Include files from webrtc/.. paths in audio_processing/
BUG=1662
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4116 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_processing/aec/aec_core_sse2.c b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
index be170da..fdc6872 100644
--- a/webrtc/modules/audio_processing/aec/aec_core_sse2.c
+++ b/webrtc/modules/audio_processing/aec/aec_core_sse2.c
@@ -413,4 +413,3 @@
WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
}
-
diff --git a/webrtc/modules/audio_processing/aec/aec_rdft.c b/webrtc/modules/audio_processing/aec/aec_rdft.c
index d4254dd..e63f367 100644
--- a/webrtc/modules/audio_processing/aec/aec_rdft.c
+++ b/webrtc/modules/audio_processing/aec/aec_rdft.c
@@ -19,12 +19,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "aec_rdft.h"
+#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
#include <math.h>
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-#include "typedefs.h"
+#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/typedefs.h"
// constants shared by all paths (C, SSE2).
float rdft_w[64];
diff --git a/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c b/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c
index eeb3152..49a4072 100644
--- a/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c
+++ b/webrtc/modules/audio_processing/aec/aec_rdft_sse2.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "aec_rdft.h"
+#include "webrtc/modules/audio_processing/aec/aec_rdft.h"
#include <emmintrin.h>
@@ -424,4 +424,3 @@
rftfsub_128 = rftfsub_128_SSE2;
rftbsub_128 = rftbsub_128_SSE2;
}
-
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.c b/webrtc/modules/audio_processing/aec/aec_resampler.c
index 126a209..13521ec 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.c
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.c
@@ -12,14 +12,14 @@
* skew by resampling the farend signal.
*/
-#include "aec_resampler.h"
+#include "webrtc/modules/audio_processing/aec/aec_resampler.h"
#include <assert.h>
+#include <math.h>
#include <stdlib.h>
#include <string.h>
-#include <math.h>
-#include "aec_core.h"
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
enum { kEstimateLengthFrames = 400 };
diff --git a/webrtc/modules/audio_processing/aec/aec_resampler.h b/webrtc/modules/audio_processing/aec/aec_resampler.h
index acf8cce..3cd0691 100644
--- a/webrtc/modules/audio_processing/aec/aec_resampler.h
+++ b/webrtc/modules/audio_processing/aec/aec_resampler.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
-#include "aec_core.h"
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
enum { kResamplingDelay = 1 };
enum { kResamplerBufferSize = FRAME_LEN * 4 };
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core.c b/webrtc/modules/audio_processing/aecm/aecm_core.c
index d3f3cd4..e4fe349 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core.c
@@ -15,8 +15,8 @@
#include <stdlib.h>
#include "webrtc/common_audio/signal_processing/include/real_fft.h"
-#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
+#include "webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h"
#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include "webrtc/system_wrappers/interface/compile_assert.h"
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
@@ -2051,5 +2051,3 @@
sizeof(int16_t) * readLen);
aecm->farBufReadPos += readLen;
}
-
-
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_neon.S b/webrtc/modules/audio_processing/aecm/aecm_core_neon.S
index b47cd28..4e28873 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_neon.S
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_neon.S
@@ -12,8 +12,8 @@
@ This file contains some functions in AECM, optimized for ARM Neon
@ platforms. Reference C code is in file aecm_core.c. Bit-exact.
-#include "aecm_defines.h"
#include "aecm_core_neon_offsets.h"
+#include "webrtc/modules/audio_processing/aecm/aecm_defines.h"
#include "webrtc/system_wrappers/interface/asm_defines.h"
GLOBAL_LABEL WebRtcAecm_kSqrtHanning
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_neon.c b/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
index c6910f0..484ad71 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_neon.c
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "aecm_core.h"
+#include "webrtc/modules/audio_processing/aecm/aecm_core.h"
#include <arm_neon.h>
#include <assert.h>
-#include "common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"
// TODO(kma): Re-write the corresponding assembly file, the offset
// generating script and makefile, to replace these C functions.
diff --git a/webrtc/modules/audio_processing/aecm/aecm_core_neon_offsets.c b/webrtc/modules/audio_processing/aecm/aecm_core_neon_offsets.c
index b7bd48d..2c302e6 100644
--- a/webrtc/modules/audio_processing/aecm/aecm_core_neon_offsets.c
+++ b/webrtc/modules/audio_processing/aecm/aecm_core_neon_offsets.c
@@ -9,7 +9,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "aecm_core.h"
+#include "webrtc/modules/audio_processing/aecm/aecm_core.h"
#include <stddef.h>
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/analog_agc.c
index 0965def..1e8e3d8 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -22,7 +22,7 @@
#ifdef AGC_DEBUG //test log
#include <stdio.h>
#endif
-#include "analog_agc.h"
+#include "webrtc/modules/audio_processing/agc/analog_agc.h"
/* The slope of in Q13*/
static const int16_t kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.h b/webrtc/modules/audio_processing/agc/analog_agc.h
index ce005fc..16ea29c 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.h
+++ b/webrtc/modules/audio_processing/agc/analog_agc.h
@@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-#include "typedefs.h"
-#include "gain_control.h"
-#include "digital_agc.h"
+#include "webrtc/modules/audio_processing/agc/digital_agc.h"
+#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
+#include "webrtc/typedefs.h"
//#define AGC_DEBUG
//#define MIC_LEVEL_FEEDBACK
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index 6046351..00565dd 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -12,7 +12,7 @@
*
*/
-#include "digital_agc.h"
+#include "webrtc/modules/audio_processing/agc/digital_agc.h"
#include <assert.h>
#include <string.h>
@@ -20,7 +20,7 @@
#include <stdio.h>
#endif
-#include "gain_control.h"
+#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
// To generate the gaintable, copy&paste the following lines to a Matlab window:
// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.h b/webrtc/modules/audio_processing/agc/digital_agc.h
index 573f3ec..6bd086f 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.h
+++ b/webrtc/modules/audio_processing/agc/digital_agc.h
@@ -14,8 +14,8 @@
#ifdef AGC_DEBUG
#include <stdio.h>
#endif
-#include "typedefs.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"
// the 32 most significant bits of A(19) * B(26) >> 13
#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
diff --git a/webrtc/modules/audio_processing/agc/include/gain_control.h b/webrtc/modules/audio_processing/agc/include/gain_control.h
index 1ed06c0..b6a1889 100644
--- a/webrtc/modules/audio_processing/agc/include/gain_control.h
+++ b/webrtc/modules/audio_processing/agc/include/gain_control.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
// Errors
#define AGC_UNSPECIFIED_ERROR 18000
diff --git a/webrtc/modules/audio_processing/audio_buffer.cc b/webrtc/modules/audio_processing/audio_buffer.cc
index db1f2eb..048d048 100644
--- a/webrtc/modules/audio_processing/audio_buffer.cc
+++ b/webrtc/modules/audio_processing/audio_buffer.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
namespace {
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 87d6972..2638bef 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "typedefs.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 1a3c6ea..4be5f0f 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -11,12 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-#include "audio_processing.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include <list>
#include <string>
-#include "scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class AudioBuffer;
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
index 04adefe..af8f907 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
@@ -14,8 +14,8 @@
#include <cstring>
#include "webrtc/modules/audio_processing/aecm/include/echo_control_mobile.h"
-#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
diff --git a/webrtc/modules/audio_processing/echo_control_mobile_impl.h b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
index 6d9e369..9b69cbc 100644
--- a/webrtc/modules/audio_processing/echo_control_mobile_impl.h
+++ b/webrtc/modules/audio_processing/echo_control_mobile_impl.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/gain_control_impl.cc b/webrtc/modules/audio_processing/gain_control_impl.cc
index 01a372a..e910e41 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.cc
+++ b/webrtc/modules/audio_processing/gain_control_impl.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "gain_control_impl.h"
+#include "webrtc/modules/audio_processing/gain_control_impl.h"
#include <cassert>
-#include "critical_section_wrapper.h"
-#include "gain_control.h"
+#include "webrtc/modules/audio_processing/agc/include/gain_control.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index 5915eeb..4c58f22 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -13,8 +13,8 @@
#include <vector>
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.cc b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
index c4bfa83..82e87c8 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.cc
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "high_pass_filter_impl.h"
+#include "webrtc/modules/audio_processing/high_pass_filter_impl.h"
#include <cassert>
-#include "critical_section_wrapper.h"
-#include "typedefs.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/typedefs.h"
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
namespace webrtc {
namespace {
diff --git a/webrtc/modules/audio_processing/high_pass_filter_impl.h b/webrtc/modules/audio_processing/high_pass_filter_impl.h
index 94a9c89..1531df9 100644
--- a/webrtc/modules/audio_processing/high_pass_filter_impl.h
+++ b/webrtc/modules/audio_processing/high_pass_filter_impl.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/level_estimator_impl.cc b/webrtc/modules/audio_processing/level_estimator_impl.cc
index 42cac99..29dbdfc 100644
--- a/webrtc/modules/audio_processing/level_estimator_impl.cc
+++ b/webrtc/modules/audio_processing/level_estimator_impl.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "level_estimator_impl.h"
+#include "webrtc/modules/audio_processing/level_estimator_impl.h"
#include <assert.h>
#include <math.h>
#include <string.h>
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-#include "critical_section_wrapper.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc {
namespace {
diff --git a/webrtc/modules/audio_processing/level_estimator_impl.h b/webrtc/modules/audio_processing/level_estimator_impl.h
index 1a06343..11191fa 100644
--- a/webrtc/modules/audio_processing/level_estimator_impl.h
+++ b/webrtc/modules/audio_processing/level_estimator_impl.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index d6162e6..0314ceb 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "noise_suppression_impl.h"
+#include "webrtc/modules/audio_processing/noise_suppression_impl.h"
#include <cassert>
-#include "critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#if defined(WEBRTC_NS_FLOAT)
-#include "noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
#elif defined(WEBRTC_NS_FIXED)
-#include "noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
#endif
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
namespace webrtc {
@@ -176,4 +176,3 @@
return apm_->kUnspecifiedError;
}
} // namespace webrtc
-
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.h b/webrtc/modules/audio_processing/noise_suppression_impl.h
index 73a2322..198cfc7 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.h
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression.h b/webrtc/modules/audio_processing/ns/include/noise_suppression.h
index c5cee9c..32b1803 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression.h
+++ b/webrtc/modules/audio_processing/ns/include/noise_suppression.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
typedef struct NsHandleT NsHandle;
diff --git a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h b/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
index 0ce89ba..e775868 100644
--- a/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
+++ b/webrtc/modules/audio_processing/ns/include/noise_suppression_x.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
typedef struct NsxHandleT NsxHandle;
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression.c b/webrtc/modules/audio_processing/ns/noise_suppression.c
index c0345c5..848467f 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression.c
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
#include <stdlib.h>
#include <string.h>
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "defines.h"
-#include "ns_core.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/ns/defines.h"
+#include "webrtc/modules/audio_processing/ns/ns_core.h"
int WebRtcNs_Create(NsHandle** NS_inst) {
*NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
diff --git a/webrtc/modules/audio_processing/ns/noise_suppression_x.c b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
index 20a296e..ef4bbe1 100644
--- a/webrtc/modules/audio_processing/ns/noise_suppression_x.c
+++ b/webrtc/modules/audio_processing/ns/noise_suppression_x.c
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
#include <stdlib.h>
-#include "common_audio/signal_processing/include/real_fft.h"
-#include "nsx_core.h"
-#include "nsx_defines.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/modules/audio_processing/ns/nsx_core.h"
+#include "webrtc/modules/audio_processing/ns/nsx_defines.h"
int WebRtcNsx_Create(NsxHandle** nsxInst) {
NsxInst_t* self = malloc(sizeof(NsxInst_t));
@@ -51,4 +51,3 @@
return WebRtcNsx_ProcessCore(
(NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame, outFrameHB);
}
-
diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
index 064477a..e7093c7 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/webrtc/modules/audio_processing/ns/ns_core.c
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <string.h>
#include <math.h>
+#include <string.h>
//#include <stdio.h>
#include <stdlib.h>
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "windows_private.h"
-#include "fft4g.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression.h"
+#include "webrtc/modules/audio_processing/ns/ns_core.h"
+#include "webrtc/modules/audio_processing/ns/windows_private.h"
+#include "webrtc/modules/audio_processing/utility/fft4g.h"
// Set Feature Extraction Parameters
void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) {
diff --git a/webrtc/modules/audio_processing/ns/ns_core.h b/webrtc/modules/audio_processing/ns/ns_core.h
index e98bfbe..50daa13 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.h
+++ b/webrtc/modules/audio_processing/ns/ns_core.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-#include "defines.h"
+#include "webrtc/modules/audio_processing/ns/defines.h"
typedef struct NSParaExtract_t_ {
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 50f2a18..bf86c72 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "noise_suppression_x.h"
+#include "webrtc/modules/audio_processing/ns/include/noise_suppression_x.h"
#include <assert.h>
#include <math.h>
-#include <string.h>
-#include <stdlib.h>
#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
-#include "common_audio/signal_processing/include/real_fft.h"
-#include "cpu_features_wrapper.h"
-#include "nsx_core.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/modules/audio_processing/ns/nsx_core.h"
+#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#if (defined WEBRTC_DETECT_ARM_NEON || defined WEBRTC_ARCH_ARM_NEON)
/* Tables are defined in ARM assembly files. */
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_neon.S b/webrtc/modules/audio_processing/ns/nsx_core_neon.S
index 24e3089..a0d4a2c 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_neon.S
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon.S
@@ -12,9 +12,9 @@
@ This file contains some functions in NS, optimized for ARM Neon
@ platforms. Reference C code is in file nsx_core.c. Bit-exact.
-#include "webrtc/system_wrappers/interface/asm_defines.h"
-#include "nsx_defines.h"
#include "nsx_core_neon_offsets.h"
+#include "webrtc/modules/audio_processing/ns/nsx_defines.h"
+#include "webrtc/system_wrappers/interface/asm_defines.h"
GLOBAL_FUNCTION WebRtcNsx_NoiseEstimationNeon
GLOBAL_FUNCTION WebRtcNsx_PrepareSpectrumNeon
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_neon.c b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
index 5e20a80..2cb2e4e 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_neon.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "nsx_core.h"
+#include "webrtc/modules/audio_processing/ns/nsx_core.h"
#include <arm_neon.h>
#include <assert.h>
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_neon_offsets.c b/webrtc/modules/audio_processing/ns/nsx_core_neon_offsets.c
index ee64a59..1ddcbe2 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_neon_offsets.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_neon_offsets.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "nsx_core.h"
+#include "webrtc/modules/audio_processing/ns/nsx_core.h"
#include <stddef.h>
diff --git a/webrtc/modules/audio_processing/processing_component.cc b/webrtc/modules/audio_processing/processing_component.cc
index 9ac1257..37721c0 100644
--- a/webrtc/modules/audio_processing/processing_component.cc
+++ b/webrtc/modules/audio_processing/processing_component.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
#include <cassert>
-#include "audio_processing_impl.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/processing_component.h b/webrtc/modules/audio_processing/processing_component.h
index b3457b5..f1f367e 100644
--- a/webrtc/modules/audio_processing/processing_component.h
+++ b/webrtc/modules/audio_processing/processing_component.h
@@ -13,7 +13,7 @@
#include <vector>
-#include "audio_processing.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class AudioProcessingImpl;
diff --git a/webrtc/modules/audio_processing/splitting_filter.cc b/webrtc/modules/audio_processing/splitting_filter.cc
index 448a454..372c8dc 100644
--- a/webrtc/modules/audio_processing/splitting_filter.cc
+++ b/webrtc/modules/audio_processing/splitting_filter.cc
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "splitting_filter.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/splitting_filter.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/splitting_filter.h b/webrtc/modules/audio_processing/splitting_filter.h
index 1655726..b6c8512 100644
--- a/webrtc/modules/audio_processing/splitting_filter.h
+++ b/webrtc/modules/audio_processing/splitting_filter.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
-#include "typedefs.h"
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"
namespace webrtc {
/*
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 80b697e..bdd2a29 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -17,7 +17,7 @@
#include <algorithm>
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/interface/module_common_types.h"
diff --git a/webrtc/modules/audio_processing/test/unit_test.cc b/webrtc/modules/audio_processing/test/unit_test.cc
index bc7c144..54c43f3 100644
--- a/webrtc/modules/audio_processing/test/unit_test.cc
+++ b/webrtc/modules/audio_processing/test/unit_test.cc
@@ -13,7 +13,7 @@
#include <algorithm>
#include <queue>
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
diff --git a/webrtc/modules/audio_processing/test/unpack.cc b/webrtc/modules/audio_processing/test/unpack.cc
index 2337131..0740143 100644
--- a/webrtc/modules/audio_processing/test/unpack.cc
+++ b/webrtc/modules/audio_processing/test/unpack.cc
@@ -15,10 +15,10 @@
#include <stdio.h>
-#include "google/gflags.h"
-#include "scoped_ptr.h"
-#include "typedefs.h"
+#include "gflags/gflags.h"
#include "webrtc/audio_processing/debug.pb.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/typedefs.h"
using webrtc::scoped_array;
diff --git a/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc b/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
index ac3bc16..f4b4711 100644
--- a/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
+++ b/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
extern "C" {
#include "webrtc/modules/audio_processing/utility/delay_estimator.h"
diff --git a/webrtc/modules/audio_processing/utility/ring_buffer.c b/webrtc/modules/audio_processing/utility/ring_buffer.c
index 617c30f..f65c6fb 100644
--- a/webrtc/modules/audio_processing/utility/ring_buffer.c
+++ b/webrtc/modules/audio_processing/utility/ring_buffer.c
@@ -11,7 +11,7 @@
// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
// otherwise specified, functions return 0 on success and -1 on error.
-#include "ring_buffer.h"
+#include "webrtc/modules/audio_processing/utility/ring_buffer.h"
#include <stddef.h> // size_t
#include <stdlib.h>
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.cc b/webrtc/modules/audio_processing/voice_detection_impl.cc
index 8a505ef..f29767c 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_detection_impl.h"
+#include "webrtc/modules/audio_processing/voice_detection_impl.h"
#include <cassert>
-#include "critical_section_wrapper.h"
-#include "webrtc_vad.h"
+#include "webrtc/common_audio/vad/include/webrtc_vad.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/audio_processing_impl.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.h b/webrtc/modules/audio_processing/voice_detection_impl.h
index 52d92e0..f52f503 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.h
+++ b/webrtc/modules/audio_processing/voice_detection_impl.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
-#include "audio_processing.h"
-#include "processing_component.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/processing_component.h"
namespace webrtc {
class AudioProcessingImpl;