = conversational_speech::Simulate(
- multiend_call, output_path);
+ auto generated_audiotrak_pairs =
+ conversational_speech::Simulate(multiend_call, output_path);
EXPECT_EQ(2u, generated_audiotrak_pairs->size());
// Check the output.
@@ -641,10 +660,10 @@
sample_rate, 1u, sample_rate * expected_duration_seconds};
for (const auto& it : *generated_audiotrak_pairs) {
RTC_LOG(LS_VERBOSE) << "checking far/near-end for <" << it.first << ">";
- CheckAudioTrackParams(
- wav_reader_factory, it.second.near_end, expeted_params);
- CheckAudioTrackParams(
- wav_reader_factory, it.second.far_end, expeted_params);
+ CheckAudioTrackParams(wav_reader_factory, it.second.near_end,
+ expeted_params);
+ CheckAudioTrackParams(wav_reader_factory, it.second.far_end,
+ expeted_params);
}
// Clean.
diff --git a/modules/audio_processing/test/conversational_speech/multiend_call.h b/modules/audio_processing/test/conversational_speech/multiend_call.h
index 09cb00c..5b6300f 100644
--- a/modules/audio_processing/test/conversational_speech/multiend_call.h
+++ b/modules/audio_processing/test/conversational_speech/multiend_call.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
#include <stddef.h>
+
#include <map>
#include <memory>
#include <set>
diff --git a/modules/audio_processing/test/fake_recording_device_unittest.cc b/modules/audio_processing/test/fake_recording_device_unittest.cc
index a14da82..da62beb 100644
--- a/modules/audio_processing/test/fake_recording_device_unittest.cc
+++ b/modules/audio_processing/test/fake_recording_device_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/test/fake_recording_device.h"
+
#include <cmath>
#include <memory>
#include <string>
@@ -15,7 +17,6 @@
#include "absl/memory/memory.h"
#include "api/array_view.h"
-#include "modules/audio_processing/test/fake_recording_device.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/test/protobuf_utils.cc b/modules/audio_processing/test/protobuf_utils.cc
index c47f8ea..f3c97ee 100644
--- a/modules/audio_processing/test/protobuf_utils.cc
+++ b/modules/audio_processing/test/protobuf_utils.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/test/protobuf_utils.h"
+
#include "rtc_base/system/arch.h"
namespace webrtc {
diff --git a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
index a553785..9906eca 100644
--- a/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
+++ b/modules/audio_processing/test/py_quality_assessment/quality_assessment/vad.cc
@@ -6,13 +6,14 @@
// in the file PATENTS. All contributing project authors may
// be found in the AUTHORS file in the root of the source tree.
+#include "common_audio/vad/include/vad.h"
+
#include <array>
#include <fstream>
#include <memory>
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
-#include "common_audio/vad/include/vad.h"
#include "common_audio/wav_file.h"
#include "rtc_base/logging.h"
diff --git a/modules/audio_processing/test/test_utils.cc b/modules/audio_processing/test/test_utils.cc
index 9f1a469..c02bc76 100644
--- a/modules/audio_processing/test/test_utils.cc
+++ b/modules/audio_processing/test/test_utils.cc
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "modules/audio_processing/test/test_utils.h"
+
#include <utility>
-#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/arch.h"
diff --git a/modules/audio_processing/test/test_utils.h b/modules/audio_processing/test/test_utils.h
index 3637431..0dd4a40 100644
--- a/modules/audio_processing/test/test_utils.h
+++ b/modules/audio_processing/test/test_utils.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
#include <math.h>
+
#include <iterator>
#include <limits>
#include <memory>
diff --git a/modules/audio_processing/test/wav_based_simulator.cc b/modules/audio_processing/test/wav_based_simulator.cc
index 1160ba8..4b46590 100644
--- a/modules/audio_processing/test/wav_based_simulator.cc
+++ b/modules/audio_processing/test/wav_based_simulator.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/test/wav_based_simulator.h"
#include <stdio.h>
+
#include <iostream>
#include "modules/audio_processing/test/test_utils.h"
diff --git a/modules/audio_processing/test/wav_based_simulator.h b/modules/audio_processing/test/wav_based_simulator.h
index 3dfd256..991f1db 100644
--- a/modules/audio_processing/test/wav_based_simulator.h
+++ b/modules/audio_processing/test/wav_based_simulator.h
@@ -14,7 +14,6 @@
#include <vector>
#include "modules/audio_processing/test/audio_processing_simulator.h"
-
#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/modules/audio_processing/transient/file_utils_unittest.cc b/modules/audio_processing/transient/file_utils_unittest.cc
index 0bded02..1bcf6f9 100644
--- a/modules/audio_processing/transient/file_utils_unittest.cc
+++ b/modules/audio_processing/transient/file_utils_unittest.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/transient/file_utils.h"
#include <string.h>
+
#include <memory>
#include <string>
#include <vector>
diff --git a/modules/audio_processing/transient/transient_detector.cc b/modules/audio_processing/transient/transient_detector.cc
index b328a0e..f03a2ea 100644
--- a/modules/audio_processing/transient/transient_detector.cc
+++ b/modules/audio_processing/transient/transient_detector.cc
@@ -12,6 +12,7 @@
#include <float.h>
#include <string.h>
+
#include <algorithm>
#include <cmath>
@@ -161,10 +162,9 @@
return 1.f;
}
RTC_DCHECK_NE(0, reference_energy_);
- float result =
- 1.f / (1.f + std::exp(kReferenceNonLinearity *
- (kEnergyRatioThreshold -
- reference_energy / reference_energy_)));
+ float result = 1.f / (1.f + std::exp(kReferenceNonLinearity *
+ (kEnergyRatioThreshold -
+ reference_energy / reference_energy_)));
reference_energy_ =
kMemory * reference_energy_ + (1.f - kMemory) * reference_energy;
diff --git a/modules/audio_processing/transient/transient_detector.h b/modules/audio_processing/transient/transient_detector.h
index 23b88f8..5ede2e8 100644
--- a/modules/audio_processing/transient/transient_detector.h
+++ b/modules/audio_processing/transient/transient_detector.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_DETECTOR_H_
#include <stddef.h>
+
#include <deque>
#include <memory>
diff --git a/modules/audio_processing/transient/transient_suppression_test.cc b/modules/audio_processing/transient/transient_suppression_test.cc
index 57bddb6..85db391 100644
--- a/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/modules/audio_processing/transient/transient_suppression_test.cc
@@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/transient/transient_suppressor.h"
-
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
@@ -19,6 +17,7 @@
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc/agc.h"
+#include "modules/audio_processing/transient/transient_suppressor.h"
#include "rtc_base/flags.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
diff --git a/modules/audio_processing/transient/transient_suppressor.cc b/modules/audio_processing/transient/transient_suppressor.cc
index 58d0df0..2463efa 100644
--- a/modules/audio_processing/transient/transient_suppressor.cc
+++ b/modules/audio_processing/transient/transient_suppressor.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/transient/transient_suppressor.h"
#include <string.h>
+
#include <cmath>
#include <complex>
#include <deque>
@@ -351,8 +352,7 @@
// If a restoration takes place, the |magnitudes_| are updated to the new value.
void TransientSuppressor::HardRestoration(float* spectral_mean) {
const float detector_result =
- 1.f -
- std::pow(1.f - detector_smoothed_, using_reference_ ? 200.f : 50.f);
+ 1.f - std::pow(1.f - detector_smoothed_, using_reference_ ? 200.f : 50.f);
// To restore, we get the peaks in the spectrum. If higher than the previous
// spectral mean we adjust them.
for (size_t i = 0; i < complex_analysis_length_; ++i) {
diff --git a/modules/audio_processing/transient/transient_suppressor.h b/modules/audio_processing/transient/transient_suppressor.h
index ae51966..2322b8f 100644
--- a/modules/audio_processing/transient/transient_suppressor.h
+++ b/modules/audio_processing/transient/transient_suppressor.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include "rtc_base/gtest_prod_util.h"
diff --git a/modules/audio_processing/transient/wpd_tree.h b/modules/audio_processing/transient/wpd_tree.h
index b62135d..c54220f7 100644
--- a/modules/audio_processing/transient/wpd_tree.h
+++ b/modules/audio_processing/transient/wpd_tree.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_TRANSIENT_WPD_TREE_H_
#include <stddef.h>
+
#include <memory>
#include "modules/audio_processing/transient/wpd_node.h"
diff --git a/modules/audio_processing/utility/block_mean_calculator_unittest.cc b/modules/audio_processing/utility/block_mean_calculator_unittest.cc
index 1f4ebf1..e829f69 100644
--- a/modules/audio_processing/utility/block_mean_calculator_unittest.cc
+++ b/modules/audio_processing/utility/block_mean_calculator_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/utility/block_mean_calculator.h"
+
#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_processing/utility/delay_estimator.cc b/modules/audio_processing/utility/delay_estimator.cc
index a15b914..fe750f5 100644
--- a/modules/audio_processing/utility/delay_estimator.cc
+++ b/modules/audio_processing/utility/delay_estimator.cc
@@ -12,6 +12,7 @@
#include <stdlib.h>
#include <string.h>
+
#include <algorithm>
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/utility/delay_estimator_unittest.cc b/modules/audio_processing/utility/delay_estimator_unittest.cc
index 324bc37..d3463aa 100644
--- a/modules/audio_processing/utility/delay_estimator_unittest.cc
+++ b/modules/audio_processing/utility/delay_estimator_unittest.cc
@@ -9,6 +9,7 @@
*/
#include "modules/audio_processing/utility/delay_estimator.h"
+
#include "modules/audio_processing/utility/delay_estimator_internal.h"
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "test/gtest.h"
diff --git a/modules/audio_processing/utility/ooura_fft_mips.cc b/modules/audio_processing/utility/ooura_fft_mips.cc
index 9fe577d..42b9d3a 100644
--- a/modules/audio_processing/utility/ooura_fft_mips.cc
+++ b/modules/audio_processing/utility/ooura_fft_mips.cc
@@ -9,7 +9,6 @@
*/
#include "modules/audio_processing/utility/ooura_fft.h"
-
#include "modules/audio_processing/utility/ooura_fft_tables_common.h"
namespace webrtc {
diff --git a/modules/audio_processing/utility/ooura_fft_neon.cc b/modules/audio_processing/utility/ooura_fft_neon.cc
index 401387a..95b5f09 100644
--- a/modules/audio_processing/utility/ooura_fft_neon.cc
+++ b/modules/audio_processing/utility/ooura_fft_neon.cc
@@ -14,10 +14,9 @@
* Based on the sse2 version.
*/
-#include "modules/audio_processing/utility/ooura_fft.h"
-
#include <arm_neon.h>
+#include "modules/audio_processing/utility/ooura_fft.h"
#include "modules/audio_processing/utility/ooura_fft_tables_common.h"
#include "modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h"
diff --git a/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h b/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
index b6e4a07..10aebac 100644
--- a/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
+++ b/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
@@ -86,7 +86,10 @@
0.956940353f, -0.956940353f,
};
ALIGN16_BEG const float ALIGN16_END cftmdl_wk1r[4] = {
- 0.707106769f, 0.707106769f, 0.707106769f, -0.707106769f,
+ 0.707106769f,
+ 0.707106769f,
+ 0.707106769f,
+ -0.707106769f,
};
#endif
diff --git a/modules/audio_processing/vad/pole_zero_filter.cc b/modules/audio_processing/vad/pole_zero_filter.cc
index 4156d7e..e7a6113 100644
--- a/modules/audio_processing/vad/pole_zero_filter.cc
+++ b/modules/audio_processing/vad/pole_zero_filter.cc
@@ -11,6 +11,7 @@
#include "modules/audio_processing/vad/pole_zero_filter.h"
#include <string.h>
+
#include <algorithm>
namespace webrtc {
diff --git a/modules/audio_processing/vad/vad_audio_proc.h b/modules/audio_processing/vad/vad_audio_proc.h
index 9be3467..4a71ce3 100644
--- a/modules/audio_processing/vad/vad_audio_proc.h
+++ b/modules/audio_processing/vad/vad_audio_proc.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include "modules/audio_processing/vad/common.h" // AudioFeatures, kSampleR...
diff --git a/modules/audio_processing/vad/voice_activity_detector.h b/modules/audio_processing/vad/voice_activity_detector.h
index d140fe2..a19883d 100644
--- a/modules/audio_processing/vad/voice_activity_detector.h
+++ b/modules/audio_processing/vad/voice_activity_detector.h
@@ -13,6 +13,7 @@
#include <stddef.h>
#include <stdint.h>
+
#include <memory>
#include <vector>
diff --git a/modules/audio_processing/voice_detection_impl.h b/modules/audio_processing/voice_detection_impl.h
index 6800566..4007f67 100644
--- a/modules/audio_processing/voice_detection_impl.h
+++ b/modules/audio_processing/voice_detection_impl.h
@@ -12,6 +12,7 @@
#define MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#include <stddef.h>
+
#include <memory>
#include "modules/audio_processing/include/audio_processing.h"
