Include files from webrtc/.. paths in common_audio/

BUG=1662 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1535005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4107 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-27 09:49:58 +00:00 · 2013-05-27 09:49:58 +00:00 · aa30bb7ef5
commit aa30bb7ef5
parent 0afd84067a
63 changed files with 108 additions and 115 deletions
--- a/webrtc/common_audio/resampler/include/resampler.h
+++ b/webrtc/common_audio/resampler/include/resampler.h
@ -16,7 +16,7 @@
 #ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
 #define WEBRTC_RESAMPLER_RESAMPLER_H_

-#include "typedefs.h"
+#include "webrtc/typedefs.h"

 namespace webrtc
 {
--- a/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
+++ b/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
@ -230,4 +230,3 @@ INSTANTIATE_TEST_CASE_P(
        std::tr1::make_tuple(192000, 32000, -21.02, -10.94)));

 }  // namespace webrtc
-
--- a/webrtc/common_audio/resampler/resampler.cc
+++ b/webrtc/common_audio/resampler/resampler.cc
@ -16,8 +16,8 @@
 #include <stdlib.h>
 #include <string.h>

-#include "signal_processing_library.h"
-#include "resampler.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"


 namespace webrtc
--- a/webrtc/common_audio/resampler/resampler_unittest.cc
+++ b/webrtc/common_audio/resampler/resampler_unittest.cc
@ -8,9 +8,9 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"

-#include "common_audio/resampler/include/resampler.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"

 // TODO(andrew): this is a work-in-progress. Many more tests are needed.

--- a/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
+++ b/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
@ -367,4 +367,3 @@ INSTANTIATE_TEST_CASE_P(
        std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52)));

 }  // namespace webrtc
-
--- a/webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
+++ b/webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_AutoCorrToReflCoef(const int32_t *R, int use_order, int16_t *K)
 {
--- a/webrtc/common_audio/signal_processing/auto_correlation.c
+++ b/webrtc/common_audio/signal_processing/auto_correlation.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
                              int in_vector_length,
--- a/webrtc/common_audio/signal_processing/complex_bit_reverse.c
+++ b/webrtc/common_audio/signal_processing/complex_bit_reverse.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 /* Tables for data buffer indexes that are bit reversed and thus need to be
 * swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
@ -106,4 +106,3 @@ void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
    }
  }
 }
-
--- a/webrtc/common_audio/signal_processing/complex_fft.c
+++ b/webrtc/common_audio/signal_processing/complex_fft.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 #define CFFTSFT 14
 #define CFFTRND 1
--- a/webrtc/common_audio/signal_processing/copy_set_operations.c
+++ b/webrtc/common_audio/signal_processing/copy_set_operations.c
@ -25,7 +25,7 @@
 */

 #include <string.h>
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"


 void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, int length)
--- a/webrtc/common_audio/signal_processing/cross_correlation.c
+++ b/webrtc/common_audio/signal_processing/cross_correlation.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 /* C version of WebRtcSpl_CrossCorrelation() for generic platforms. */
 void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
--- a/webrtc/common_audio/signal_processing/division_operations.c
+++ b/webrtc/common_audio/signal_processing/division_operations.c
@ -21,7 +21,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
 {
--- a/webrtc/common_audio/signal_processing/dot_product_with_scale.c
+++ b/webrtc/common_audio/signal_processing/dot_product_with_scale.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
                                      const int16_t* vector2,
--- a/webrtc/common_audio/signal_processing/downsample_fast.c
+++ b/webrtc/common_audio/signal_processing/downsample_fast.c
@ -8,7 +8,7 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // TODO(Bjornv): Change the function parameter order to WebRTC code style.
 // C version of WebRtcSpl_DownsampleFast() for generic platforms.
--- a/webrtc/common_audio/signal_processing/energy.c
+++ b/webrtc/common_audio/signal_processing/energy.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor)
 {
--- a/webrtc/common_audio/signal_processing/filter_ar.c
+++ b/webrtc/common_audio/signal_processing/filter_ar.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int WebRtcSpl_FilterAR(const int16_t* a,
                       int a_length,
--- a/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@ -9,7 +9,7 @@
 */
 #include <assert.h>

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // TODO(bjornv): Change the return type to report errors.

@ -40,4 +40,3 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
    data_out[i] = (int16_t)((output + 2048) >> 12);
  }
 }
-
--- a/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
+++ b/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_FilterMAFastQ12(int16_t* in_ptr,
                               int16_t* out_ptr,
--- a/webrtc/common_audio/signal_processing/get_hanning_window.c
+++ b/webrtc/common_audio/signal_processing/get_hanning_window.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Hanning table with 256 entries
 static const int16_t kHanningTable[] = {
--- a/webrtc/common_audio/signal_processing/get_scaling_square.c
+++ b/webrtc/common_audio/signal_processing/get_scaling_square.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int WebRtcSpl_GetScalingSquare(int16_t *in_vector, int in_vector_length, int times)
 {
--- a/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
+++ b/webrtc/common_audio/signal_processing/ilbc_specific_functions.c
@ -19,7 +19,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_ReverseOrderMultArrayElements(int16_t *out, const int16_t *in,
                                             const int16_t *win,
--- a/webrtc/common_audio/signal_processing/include/real_fft.h
+++ b/webrtc/common_audio/signal_processing/include/real_fft.h
@ -11,7 +11,7 @@
 #ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
 #define WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_

-#include "typedefs.h"
+#include "webrtc/typedefs.h"

 struct RealFFT;

--- a/webrtc/common_audio/signal_processing/include/spl_inl.h
+++ b/webrtc/common_audio/signal_processing/include/spl_inl.h
@ -16,7 +16,7 @@
 #define WEBRTC_SPL_SPL_INL_H_

 #ifdef WEBRTC_ARCH_ARM_V7
-#include "spl_inl_armv7.h"
+#include "webrtc/common_audio/signal_processing/include/spl_inl_armv7.h"
 #else

 static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
--- a/webrtc/common_audio/signal_processing/levinson_durbin.c
+++ b/webrtc/common_audio/signal_processing/levinson_durbin.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 #define SPL_LEVINSON_MAXORDER 20

--- a/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
+++ b/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 #define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50

--- a/webrtc/common_audio/signal_processing/min_max_operations.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations.c
@ -24,7 +24,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 #include <stdlib.h>

--- a/webrtc/common_audio/signal_processing/min_max_operations_mips.c
+++ b/webrtc/common_audio/signal_processing/min_max_operations_mips.c
@ -16,7 +16,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Maximum absolute value of word16 vector.
 int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length) {
@ -383,4 +383,3 @@ int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, int length) {

  return minimum;
 }
-
--- a/webrtc/common_audio/signal_processing/randomization_functions.c
+++ b/webrtc/common_audio/signal_processing/randomization_functions.c
@ -20,7 +20,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 static const int16_t kRandNTable[] = {
    9178,    -7260,       40,    10189,     4894,    -3531,   -13779,    14764,
--- a/webrtc/common_audio/signal_processing/real_fft.c
+++ b/webrtc/common_audio/signal_processing/real_fft.c
@ -8,11 +8,11 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"

 #include <stdlib.h>

-#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 struct RealFFT {
  int order;
--- a/webrtc/common_audio/signal_processing/real_fft_unittest.cc
+++ b/webrtc/common_audio/signal_processing/real_fft_unittest.cc
@ -8,11 +8,11 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "common_audio/signal_processing/include/real_fft.h"
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"

-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"

 namespace webrtc {
 namespace {
--- a/webrtc/common_audio/signal_processing/refl_coef_to_lpc.c
+++ b/webrtc/common_audio/signal_processing/refl_coef_to_lpc.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
 {
--- a/webrtc/common_audio/signal_processing/resample.c
+++ b/webrtc/common_audio/signal_processing/resample.c
@ -15,8 +15,8 @@
 *
 */

-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"

 // Declaration of internally used functions
 static void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In, int16_t *Out,
--- a/webrtc/common_audio/signal_processing/resample_48khz.c
+++ b/webrtc/common_audio/signal_processing/resample_48khz.c
@ -16,8 +16,8 @@
 */

 #include <string.h>
-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"

 ////////////////////////////
 ///// 48 kHz -> 16 kHz /////
--- a/webrtc/common_audio/signal_processing/resample_by_2.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 #ifdef WEBRTC_ARCH_ARM_V7

--- a/webrtc/common_audio/signal_processing/resample_by_2_internal.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2_internal.c
@ -14,7 +14,7 @@
 *
 */

-#include "resample_by_2_internal.h"
+#include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"

 // allpass filter coefficients.
 static const int16_t kResampleAllpass[2][3] = {
--- a/webrtc/common_audio/signal_processing/resample_by_2_internal.h
+++ b/webrtc/common_audio/signal_processing/resample_by_2_internal.h
@ -17,7 +17,7 @@
 #ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
 #define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_

-#include "typedefs.h"
+#include "webrtc/typedefs.h"

 /*******************************************************************
 * resample_by_2_fast.c
--- a/webrtc/common_audio/signal_processing/resample_by_2_mips.c
+++ b/webrtc/common_audio/signal_processing/resample_by_2_mips.c
@ -17,7 +17,7 @@

 #if defined(MIPS32_LE)

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // allpass filter coefficients.
 static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528};
@ -288,4 +288,3 @@ void WebRtcSpl_DownsampleBy2(const int16_t* in,
 }

 #endif  // #if defined(MIPS32_LE)
-
--- a/webrtc/common_audio/signal_processing/resample_fractional.c
+++ b/webrtc/common_audio/signal_processing/resample_fractional.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // interpolation coefficients
 static const int16_t kCoefficients48To32[2][8] = {
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@ -8,8 +8,8 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "signal_processing_library.h"
-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 static const int kVector16Size = 9;
 static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963,
--- a/webrtc/common_audio/signal_processing/spl_init.c
+++ b/webrtc/common_audio/signal_processing/spl_init.c
@ -14,9 +14,9 @@
 * Some code came from common/rtcd.c in the WebM project.
 */

-#include "common_audio/signal_processing/include/real_fft.h"
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/common_audio/signal_processing/include/real_fft.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"

 /* Declare function pointers. */
 MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
--- a/webrtc/common_audio/signal_processing/spl_sqrt.c
+++ b/webrtc/common_audio/signal_processing/spl_sqrt.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int32_t WebRtcSpl_SqrtLocal(int32_t in);

--- a/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
+++ b/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
@ -28,7 +28,7 @@

 // Minor modifications in code style for WebRTC, 2012.

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 /*
 * Algorithm:
--- a/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
+++ b/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
@ -108,4 +108,3 @@ DEFINE_FUNCTION WebRtcSpl_SqrtFloor

  bic    r0, r2, #3 << 30  @ for rounding add: cmp r0, r2  adc r2, #1
  bx lr
-
--- a/webrtc/common_audio/signal_processing/spl_version.c
+++ b/webrtc/common_audio/signal_processing/spl_version.c
@ -16,7 +16,7 @@
 */

 #include <string.h>
-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes)
 {
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@ -13,7 +13,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 // Number of samples in a low/high-band frame.
 enum
--- a/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
+++ b/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@ -15,7 +15,7 @@
 *
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
                                      int16_t *yQ15)
--- a/webrtc/common_audio/signal_processing/vector_scaling_operations.c
+++ b/webrtc/common_audio/signal_processing/vector_scaling_operations.c
@ -20,7 +20,7 @@
 * WebRtcSpl_ScaleAndAddVectorsWithRoundC()
 */

-#include "signal_processing_library.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

 void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length,
                                 const int16_t *in, int16_t right_shifts)
--- a/webrtc/common_audio/vad/include/webrtc_vad.h
+++ b/webrtc/common_audio/vad/include/webrtc_vad.h
@ -16,7 +16,7 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
 #define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_

-#include "typedefs.h"  // NOLINT
+#include "webrtc/typedefs.h"

 typedef struct WebRtcVadInst VadInst;

--- a/webrtc/common_audio/vad/vad_core.c
+++ b/webrtc/common_audio/vad/vad_core.c
@ -8,13 +8,13 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "vad_core.h"
+#include "webrtc/common_audio/vad/vad_core.h"

-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_filterbank.h"
-#include "vad_gmm.h"
-#include "vad_sp.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/vad/vad_filterbank.h"
+#include "webrtc/common_audio/vad/vad_gmm.h"
+#include "webrtc/common_audio/vad/vad_sp.h"
+#include "webrtc/typedefs.h"

 // Spectrum Weighting
 static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
--- a/webrtc/common_audio/vad/vad_core.h
+++ b/webrtc/common_audio/vad/vad_core.h
@ -16,8 +16,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
 #define WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_

-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"

 enum { kNumChannels = 6 };  // Number of frequency bands (named channels).
 enum { kNumGaussians = 2 };  // Number of Gaussians per channel in the GMM.
--- a/webrtc/common_audio/vad/vad_core_unittest.cc
+++ b/webrtc/common_audio/vad/vad_core_unittest.cc
@ -10,12 +10,12 @@

 #include <stdlib.h>

-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/typedefs.h"

 extern "C" {
-#include "vad_core.h"
+#include "webrtc/common_audio/vad/vad_core.h"
 }

 namespace {
--- a/webrtc/common_audio/vad/vad_filterbank.c
+++ b/webrtc/common_audio/vad/vad_filterbank.c
@ -8,12 +8,12 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "vad_filterbank.h"
+#include "webrtc/common_audio/vad/vad_filterbank.h"

 #include <assert.h>

-#include "signal_processing_library.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"

 // Constants used in LogOfEnergy().
 static const int16_t kLogConst = 24660;  // 160*log10(2) in Q9.
--- a/webrtc/common_audio/vad/vad_filterbank.h
+++ b/webrtc/common_audio/vad/vad_filterbank.h
@ -15,8 +15,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
 #define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_

-#include "typedefs.h"
-#include "vad_core.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/typedefs.h"

 // Takes |data_length| samples of |data_in| and calculates the logarithm of the
 // energy of each of the |kNumChannels| = 6 frequency bands used by the VAD:
--- a/webrtc/common_audio/vad/vad_filterbank_unittest.cc
+++ b/webrtc/common_audio/vad/vad_filterbank_unittest.cc
@ -10,13 +10,13 @@

 #include <stdlib.h>

-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/typedefs.h"

 extern "C" {
-#include "vad_core.h"
-#include "vad_filterbank.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/common_audio/vad/vad_filterbank.h"
 }

 namespace {
--- a/webrtc/common_audio/vad/vad_gmm.c
+++ b/webrtc/common_audio/vad/vad_gmm.c
@ -8,10 +8,10 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "vad_gmm.h"
+#include "webrtc/common_audio/vad/vad_gmm.h"

-#include "signal_processing_library.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/typedefs.h"

 static const int32_t kCompVar = 22005;
 static const int16_t kLog2Exp = 5909;  // log2(exp(1)) in Q12.
--- a/webrtc/common_audio/vad/vad_gmm.h
+++ b/webrtc/common_audio/vad/vad_gmm.h
@ -13,7 +13,7 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
 #define WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_

-#include "typedefs.h"
+#include "webrtc/typedefs.h"

 // Calculates the probability for |input|, given that |input| comes from a
 // normal distribution with mean and standard deviation (|mean|, |std|).
--- a/webrtc/common_audio/vad/vad_gmm_unittest.cc
+++ b/webrtc/common_audio/vad/vad_gmm_unittest.cc
@ -8,12 +8,12 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/typedefs.h"

 extern "C" {
-#include "vad_gmm.h"
+#include "webrtc/common_audio/vad/vad_gmm.h"
 }

 namespace {
--- a/webrtc/common_audio/vad/vad_sp.c
+++ b/webrtc/common_audio/vad/vad_sp.c
@ -8,13 +8,13 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "common_audio/vad/vad_sp.h"
+#include "webrtc/common_audio/vad/vad_sp.h"

 #include <assert.h>

-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "common_audio/vad/vad_core.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/typedefs.h"

 // Allpass filter coefficients, upper and lower, in Q13.
 // Upper: 0.64, Lower: 0.17.
--- a/webrtc/common_audio/vad/vad_sp.h
+++ b/webrtc/common_audio/vad/vad_sp.h
@ -14,8 +14,8 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
 #define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_

-#include "typedefs.h"
-#include "vad_core.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/typedefs.h"

 // Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
 //
--- a/webrtc/common_audio/vad/vad_sp_unittest.cc
+++ b/webrtc/common_audio/vad/vad_sp_unittest.cc
@ -10,13 +10,13 @@

 #include <stdlib.h>

-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_audio/vad/vad_unittest.h"
+#include "webrtc/typedefs.h"

 extern "C" {
-#include "vad_core.h"
-#include "vad_sp.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/common_audio/vad/vad_sp.h"
 }

 namespace {
--- a/webrtc/common_audio/vad/vad_unittest.cc
+++ b/webrtc/common_audio/vad/vad_unittest.cc
@ -8,15 +8,15 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "vad_unittest.h"
+#include "webrtc/common_audio/vad/vad_unittest.h"

 #include <stdlib.h>

-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"

-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "common_audio/vad/include/webrtc_vad.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/vad/include/webrtc_vad.h"
+#include "webrtc/typedefs.h"

 VadTest::VadTest() {}

--- a/webrtc/common_audio/vad/vad_unittest.h
+++ b/webrtc/common_audio/vad/vad_unittest.h
@ -13,9 +13,9 @@

 #include <stddef.h>  // size_t

-#include "gtest/gtest.h"
+#include "testing/gtest/include/gtest/gtest.h"

-#include "typedefs.h"
+#include "webrtc/typedefs.h"

 namespace {

--- a/webrtc/common_audio/vad/webrtc_vad.c
+++ b/webrtc/common_audio/vad/webrtc_vad.c
@ -8,14 +8,14 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "common_audio/vad/include/webrtc_vad.h"
+#include "webrtc/common_audio/vad/include/webrtc_vad.h"

 #include <stdlib.h>
 #include <string.h>

-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "common_audio/vad/vad_core.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_audio/vad/vad_core.h"
+#include "webrtc/typedefs.h"

 static const int kInitCheck = 42;
 static const int kValidRates[] = { 8000, 16000, 32000, 48000 };