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-rw-r--r--build/generate_asm_header.gypi3
-rw-r--r--common_audio/signal_processing/include/signal_processing_library.h122
-rw-r--r--modules/audio_processing/aecm/aecm_core.h8
-rw-r--r--modules/audio_processing/ns/nsx_core.h52
4 files changed, 96 insertions, 89 deletions
diff --git a/build/generate_asm_header.gypi b/build/generate_asm_header.gypi
index e9f88ea7..f470288b 100644
--- a/build/generate_asm_header.gypi
+++ b/build/generate_asm_header.gypi
@@ -48,7 +48,8 @@
'<(webrtc_root)/build/generate_asm_header.py',
'--compiler=<!(/bin/echo -n ${ANDROID_GOMA_WRAPPER} '
'<(android_toolchain)/*-gcc)',
- '--options=-I.. -I<@(android_ndk_include) -S', # Compiler options.
+ # Compiler options.
+ '--options=-I<(webrtc_root)/.. -I<@(android_ndk_include) -S',
'--dir=<(out_dir)',
'<(RULE_INPUT_PATH)',
],
diff --git a/common_audio/signal_processing/include/signal_processing_library.h b/common_audio/signal_processing/include/signal_processing_library.h
index 50a89c37..0fe97c27 100644
--- a/common_audio/signal_processing/include/signal_processing_library.h
+++ b/common_audio/signal_processing/include/signal_processing_library.h
@@ -19,7 +19,7 @@
#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
#include <string.h>
-#include "typedefs.h"
+#include "webrtc/typedefs.h"
// Macros specific for the fixed point implementation
#define WEBRTC_SPL_WORD16_MAX 32767
@@ -28,8 +28,8 @@
#define WEBRTC_SPL_WORD32_MIN (WebRtc_Word32)0x80000000
#define WEBRTC_SPL_MAX_LPC_ORDER 14
#define WEBRTC_SPL_MAX_SEED_USED 0x80000000L
-#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
-#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
+#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
+#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
// TODO(kma/bjorn): For the next two macros, investigate how to correct the code
// for inputs of a = WEBRTC_SPL_WORD16_MIN or WEBRTC_SPL_WORD32_MIN.
#define WEBRTC_SPL_ABS_W16(a) \
@@ -152,15 +152,15 @@ extern "C"
#endif
#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(char))
+ memcpy(v1, v2, (length) * sizeof(char))
#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
+ memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
- memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
+ memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
// inline functions:
-#include "spl_inl.h"
+#include "webrtc/common_audio/signal_processing/include/spl_inl.h"
// Initialize SPL. Currently it contains only function pointer initialization.
// If the underlying platform is known to be ARM-Neon (WEBRTC_ARCH_ARM_NEON
@@ -636,8 +636,8 @@ void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
WebRtc_Word16* out_vector);
// End: Signal processing operations.
-// Randomization functions. Implementations collected in randomization_functions.c and
-// descriptions at bottom of this file.
+// Randomization functions. Implementations collected in
+// randomization_functions.c and descriptions at bottom of this file.
WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
@@ -681,12 +681,17 @@ int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
int scaling);
// Filter operations.
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
- G_CONST WebRtc_Word16* in_vector, int in_vector_length,
- WebRtc_Word16* filter_state, int filter_state_length,
+int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef,
+ int ar_coef_length,
+ G_CONST WebRtc_Word16* in_vector,
+ int in_vector_length,
+ WebRtc_Word16* filter_state,
+ int filter_state_length,
WebRtc_Word16* filter_state_low,
- int filter_state_low_length, WebRtc_Word16* out_vector,
- WebRtc_Word16* out_vector_low, int out_vector_low_length);
+ int filter_state_low_length,
+ WebRtc_Word16* out_vector,
+ WebRtc_Word16* out_vector_low,
+ int out_vector_low_length);
void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
WebRtc_Word16* out_vector,
@@ -797,11 +802,10 @@ void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
******************************************************************/
// state structure for 22 -> 16 resampler
-typedef struct
-{
- WebRtc_Word32 S_22_44[8];
- WebRtc_Word32 S_44_32[8];
- WebRtc_Word32 S_32_16[8];
+typedef struct {
+ WebRtc_Word32 S_22_44[8];
+ WebRtc_Word32 S_44_32[8];
+ WebRtc_Word32 S_32_16[8];
} WebRtcSpl_State22khzTo16khz;
void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
@@ -812,10 +816,9 @@ void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
// state structure for 16 -> 22 resampler
-typedef struct
-{
- WebRtc_Word32 S_16_32[8];
- WebRtc_Word32 S_32_22[8];
+typedef struct {
+ WebRtc_Word32 S_16_32[8];
+ WebRtc_Word32 S_32_22[8];
} WebRtcSpl_State16khzTo22khz;
void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
@@ -826,11 +829,10 @@ void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
// state structure for 22 -> 8 resampler
-typedef struct
-{
- WebRtc_Word32 S_22_22[16];
- WebRtc_Word32 S_22_16[8];
- WebRtc_Word32 S_16_8[8];
+typedef struct {
+ WebRtc_Word32 S_22_22[16];
+ WebRtc_Word32 S_22_16[8];
+ WebRtc_Word32 S_16_8[8];
} WebRtcSpl_State22khzTo8khz;
void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -840,11 +842,10 @@ void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
// state structure for 8 -> 22 resampler
-typedef struct
-{
- WebRtc_Word32 S_8_16[8];
- WebRtc_Word32 S_16_11[8];
- WebRtc_Word32 S_11_22[8];
+typedef struct {
+ WebRtc_Word32 S_8_16[8];
+ WebRtc_Word32 S_16_11[8];
+ WebRtc_Word32 S_11_22[8];
} WebRtcSpl_State8khzTo22khz;
void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -884,11 +885,10 @@ void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
*
******************************************************************/
-typedef struct
-{
- WebRtc_Word32 S_48_48[16];
- WebRtc_Word32 S_48_32[8];
- WebRtc_Word32 S_32_16[8];
+typedef struct {
+ WebRtc_Word32 S_48_48[16];
+ WebRtc_Word32 S_48_32[8];
+ WebRtc_Word32 S_32_16[8];
} WebRtcSpl_State48khzTo16khz;
void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -897,11 +897,10 @@ void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
-typedef struct
-{
- WebRtc_Word32 S_16_32[8];
- WebRtc_Word32 S_32_24[8];
- WebRtc_Word32 S_24_48[8];
+typedef struct {
+ WebRtc_Word32 S_16_32[8];
+ WebRtc_Word32 S_32_24[8];
+ WebRtc_Word32 S_24_48[8];
} WebRtcSpl_State16khzTo48khz;
void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -910,12 +909,11 @@ void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
-typedef struct
-{
- WebRtc_Word32 S_48_24[8];
- WebRtc_Word32 S_24_24[16];
- WebRtc_Word32 S_24_16[8];
- WebRtc_Word32 S_16_8[8];
+typedef struct {
+ WebRtc_Word32 S_48_24[8];
+ WebRtc_Word32 S_24_24[16];
+ WebRtc_Word32 S_24_16[8];
+ WebRtc_Word32 S_16_8[8];
} WebRtcSpl_State48khzTo8khz;
void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -924,12 +922,11 @@ void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
-typedef struct
-{
- WebRtc_Word32 S_8_16[8];
- WebRtc_Word32 S_16_12[8];
- WebRtc_Word32 S_12_24[8];
- WebRtc_Word32 S_24_48[8];
+typedef struct {
+ WebRtc_Word32 S_8_16[8];
+ WebRtc_Word32 S_16_12[8];
+ WebRtc_Word32 S_12_24[8];
+ WebRtc_Word32 S_24_48[8];
} WebRtcSpl_State8khzTo48khz;
void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
@@ -948,8 +945,8 @@ void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
WebRtc_Word16* out, WebRtc_Word32* filtState);
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
- WebRtc_Word32* filtState);
+void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
+ WebRtc_Word16* out, WebRtc_Word32* filtState);
/************************************************************
* END OF RESAMPLING FUNCTIONS
@@ -967,8 +964,8 @@ void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
#ifdef __cplusplus
}
-#endif // __cplusplus
-#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
+#endif // __cplusplus
+#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
//
// WebRtcSpl_AddSatW16(...)
@@ -1729,9 +1726,10 @@ void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
// This function gives the version string of the Signal Processing Library.
//
// Input:
-// - length_in_bytes : The size of Allocated space (in Bytes) where
-// the version number is written to (in string format).
+// - length_in_bytes : The size of Allocated space (in Bytes) where
+// the version number is written to (in string format).
//
// Output:
-// - version : Pointer to a buffer where the version number is written to.
+// - version : Pointer to a buffer where the version number is
+// written to.
//
diff --git a/modules/audio_processing/aecm/aecm_core.h b/modules/audio_processing/aecm/aecm_core.h
index 1038258d..8e78de7a 100644
--- a/modules/audio_processing/aecm/aecm_core.h
+++ b/modules/audio_processing/aecm/aecm_core.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-// Performs echo control (suppression) with fft routines in fixed-point
+// Performs echo control (suppression) with fft routines in fixed-point.
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "modules/audio_processing/aecm/aecm_defines.h"
-#include "typedefs.h"
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/aecm/aecm_defines.h"
+#include "webrtc/typedefs.h"
#ifdef _MSC_VER // visual c++
#define ALIGN8_BEG __declspec(align(8))
diff --git a/modules/audio_processing/ns/nsx_core.h b/modules/audio_processing/ns/nsx_core.h
index e5b3919a..1afbc5c1 100644
--- a/modules/audio_processing/ns/nsx_core.h
+++ b/modules/audio_processing/ns/nsx_core.h
@@ -11,14 +11,14 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "modules/audio_processing/ns/nsx_defines.h"
-#include "typedefs.h"
-
#ifdef NS_FILEDEBUG
#include <stdio.h>
#endif
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_processing/ns/nsx_defines.h"
+#include "webrtc/typedefs.h"
+
typedef struct NsxInst_t_ {
WebRtc_UWord32 fs;
@@ -44,7 +44,8 @@ typedef struct NsxInst_t_ {
WebRtc_Word32 maxLrt;
WebRtc_Word32 minLrt;
- WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
+ // Log LRT factor with time-smoothing in Q8.
+ WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL];
WebRtc_Word32 featureLogLrt;
WebRtc_Word32 thresholdLogLrt;
WebRtc_Word16 weightLogLrt;
@@ -57,35 +58,42 @@ typedef struct NsxInst_t_ {
WebRtc_UWord32 thresholdSpecFlat;
WebRtc_Word16 weightSpecFlat;
- WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
+ // Conservative estimate of noise spectrum.
+ WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL];
WebRtc_UWord32 magnEnergy;
WebRtc_UWord32 sumMagn;
WebRtc_UWord32 curAvgMagnEnergy;
WebRtc_UWord32 timeAvgMagnEnergy;
WebRtc_UWord32 timeAvgMagnEnergyTmp;
- WebRtc_UWord32 whiteNoiseLevel; //initial noise estimate
- WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
- WebRtc_Word32 pinkNoiseNumerator; //pink noise parameter: numerator
- WebRtc_Word32 pinkNoiseExp; //pink noise parameter: power of freq
- int minNorm; //smallest normalization factor
- int zeroInputSignal; //zero input signal flag
-
- WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
- WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
- WebRtc_Word16 priorNonSpeechProb; //prior speech/noise probability // Q14
-
- int blockIndex; //frame index counter
- int modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
+ WebRtc_UWord32 whiteNoiseLevel; // Initial noise estimate.
+ // Initial magnitude spectrum estimate.
+ WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];
+ // Pink noise parameters:
+ WebRtc_Word32 pinkNoiseNumerator; // Numerator.
+ WebRtc_Word32 pinkNoiseExp; // Power of freq.
+ int minNorm; // Smallest normalization factor.
+ int zeroInputSignal; // Zero input signal flag.
+
+ // Noise spectrum from previous frame.
+ WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL];
+ // Magnitude spectrum from previous frame.
+ WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL];
+ // Prior speech/noise probability in Q14.
+ WebRtc_Word16 priorNonSpeechProb;
+
+ int blockIndex; // Frame index counter.
+ // Parameter for updating or estimating thresholds/weights for prior model.
+ int modelUpdate;
int cntThresUpdate;
- //histograms for parameter estimation
+ // Histograms for parameter estimation.
WebRtc_Word16 histLrt[HIST_PAR_EST];
WebRtc_Word16 histSpecFlat[HIST_PAR_EST];
WebRtc_Word16 histSpecDiff[HIST_PAR_EST];
- //quantities for high band estimate
- WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
+ // Quantities for high band estimate.
+ WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; // Q0
int qNoise;
int prevQNoise;
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-25 10:37:31 +0000