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Include files from webrtc/.. paths in common_audio/

Browse Source 
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
This commit is contained in:
pbos@webrtc.org 2013-05-27 09:49:58 +00:00
parent 0afd84067a
commit aa30bb7ef5
63 changed files with 108 additions and 115 deletions
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2
webrtc/common_audio/resampler/include/resampler.h
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@ -16,7 +16,7 @@
#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_ #ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
#define WEBRTC_RESAMPLER_RESAMPLER_H_ #define WEBRTC_RESAMPLER_RESAMPLER_H_
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace webrtc namespace webrtc
{ {

1
webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
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@ -230,4 +230,3 @@ INSTANTIATE_TEST_CASE_P(
std::tr1::make_tuple(192000, 32000, -21.02, -10.94))); std::tr1::make_tuple(192000, 32000, -21.02, -10.94)));
} // namespace webrtc } // namespace webrtc

4
webrtc/common_audio/resampler/resampler.cc
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@ -16,8 +16,8 @@
#include <stdlib.h> #include <stdlib.h>
#include <string.h> #include <string.h>
#include "signal_processing_library.h" #include "webrtc/common_audio/resampler/include/resampler.h"
#include "resampler.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc namespace webrtc

4
webrtc/common_audio/resampler/resampler_unittest.cc
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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree. * 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. // TODO(andrew): this is a work-in-progress. Many more tests are needed.

1
webrtc/common_audio/resampler/sinc_resampler_unittest.cc
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@ -367,4 +367,3 @@ INSTANTIATE_TEST_CASE_P(
std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52))); std::tr1::make_tuple(192000, 192000, kResamplingRMSError, -73.52)));
} // namespace webrtc } // namespace webrtc

2
webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
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@ -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) void WebRtcSpl_AutoCorrToReflCoef(const int32_t *R, int use_order, int16_t *K)
{ {

2
webrtc/common_audio/signal_processing/auto_correlation.c
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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 WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
int in_vector_length, int in_vector_length,

3
webrtc/common_audio/signal_processing/complex_bit_reverse.c
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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 /* 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 * 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) {
} }
} }
} }

2
webrtc/common_audio/signal_processing/complex_fft.c
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@ -15,7 +15,7 @@
* *
*/ */
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#define CFFTSFT 14 #define CFFTSFT 14
#define CFFTRND 1 #define CFFTRND 1

2
webrtc/common_audio/signal_processing/copy_set_operations.c
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@ -25,7 +25,7 @@
*/ */
#include <string.h> #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) void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, int length)

2
webrtc/common_audio/signal_processing/cross_correlation.c
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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. */ /* C version of WebRtcSpl_CrossCorrelation() for generic platforms. */
void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation, void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,

2
webrtc/common_audio/signal_processing/division_operations.c
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@ -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) uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
{ {

2
webrtc/common_audio/signal_processing/dot_product_with_scale.c
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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, int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
const int16_t* vector2, const int16_t* vector2,

2
webrtc/common_audio/signal_processing/downsample_fast.c
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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. // TODO(Bjornv): Change the function parameter order to WebRTC code style.
// C version of WebRtcSpl_DownsampleFast() for generic platforms. // C version of WebRtcSpl_DownsampleFast() for generic platforms.

2
webrtc/common_audio/signal_processing/energy.c
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@ -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) int32_t WebRtcSpl_Energy(int16_t* vector, int vector_length, int* scale_factor)
{ {

2
webrtc/common_audio/signal_processing/filter_ar.c
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@ -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 WebRtcSpl_FilterAR(const int16_t* a,
int a_length, int a_length,

3
webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
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@ -9,7 +9,7 @@
*/ */
#include <assert.h> #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. // 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); data_out[i] = (int16_t)((output + 2048) >> 12);
} }
} }

2
webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
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@ -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, void WebRtcSpl_FilterMAFastQ12(int16_t* in_ptr,
int16_t* out_ptr, int16_t* out_ptr,

2
webrtc/common_audio/signal_processing/get_hanning_window.c
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@ -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 // Hanning table with 256 entries
static const int16_t kHanningTable[] = { static const int16_t kHanningTable[] = {

2
webrtc/common_audio/signal_processing/get_scaling_square.c
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@ -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) int WebRtcSpl_GetScalingSquare(int16_t *in_vector, int in_vector_length, int times)
{ {

2
webrtc/common_audio/signal_processing/ilbc_specific_functions.c
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@ -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, void WebRtcSpl_ReverseOrderMultArrayElements(int16_t *out, const int16_t *in,
const int16_t *win, const int16_t *win,

2
webrtc/common_audio/signal_processing/include/real_fft.h
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@ -11,7 +11,7 @@
#ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_ #ifndef WEBRTC_COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
#define 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; struct RealFFT;

2
webrtc/common_audio/signal_processing/include/spl_inl.h
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@ -16,7 +16,7 @@
#define WEBRTC_SPL_SPL_INL_H_ #define WEBRTC_SPL_SPL_INL_H_
#ifdef WEBRTC_ARCH_ARM_V7 #ifdef WEBRTC_ARCH_ARM_V7
#include "spl_inl_armv7.h" #include "webrtc/common_audio/signal_processing/include/spl_inl_armv7.h"
#else #else
static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) { static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {

2
webrtc/common_audio/signal_processing/levinson_durbin.c
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@ -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 #define SPL_LEVINSON_MAXORDER 20

2
webrtc/common_audio/signal_processing/lpc_to_refl_coef.c
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@ -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 #define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50

2
webrtc/common_audio/signal_processing/min_max_operations.c
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@ -24,7 +24,7 @@
* *
*/ */
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include <stdlib.h> #include <stdlib.h>

3
webrtc/common_audio/signal_processing/min_max_operations_mips.c
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@ -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. // Maximum absolute value of word16 vector.
int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, int length) { 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; return minimum;
} }

2
webrtc/common_audio/signal_processing/randomization_functions.c
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@ -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[] = { static const int16_t kRandNTable[] = {
9178, -7260, 40, 10189, 4894, -3531, -13779, 14764, 9178, -7260, 40, 10189, 4894, -3531, -13779, 14764,

4
webrtc/common_audio/signal_processing/real_fft.c
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <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 { struct RealFFT {
int order; int order;

8
webrtc/common_audio/signal_processing/real_fft_unittest.cc
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
namespace webrtc { namespace webrtc {
namespace { namespace {

2
webrtc/common_audio/signal_processing/refl_coef_to_lpc.c
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@ -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) void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
{ {

4
webrtc/common_audio/signal_processing/resample.c
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@ -15,8 +15,8 @@
* *
*/ */
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "resample_by_2_internal.h" #include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
// Declaration of internally used functions // Declaration of internally used functions
static void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In, int16_t *Out, static void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In, int16_t *Out,

4
webrtc/common_audio/signal_processing/resample_48khz.c
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@ -16,8 +16,8 @@
*/ */
#include <string.h> #include <string.h>
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "resample_by_2_internal.h" #include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
//////////////////////////// ////////////////////////////
///// 48 kHz -> 16 kHz ///// ///// 48 kHz -> 16 kHz /////

2
webrtc/common_audio/signal_processing/resample_by_2.c
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@ -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 #ifdef WEBRTC_ARCH_ARM_V7

2
webrtc/common_audio/signal_processing/resample_by_2_internal.c
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@ -14,7 +14,7 @@
* *
*/ */
#include "resample_by_2_internal.h" #include "webrtc/common_audio/signal_processing/resample_by_2_internal.h"
// allpass filter coefficients. // allpass filter coefficients.
static const int16_t kResampleAllpass[2][3] = { static const int16_t kResampleAllpass[2][3] = {

2
webrtc/common_audio/signal_processing/resample_by_2_internal.h
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@ -17,7 +17,7 @@
#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_ #ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
#define 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 * resample_by_2_fast.c

3
webrtc/common_audio/signal_processing/resample_by_2_mips.c
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@ -17,7 +17,7 @@
#if defined(MIPS32_LE) #if defined(MIPS32_LE)
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
// allpass filter coefficients. // allpass filter coefficients.
static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528}; 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) #endif // #if defined(MIPS32_LE)

2
webrtc/common_audio/signal_processing/resample_fractional.c
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@ -15,7 +15,7 @@
* *
*/ */
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
// interpolation coefficients // interpolation coefficients
static const int16_t kCoefficients48To32[2][8] = { static const int16_t kCoefficients48To32[2][8] = {

4
webrtc/common_audio/signal_processing/signal_processing_unittest.cc
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "signal_processing_library.h" #include "testing/gtest/include/gtest/gtest.h"
#include "gtest/gtest.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
static const int kVector16Size = 9; static const int kVector16Size = 9;
static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963, static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963,

6
webrtc/common_audio/signal_processing/spl_init.c
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@ -14,9 +14,9 @@
* Some code came from common/rtcd.c in the WebM project. * Some code came from common/rtcd.c in the WebM project.
*/ */
#include "common_audio/signal_processing/include/real_fft.h" #include "webrtc/common_audio/signal_processing/include/real_fft.h"
#include "common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "system_wrappers/interface/cpu_features_wrapper.h" #include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
/* Declare function pointers. */ /* Declare function pointers. */
MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16; MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;

2
webrtc/common_audio/signal_processing/spl_sqrt.c
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@ -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); int32_t WebRtcSpl_SqrtLocal(int32_t in);

2
webrtc/common_audio/signal_processing/spl_sqrt_floor.c
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@ -28,7 +28,7 @@
// Minor modifications in code style for WebRTC, 2012. // 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: * Algorithm:

1
webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
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@ -108,4 +108,3 @@ DEFINE_FUNCTION WebRtcSpl_SqrtFloor
bic r0, r2, #3 << 30 @ for rounding add: cmp r0, r2 adc r2, #1 bic r0, r2, #3 << 30 @ for rounding add: cmp r0, r2 adc r2, #1
bx lr bx lr

2
webrtc/common_audio/signal_processing/spl_version.c
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@ -16,7 +16,7 @@
*/ */
#include <string.h> #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) int16_t WebRtcSpl_get_version(char* version, int16_t length_in_bytes)
{ {

2
webrtc/common_audio/signal_processing/splitting_filter.c
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@ -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. // Number of samples in a low/high-band frame.
enum enum

2
webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
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@ -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, void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length,
int16_t *yQ15) int16_t *yQ15)

2
webrtc/common_audio/signal_processing/vector_scaling_operations.c
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@ -20,7 +20,7 @@
* WebRtcSpl_ScaleAndAddVectorsWithRoundC() * 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, void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length,
const int16_t *in, int16_t right_shifts) const int16_t *in, int16_t right_shifts)

2
webrtc/common_audio/vad/include/webrtc_vad.h
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@ -16,7 +16,7 @@
#ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT #ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ // NOLINT
#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_ #define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
#include "typedefs.h" // NOLINT #include "webrtc/typedefs.h"
typedef struct WebRtcVadInst VadInst; typedef struct WebRtcVadInst VadInst;

12
webrtc/common_audio/vad/vad_core.c
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_filterbank.h"
#include "vad_filterbank.h" #include "webrtc/common_audio/vad/vad_gmm.h"
#include "vad_gmm.h" #include "webrtc/common_audio/vad/vad_sp.h"
#include "vad_sp.h" #include "webrtc/typedefs.h"
// Spectrum Weighting // Spectrum Weighting
static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 }; static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };

4
webrtc/common_audio/vad/vad_core.h
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@ -16,8 +16,8 @@
#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_ #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
#define 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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
enum { kNumChannels = 6 }; // Number of frequency bands (named channels). enum { kNumChannels = 6 }; // Number of frequency bands (named channels).
enum { kNumGaussians = 2 }; // Number of Gaussians per channel in the GMM. enum { kNumGaussians = 2 }; // Number of Gaussians per channel in the GMM.

8
webrtc/common_audio/vad/vad_core_unittest.cc
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@ -10,12 +10,12 @@
#include <stdlib.h> #include <stdlib.h>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_unittest.h"
#include "vad_unittest.h" #include "webrtc/typedefs.h"
extern "C" { extern "C" {
#include "vad_core.h" #include "webrtc/common_audio/vad/vad_core.h"
} }
namespace { namespace {

6
webrtc/common_audio/vad/vad_filterbank.c
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <assert.h>
#include "signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
// Constants used in LogOfEnergy(). // Constants used in LogOfEnergy().
static const int16_t kLogConst = 24660; // 160*log10(2) in Q9. static const int16_t kLogConst = 24660; // 160*log10(2) in Q9.

4
webrtc/common_audio/vad/vad_filterbank.h
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@ -15,8 +15,8 @@
#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_ #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
#define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_ #define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "vad_core.h" #include "webrtc/typedefs.h"
// Takes |data_length| samples of |data_in| and calculates the logarithm of the // 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: // energy of each of the |kNumChannels| = 6 frequency bands used by the VAD:

10
webrtc/common_audio/vad/vad_filterbank_unittest.cc
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@ -10,13 +10,13 @@
#include <stdlib.h> #include <stdlib.h>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_unittest.h"
#include "vad_unittest.h" #include "webrtc/typedefs.h"
extern "C" { extern "C" {
#include "vad_core.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "vad_filterbank.h" #include "webrtc/common_audio/vad/vad_filterbank.h"
} }
namespace { namespace {

6
webrtc/common_audio/vad/vad_gmm.c
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
static const int32_t kCompVar = 22005; static const int32_t kCompVar = 22005;
static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12. static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.

2
webrtc/common_audio/vad/vad_gmm.h
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@ -13,7 +13,7 @@
#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_ #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
#define 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 // Calculates the probability for |input|, given that |input| comes from a
// normal distribution with mean and standard deviation (|mean|, |std|). // normal distribution with mean and standard deviation (|mean|, |std|).

8
webrtc/common_audio/vad/vad_gmm_unittest.cc
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "typedefs.h" #include "webrtc/common_audio/vad/vad_unittest.h"
#include "vad_unittest.h" #include "webrtc/typedefs.h"
extern "C" { extern "C" {
#include "vad_gmm.h" #include "webrtc/common_audio/vad/vad_gmm.h"
} }
namespace { namespace {

8
webrtc/common_audio/vad/vad_sp.c
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <assert.h>
#include "common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/vad/vad_core.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
// Allpass filter coefficients, upper and lower, in Q13. // Allpass filter coefficients, upper and lower, in Q13.
// Upper: 0.64, Lower: 0.17. // Upper: 0.64, Lower: 0.17.

4
webrtc/common_audio/vad/vad_sp.h
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@ -14,8 +14,8 @@
#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_ #ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
#define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_ #define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "vad_core.h" #include "webrtc/typedefs.h"
// Downsamples the signal by a factor 2, eg. 32->16 or 16->8. // Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
// //

10
webrtc/common_audio/vad/vad_sp_unittest.cc
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@ -10,13 +10,13 @@
#include <stdlib.h> #include <stdlib.h>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "typedefs.h" #include "webrtc/common_audio/vad/vad_unittest.h"
#include "vad_unittest.h" #include "webrtc/typedefs.h"
extern "C" { extern "C" {
#include "vad_core.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "vad_sp.h" #include "webrtc/common_audio/vad/vad_sp.h"
} }
namespace { namespace {

10
webrtc/common_audio/vad/vad_unittest.cc
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <stdlib.h>
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/vad/include/webrtc_vad.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
VadTest::VadTest() {} VadTest::VadTest() {}

4
webrtc/common_audio/vad/vad_unittest.h
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@ -13,9 +13,9 @@
#include <stddef.h> // size_t #include <stddef.h> // size_t
#include "gtest/gtest.h" #include "testing/gtest/include/gtest/gtest.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
namespace { namespace {

8
webrtc/common_audio/vad/webrtc_vad.c
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <stdlib.h>
#include <string.h> #include <string.h>
#include "common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "common_audio/vad/vad_core.h" #include "webrtc/common_audio/vad/vad_core.h"
#include "typedefs.h" #include "webrtc/typedefs.h"
static const int kInitCheck = 42; static const int kInitCheck = 42;
static const int kValidRates[] = { 8000, 16000, 32000, 48000 }; static const int kValidRates[] = { 8000, 16000, 32000, 48000 };