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update code.

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This commit is contained in:
luocai 2024-09-05 11:22:29 +08:00
parent 268d8160ea
commit 76d9e6c8cb
29 changed files with 8798 additions and 0 deletions
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17
VocieProcess/CMakeLists.txt
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@ -26,11 +26,22 @@ add_library(VocieProcess
api/units/timestamp.h api/units/timestamp.cc
common_audio/channel_buffer.h common_audio/channel_buffer.cc
common_audio/ring_buffer.h common_audio/ring_buffer.c
common_audio/resampler/push_sinc_resampler.h common_audio/resampler/push_sinc_resampler.cc
common_audio/resampler/sinc_resampler.h common_audio/resampler/sinc_resampler.cc
common_audio/signal_processing/complex_bit_reverse.c
common_audio/signal_processing/complex_fft.c
common_audio/signal_processing/cross_correlation.c
common_audio/signal_processing/division_operations.c
common_audio/signal_processing/dot_product_with_scale.h common_audio/signal_processing/dot_product_with_scale.cc
common_audio/signal_processing/downsample_fast.c
common_audio/signal_processing/min_max_operations.c
common_audio/signal_processing/randomization_functions.c
common_audio/signal_processing/real_fft.c
common_audio/signal_processing/spl_init.c
common_audio/signal_processing/vector_scaling_operations.c
common_audio/third_party/ooura/fft_size_128/ooura_fft.h common_audio/third_party/ooura/fft_size_128/ooura_fft.cc
common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
@ -117,9 +128,14 @@ add_library(VocieProcess
modules/audio_processing/aec3/suppression_gain.h modules/audio_processing/aec3/suppression_gain.cc
modules/audio_processing/aec3/transparent_mode.h modules/audio_processing/aec3/transparent_mode.cc
modules/audio_processing/aecm/aecm_core.h modules/audio_processing/aecm/aecm_core.cc modules/audio_processing/aecm/aecm_core_c.cc
modules/audio_processing/aecm/echo_control_mobile.h modules/audio_processing/aecm/echo_control_mobile.cc
modules/audio_processing/logging/apm_data_dumper.h modules/audio_processing/logging/apm_data_dumper.cc
modules/audio_processing/utility/cascaded_biquad_filter.h modules/audio_processing/utility/cascaded_biquad_filter.cc
modules/audio_processing/utility/delay_estimator_wrapper.h modules/audio_processing/utility/delay_estimator_wrapper.cc
modules/audio_processing/utility/delay_estimator.h modules/audio_processing/utility/delay_estimator.cc
)
target_compile_definitions(VocieProcess
@ -133,6 +149,7 @@ target_compile_definitions(VocieProcess
target_include_directories(VocieProcess
PRIVATE ${CMAKE_CURRENT_SOURCE_DIR}
INTERFACE ${CMAKE_CURRENT_SOURCE_DIR}
)
target_link_libraries(VocieProcess

232
VocieProcess/common_audio/ring_buffer.c Normal file
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@ -0,0 +1,232 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
// otherwise specified, functions return 0 on success and -1 on error.
#include "common_audio/ring_buffer.h"
#include <stddef.h> // size_t
#include <stdlib.h>
#include <string.h>
// Get address of region(s) from which we can read data.
// If the region is contiguous, `data_ptr_bytes_2` will be zero.
// If non-contiguous, `data_ptr_bytes_2` will be the size in bytes of the second
// region. Returns room available to be read or `element_count`, whichever is
// smaller.
static size_t GetBufferReadRegions(RingBuffer* buf,
size_t element_count,
void** data_ptr_1,
size_t* data_ptr_bytes_1,
void** data_ptr_2,
size_t* data_ptr_bytes_2) {
const size_t readable_elements = WebRtc_available_read(buf);
const size_t read_elements = (readable_elements < element_count ?
readable_elements : element_count);
const size_t margin = buf->element_count - buf->read_pos;
// Check to see if read is not contiguous.
if (read_elements > margin) {
// Write data in two blocks that wrap the buffer.
*data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
*data_ptr_bytes_1 = margin * buf->element_size;
*data_ptr_2 = buf->data;
*data_ptr_bytes_2 = (read_elements - margin) * buf->element_size;
} else {
*data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
*data_ptr_bytes_1 = read_elements * buf->element_size;
*data_ptr_2 = NULL;
*data_ptr_bytes_2 = 0;
}
return read_elements;
}
RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size) {
RingBuffer* self = NULL;
if (element_count == 0 || element_size == 0) {
return NULL;
}
self = malloc(sizeof(RingBuffer));
if (!self) {
return NULL;
}
self->data = malloc(element_count * element_size);
if (!self->data) {
free(self);
self = NULL;
return NULL;
}
self->element_count = element_count;
self->element_size = element_size;
WebRtc_InitBuffer(self);
return self;
}
void WebRtc_InitBuffer(RingBuffer* self) {
self->read_pos = 0;
self->write_pos = 0;
self->rw_wrap = SAME_WRAP;
// Initialize buffer to zeros
memset(self->data, 0, self->element_count * self->element_size);
}
void WebRtc_FreeBuffer(void* handle) {
RingBuffer* self = (RingBuffer*)handle;
if (!self) {
return;
}
free(self->data);
free(self);
}
size_t WebRtc_ReadBuffer(RingBuffer* self,
void** data_ptr,
void* data,
size_t element_count) {
if (self == NULL) {
return 0;
}
if (data == NULL) {
return 0;
}
{
void* buf_ptr_1 = NULL;
void* buf_ptr_2 = NULL;
size_t buf_ptr_bytes_1 = 0;
size_t buf_ptr_bytes_2 = 0;
const size_t read_count = GetBufferReadRegions(self,
element_count,
&buf_ptr_1,
&buf_ptr_bytes_1,
&buf_ptr_2,
&buf_ptr_bytes_2);
if (buf_ptr_bytes_2 > 0) {
// We have a wrap around when reading the buffer. Copy the buffer data to
// `data` and point to it.
memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
buf_ptr_1 = data;
} else if (!data_ptr) {
// No wrap, but a memcpy was requested.
memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
}
if (data_ptr) {
// `buf_ptr_1` == `data` in the case of a wrap.
*data_ptr = read_count == 0 ? NULL : buf_ptr_1;
}
// Update read position
WebRtc_MoveReadPtr(self, (int) read_count);
return read_count;
}
}
size_t WebRtc_WriteBuffer(RingBuffer* self,
const void* data,
size_t element_count) {
if (!self) {
return 0;
}
if (!data) {
return 0;
}
{
const size_t free_elements = WebRtc_available_write(self);
const size_t write_elements = (free_elements < element_count ? free_elements
: element_count);
size_t n = write_elements;
const size_t margin = self->element_count - self->write_pos;
if (write_elements > margin) {
// Buffer wrap around when writing.
memcpy(self->data + self->write_pos * self->element_size,
data, margin * self->element_size);
self->write_pos = 0;
n -= margin;
self->rw_wrap = DIFF_WRAP;
}
memcpy(self->data + self->write_pos * self->element_size,
((const char*) data) + ((write_elements - n) * self->element_size),
n * self->element_size);
self->write_pos += n;
return write_elements;
}
}
int WebRtc_MoveReadPtr(RingBuffer* self, int element_count) {
if (!self) {
return 0;
}
{
// We need to be able to take care of negative changes, hence use "int"
// instead of "size_t".
const int free_elements = (int) WebRtc_available_write(self);
const int readable_elements = (int) WebRtc_available_read(self);
int read_pos = (int) self->read_pos;
if (element_count > readable_elements) {
element_count = readable_elements;
}
if (element_count < -free_elements) {
element_count = -free_elements;
}
read_pos += element_count;
if (read_pos > (int) self->element_count) {
// Buffer wrap around. Restart read position and wrap indicator.
read_pos -= (int) self->element_count;
self->rw_wrap = SAME_WRAP;
}
if (read_pos < 0) {
// Buffer wrap around. Restart read position and wrap indicator.
read_pos += (int) self->element_count;
self->rw_wrap = DIFF_WRAP;
}
self->read_pos = (size_t) read_pos;
return element_count;
}
}
size_t WebRtc_available_read(const RingBuffer* self) {
if (!self) {
return 0;
}
if (self->rw_wrap == SAME_WRAP) {
return self->write_pos - self->read_pos;
} else {
return self->element_count - self->read_pos + self->write_pos;
}
}
size_t WebRtc_available_write(const RingBuffer* self) {
if (!self) {
return 0;
}
return self->element_count - WebRtc_available_read(self);
}

79
VocieProcess/common_audio/ring_buffer.h Normal file
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@ -0,0 +1,79 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
// otherwise specified, functions return 0 on success and -1 on error.
#ifndef COMMON_AUDIO_RING_BUFFER_H_
#define COMMON_AUDIO_RING_BUFFER_H_
// TODO(alessiob): Used by AEC, AECm and AudioRingBuffer. Remove when possible.
#ifdef __cplusplus
extern "C" {
#endif
#include <stddef.h> // size_t
enum Wrap { SAME_WRAP, DIFF_WRAP };
typedef struct RingBuffer {
size_t read_pos;
size_t write_pos;
size_t element_count;
size_t element_size;
enum Wrap rw_wrap;
char* data;
} RingBuffer;
// Creates and initializes the buffer. Returns null on failure.
RingBuffer* WebRtc_CreateBuffer(size_t element_count, size_t element_size);
void WebRtc_InitBuffer(RingBuffer* handle);
void WebRtc_FreeBuffer(void* handle);
// Reads data from the buffer. Returns the number of elements that were read.
// The `data_ptr` will point to the address where the read data is located.
// If no data can be read, `data_ptr` is set to `NULL`. If all data can be read
// without buffer wrap around then `data_ptr` will point to the location in the
// buffer. Otherwise, the data will be copied to `data` (memory allocation done
// by the user) and `data_ptr` points to the address of `data`. `data_ptr` is
// only guaranteed to be valid until the next call to WebRtc_WriteBuffer().
//
// To force a copying to `data`, pass a null `data_ptr`.
//
// Returns number of elements read.
size_t WebRtc_ReadBuffer(RingBuffer* handle,
void** data_ptr,
void* data,
size_t element_count);
// Writes `data` to buffer and returns the number of elements written.
size_t WebRtc_WriteBuffer(RingBuffer* handle,
const void* data,
size_t element_count);
// Moves the buffer read position and returns the number of elements moved.
// Positive `element_count` moves the read position towards the write position,
// that is, flushing the buffer. Negative `element_count` moves the read
// position away from the the write position, that is, stuffing the buffer.
// Returns number of elements moved.
int WebRtc_MoveReadPtr(RingBuffer* handle, int element_count);
// Returns number of available elements to read.
size_t WebRtc_available_read(const RingBuffer* handle);
// Returns number of available elements for write.
size_t WebRtc_available_write(const RingBuffer* handle);
#ifdef __cplusplus
}
#endif
#endif // COMMON_AUDIO_RING_BUFFER_H_

108
VocieProcess/common_audio/signal_processing/complex_bit_reverse.c Normal file
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@ -0,0 +1,108 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/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
* operations, while index_7[{1, 3, 5, ...}] are for the right side of the
* operation. Same for index_8.
*/
/* Indexes for the case of stages == 7. */
static const int16_t index_7[112] = {
1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
103, 115, 111, 123
};
/* Indexes for the case of stages == 8. */
static const int16_t index_8[240] = {
1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
203, 211, 207, 243, 215, 235, 223, 251, 239, 247
};
void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
/* For any specific value of stages, we know exactly the indexes that are
* bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
* stages are 7 and 8, so we use tables to save unnecessary iterations and
* calculations for these two cases.
*/
if (stages == 7 || stages == 8) {
int m = 0;
int length = 112;
const int16_t* index = index_7;
if (stages == 8) {
length = 240;
index = index_8;
}
/* Decimation in time. Swap the elements with bit-reversed indexes. */
for (m = 0; m < length; m += 2) {
/* We declare a int32_t* type pointer, to load both the 16-bit real
* and imaginary elements from complex_data in one instruction, reducing
* complexity.
*/
int32_t* complex_data_ptr = (int32_t*)complex_data;
int32_t temp = 0;
temp = complex_data_ptr[index[m]]; /* Real and imaginary */
complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
complex_data_ptr[index[m + 1]] = temp;
}
}
else {
int m = 0, mr = 0, l = 0;
int n = 1 << stages;
int nn = n - 1;
/* Decimation in time - re-order data */
for (m = 1; m <= nn; ++m) {
int32_t* complex_data_ptr = (int32_t*)complex_data;
int32_t temp = 0;
/* Find out indexes that are bit-reversed. */
l = n;
do {
l >>= 1;
} while (l > nn - mr);
mr = (mr & (l - 1)) + l;
if (mr <= m) {
continue;
}
/* Swap the elements with bit-reversed indexes.
* This is similar to the loop in the stages == 7 or 8 cases.
*/
temp = complex_data_ptr[m]; /* Real and imaginary */
complex_data_ptr[m] = complex_data_ptr[mr];
complex_data_ptr[mr] = temp;
}
}
}

299
VocieProcess/common_audio/signal_processing/complex_fft.c Normal file
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@ -0,0 +1,299 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the function WebRtcSpl_ComplexFFT().
* The description header can be found in signal_processing_library.h
*
*/
#include "common_audio/signal_processing/complex_fft_tables.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/system/arch.h"
#define CFFTSFT 14
#define CFFTRND 1
#define CFFTRND2 16384
#define CIFFTSFT 14
#define CIFFTRND 1
int WebRtcSpl_ComplexFFT(int16_t frfi[], int stages, int mode)
{
int i, j, l, k, istep, n, m;
int16_t wr, wi;
int32_t tr32, ti32, qr32, qi32;
/* The 1024-value is a constant given from the size of kSinTable1024[],
* and should not be changed depending on the input parameter 'stages'
*/
n = 1 << stages;
if (n > 1024)
return -1;
l = 1;
k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
depending on the input parameter 'stages' */
if (mode == 0)
{
// mode==0: Low-complexity and Low-accuracy mode
while (l < n)
{
istep = l << 1;
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = -kSinTable1024[j];
for (i = m; i < n; i += istep)
{
j = i + l;
tr32 = (wr * frfi[2 * j] - wi * frfi[2 * j + 1]) >> 15;
ti32 = (wr * frfi[2 * j + 1] + wi * frfi[2 * j]) >> 15;
qr32 = (int32_t)frfi[2 * i];
qi32 = (int32_t)frfi[2 * i + 1];
frfi[2 * j] = (int16_t)((qr32 - tr32) >> 1);
frfi[2 * j + 1] = (int16_t)((qi32 - ti32) >> 1);
frfi[2 * i] = (int16_t)((qr32 + tr32) >> 1);
frfi[2 * i + 1] = (int16_t)((qi32 + ti32) >> 1);
}
}
--k;
l = istep;
}
} else
{
// mode==1: High-complexity and High-accuracy mode
while (l < n)
{
istep = l << 1;
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = -kSinTable1024[j];
#ifdef WEBRTC_ARCH_ARM_V7
int32_t wri = 0;
__asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
"r"((int32_t)wr), "r"((int32_t)wi));
#endif
for (i = m; i < n; i += istep)
{
j = i + l;
#ifdef WEBRTC_ARCH_ARM_V7
register int32_t frfi_r;
__asm __volatile(
"pkhbt %[frfi_r], %[frfi_even], %[frfi_odd],"
" lsl #16\n\t"
"smlsd %[tr32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
"smladx %[ti32], %[wri], %[frfi_r], %[cfftrnd]\n\t"
:[frfi_r]"=&r"(frfi_r),
[tr32]"=&r"(tr32),
[ti32]"=r"(ti32)
:[frfi_even]"r"((int32_t)frfi[2*j]),
[frfi_odd]"r"((int32_t)frfi[2*j +1]),
[wri]"r"(wri),
[cfftrnd]"r"(CFFTRND));
#else
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CFFTRND;
#endif
tr32 >>= 15 - CFFTSFT;
ti32 >>= 15 - CFFTSFT;
qr32 = ((int32_t)frfi[2 * i]) * (1 << CFFTSFT);
qi32 = ((int32_t)frfi[2 * i + 1]) * (1 << CFFTSFT);
frfi[2 * j] = (int16_t)(
(qr32 - tr32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * j + 1] = (int16_t)(
(qi32 - ti32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * i] = (int16_t)(
(qr32 + tr32 + CFFTRND2) >> (1 + CFFTSFT));
frfi[2 * i + 1] = (int16_t)(
(qi32 + ti32 + CFFTRND2) >> (1 + CFFTSFT));
}
}
--k;
l = istep;
}
}
return 0;
}
int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
{
size_t i, j, l, istep, n, m;
int k, scale, shift;
int16_t wr, wi;
int32_t tr32, ti32, qr32, qi32;
int32_t tmp32, round2;
/* The 1024-value is a constant given from the size of kSinTable1024[],
* and should not be changed depending on the input parameter 'stages'
*/
n = ((size_t)1) << stages;
if (n > 1024)
return -1;
scale = 0;
l = 1;
k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
depending on the input parameter 'stages' */
while (l < n)
{
// variable scaling, depending upon data
shift = 0;
round2 = 8192;
tmp32 = WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
if (tmp32 > 13573)
{
shift++;
scale++;
round2 <<= 1;
}
if (tmp32 > 27146)
{
shift++;
scale++;
round2 <<= 1;
}
istep = l << 1;
if (mode == 0)
{
// mode==0: Low-complexity and Low-accuracy mode
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = kSinTable1024[j];
for (i = m; i < n; i += istep)
{
j = i + l;
tr32 = (wr * frfi[2 * j] - wi * frfi[2 * j + 1]) >> 15;
ti32 = (wr * frfi[2 * j + 1] + wi * frfi[2 * j]) >> 15;
qr32 = (int32_t)frfi[2 * i];
qi32 = (int32_t)frfi[2 * i + 1];
frfi[2 * j] = (int16_t)((qr32 - tr32) >> shift);
frfi[2 * j + 1] = (int16_t)((qi32 - ti32) >> shift);
frfi[2 * i] = (int16_t)((qr32 + tr32) >> shift);
frfi[2 * i + 1] = (int16_t)((qi32 + ti32) >> shift);
}
}
} else
{
// mode==1: High-complexity and High-accuracy mode
for (m = 0; m < l; ++m)
{
j = m << k;
/* The 256-value is a constant given as 1/4 of the size of
* kSinTable1024[], and should not be changed depending on the input
* parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
*/
wr = kSinTable1024[j + 256];
wi = kSinTable1024[j];
#ifdef WEBRTC_ARCH_ARM_V7
int32_t wri = 0;
__asm __volatile("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
"r"((int32_t)wr), "r"((int32_t)wi));
#endif
for (i = m; i < n; i += istep)
{
j = i + l;
#ifdef WEBRTC_ARCH_ARM_V7
register int32_t frfi_r;
__asm __volatile(
"pkhbt %[frfi_r], %[frfi_even], %[frfi_odd], lsl #16\n\t"
"smlsd %[tr32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
"smladx %[ti32], %[wri], %[frfi_r], %[cifftrnd]\n\t"
:[frfi_r]"=&r"(frfi_r),
[tr32]"=&r"(tr32),
[ti32]"=r"(ti32)
:[frfi_even]"r"((int32_t)frfi[2*j]),
[frfi_odd]"r"((int32_t)frfi[2*j +1]),
[wri]"r"(wri),
[cifftrnd]"r"(CIFFTRND)
);
#else
tr32 = wr * frfi[2 * j] - wi * frfi[2 * j + 1] + CIFFTRND;
ti32 = wr * frfi[2 * j + 1] + wi * frfi[2 * j] + CIFFTRND;
#endif
tr32 >>= 15 - CIFFTSFT;
ti32 >>= 15 - CIFFTSFT;
qr32 = ((int32_t)frfi[2 * i]) * (1 << CIFFTSFT);
qi32 = ((int32_t)frfi[2 * i + 1]) * (1 << CIFFTSFT);
frfi[2 * j] = (int16_t)(
(qr32 - tr32 + round2) >> (shift + CIFFTSFT));
frfi[2 * j + 1] = (int16_t)(
(qi32 - ti32 + round2) >> (shift + CIFFTSFT));
frfi[2 * i] = (int16_t)(
(qr32 + tr32 + round2) >> (shift + CIFFTSFT));
frfi[2 * i + 1] = (int16_t)(
(qi32 + ti32 + round2) >> (shift + CIFFTSFT));
}
}
}
--k;
l = istep;
}
return scale;
}

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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
#include <stdint.h>
static const int16_t kSinTable1024[] = {
0, 201, 402, 603, 804, 1005, 1206, 1406, 1607,
1808, 2009, 2209, 2410, 2610, 2811, 3011, 3211, 3411,
3611, 3811, 4011, 4210, 4409, 4608, 4807, 5006, 5205,
5403, 5601, 5799, 5997, 6195, 6392, 6589, 6786, 6982,
7179, 7375, 7571, 7766, 7961, 8156, 8351, 8545, 8739,
8932, 9126, 9319, 9511, 9703, 9895, 10087, 10278, 10469,
10659, 10849, 11038, 11227, 11416, 11604, 11792, 11980, 12166,
12353, 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268, 15446,
15623, 15799, 15975, 16150, 16325, 16499, 16672, 16845, 17017,
17189, 17360, 17530, 17699, 17868, 18036, 18204, 18371, 18537,
18702, 18867, 19031, 19194, 19357, 19519, 19680, 19840, 20000,
20159, 20317, 20474, 20631, 20787, 20942, 21096, 21249, 21402,
21554, 21705, 21855, 22004, 22153, 22301, 22448, 22594, 22739,
22883, 23027, 23169, 23311, 23452, 23592, 23731, 23869, 24006,
24143, 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198, 26318,
26437, 26556, 26673, 26789, 26905, 27019, 27132, 27244, 27355,
27466, 27575, 27683, 27790, 27896, 28001, 28105, 28208, 28309,
28410, 28510, 28608, 28706, 28802, 28897, 28992, 29085, 29177,
29268, 29358, 29446, 29534, 29621, 29706, 29790, 29873, 29955,
30036, 30116, 30195, 30272, 30349, 30424, 30498, 30571, 30643,
30713, 30783, 30851, 30918, 30984, 31049, 31113, 31175, 31236,
31297, 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097, 32137,
32176, 32213, 32249, 32284, 32318, 32350, 32382, 32412, 32441,
32468, 32495, 32520, 32544, 32567, 32588, 32609, 32628, 32646,
32662, 32678, 32692, 32705, 32717, 32727, 32736, 32744, 32751,
32757, 32761, 32764, 32766, 32767, 32766, 32764, 32761, 32757,
32751, 32744, 32736, 32727, 32717, 32705, 32692, 32678, 32662,
32646, 32628, 32609, 32588, 32567, 32544, 32520, 32495, 32468,
32441, 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833, 31785,
31735, 31684, 31633, 31580, 31525, 31470, 31413, 31356, 31297,
31236, 31175, 31113, 31049, 30984, 30918, 30851, 30783, 30713,
30643, 30571, 30498, 30424, 30349, 30272, 30195, 30116, 30036,
29955, 29873, 29790, 29706, 29621, 29534, 29446, 29358, 29268,
29177, 29085, 28992, 28897, 28802, 28706, 28608, 28510, 28410,
28309, 28208, 28105, 28001, 27896, 27790, 27683, 27575, 27466,
27355, 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456, 25329,
25201, 25072, 24942, 24811, 24679, 24546, 24413, 24278, 24143,
24006, 23869, 23731, 23592, 23452, 23311, 23169, 23027, 22883,
22739, 22594, 22448, 22301, 22153, 22004, 21855, 21705, 21554,
21402, 21249, 21096, 20942, 20787, 20631, 20474, 20317, 20159,
20000, 19840, 19680, 19519, 19357, 19194, 19031, 18867, 18702,
18537, 18371, 18204, 18036, 17868, 17699, 17530, 17360, 17189,
17017, 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191, 14009,
13827, 13645, 13462, 13278, 13094, 12909, 12724, 12539, 12353,
12166, 11980, 11792, 11604, 11416, 11227, 11038, 10849, 10659,
10469, 10278, 10087, 9895, 9703, 9511, 9319, 9126, 8932,
8739, 8545, 8351, 8156, 7961, 7766, 7571, 7375, 7179,
6982, 6786, 6589, 6392, 6195, 5997, 5799, 5601, 5403,
5205, 5006, 4807, 4608, 4409, 4210, 4011, 3811, 3611,
3411, 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
1607, 1406, 1206, 1005, 804, 603, 402, 201, 0,
-201, -402, -603, -804, -1005, -1206, -1406, -1607, -1808,
-2009, -2209, -2410, -2610, -2811, -3011, -3211, -3411, -3611,
-3811, -4011, -4210, -4409, -4608, -4807, -5006, -5205, -5403,
-5601, -5799, -5997, -6195, -6392, -6589, -6786, -6982, -7179,
-7375, -7571, -7766, -7961, -8156, -8351, -8545, -8739, -8932,
-9126, -9319, -9511, -9703, -9895, -10087, -10278, -10469, -10659,
-10849, -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
-12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827, -14009,
-14191, -14372, -14552, -14732, -14911, -15090, -15268, -15446, -15623,
-15799, -15975, -16150, -16325, -16499, -16672, -16845, -17017, -17189,
-17360, -17530, -17699, -17868, -18036, -18204, -18371, -18537, -18702,
-18867, -19031, -19194, -19357, -19519, -19680, -19840, -20000, -20159,
-20317, -20474, -20631, -20787, -20942, -21096, -21249, -21402, -21554,
-21705, -21855, -22004, -22153, -22301, -22448, -22594, -22739, -22883,
-23027, -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
-24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201, -25329,
-25456, -25582, -25707, -25831, -25954, -26077, -26198, -26318, -26437,
-26556, -26673, -26789, -26905, -27019, -27132, -27244, -27355, -27466,
-27575, -27683, -27790, -27896, -28001, -28105, -28208, -28309, -28410,
-28510, -28608, -28706, -28802, -28897, -28992, -29085, -29177, -29268,
-29358, -29446, -29534, -29621, -29706, -29790, -29873, -29955, -30036,
-30116, -30195, -30272, -30349, -30424, -30498, -30571, -30643, -30713,
-30783, -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
-31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735, -31785,
-31833, -31880, -31926, -31970, -32014, -32056, -32097, -32137, -32176,
-32213, -32249, -32284, -32318, -32350, -32382, -32412, -32441, -32468,
-32495, -32520, -32544, -32567, -32588, -32609, -32628, -32646, -32662,
-32678, -32692, -32705, -32717, -32727, -32736, -32744, -32751, -32757,
-32761, -32764, -32766, -32767, -32766, -32764, -32761, -32757, -32751,
-32744, -32736, -32727, -32717, -32705, -32692, -32678, -32662, -32646,
-32628, -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
-32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176, -32137,
-32097, -32056, -32014, -31970, -31926, -31880, -31833, -31785, -31735,
-31684, -31633, -31580, -31525, -31470, -31413, -31356, -31297, -31236,
-31175, -31113, -31049, -30984, -30918, -30851, -30783, -30713, -30643,
-30571, -30498, -30424, -30349, -30272, -30195, -30116, -30036, -29955,
-29873, -29790, -29706, -29621, -29534, -29446, -29358, -29268, -29177,
-29085, -28992, -28897, -28802, -28706, -28608, -28510, -28410, -28309,
-28208, -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
-27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437, -26318,
-26198, -26077, -25954, -25831, -25707, -25582, -25456, -25329, -25201,
-25072, -24942, -24811, -24679, -24546, -24413, -24278, -24143, -24006,
-23869, -23731, -23592, -23452, -23311, -23169, -23027, -22883, -22739,
-22594, -22448, -22301, -22153, -22004, -21855, -21705, -21554, -21402,
-21249, -21096, -20942, -20787, -20631, -20474, -20317, -20159, -20000,
-19840, -19680, -19519, -19357, -19194, -19031, -18867, -18702, -18537,
-18371, -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
-16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623, -15446,
-15268, -15090, -14911, -14732, -14552, -14372, -14191, -14009, -13827,
-13645, -13462, -13278, -13094, -12909, -12724, -12539, -12353, -12166,
-11980, -11792, -11604, -11416, -11227, -11038, -10849, -10659, -10469,
-10278, -10087, -9895, -9703, -9511, -9319, -9126, -8932, -8739,
-8545, -8351, -8156, -7961, -7766, -7571, -7375, -7179, -6982,
-6786, -6589, -6392, -6195, -5997, -5799, -5601, -5403, -5205,
-5006, -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
-3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808, -1607,
-1406, -1206, -1005, -804, -603, -402, -201};
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_

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VocieProcess/common_audio/signal_processing/cross_correlation.c Normal file
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/signal_processing/include/signal_processing_library.h"
/* C version of WebRtcSpl_CrossCorrelation() for generic platforms. */
void WebRtcSpl_CrossCorrelationC(int32_t* cross_correlation,
const int16_t* seq1,
const int16_t* seq2,
size_t dim_seq,
size_t dim_cross_correlation,
int right_shifts,
int step_seq2) {
size_t i = 0, j = 0;
for (i = 0; i < dim_cross_correlation; i++) {
int32_t corr = 0;
for (j = 0; j < dim_seq; j++)
corr += (seq1[j] * seq2[j]) >> right_shifts;
seq2 += step_seq2;
*cross_correlation++ = corr;
}
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains implementations of the divisions
* WebRtcSpl_DivU32U16()
* WebRtcSpl_DivW32W16()
* WebRtcSpl_DivW32W16ResW16()
* WebRtcSpl_DivResultInQ31()
* WebRtcSpl_DivW32HiLow()
*
* The description header can be found in signal_processing_library.h
*
*/
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/sanitizer.h"
uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (uint32_t)(num / den);
} else
{
return (uint32_t)0xFFFFFFFF;
}
}
int32_t WebRtcSpl_DivW32W16(int32_t num, int16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (int32_t)(num / den);
} else
{
return (int32_t)0x7FFFFFFF;
}
}
int16_t WebRtcSpl_DivW32W16ResW16(int32_t num, int16_t den)
{
// Guard against division with 0
if (den != 0)
{
return (int16_t)(num / den);
} else
{
return (int16_t)0x7FFF;
}
}
int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den)
{
int32_t L_num = num;
int32_t L_den = den;
int32_t div = 0;
int k = 31;
int change_sign = 0;
if (num == 0)
return 0;
if (num < 0)
{
change_sign++;
L_num = -num;
}
if (den < 0)
{
change_sign++;
L_den = -den;
}
while (k--)
{
div <<= 1;
L_num <<= 1;
if (L_num >= L_den)
{
L_num -= L_den;
div++;
}
}
if (change_sign == 1)
{
div = -div;
}
return div;
}
int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low)
{
int16_t approx, tmp_hi, tmp_low, num_hi, num_low;
int32_t tmpW32;
approx = (int16_t)WebRtcSpl_DivW32W16((int32_t)0x1FFFFFFF, den_hi);
// result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
// tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
tmpW32 = (den_hi * approx << 1) + ((den_low * approx >> 15) << 1);
// tmpW32 = den * approx
// result in Q30 (tmpW32 = 2.0-(den*approx))
tmpW32 = (int32_t)((int64_t)0x7fffffffL - tmpW32);
// Store tmpW32 in hi and low format
tmp_hi = (int16_t)(tmpW32 >> 16);
tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
// tmpW32 = 1/den in Q29
tmpW32 = (tmp_hi * approx + (tmp_low * approx >> 15)) << 1;
// 1/den in hi and low format
tmp_hi = (int16_t)(tmpW32 >> 16);
tmp_low = (int16_t)((tmpW32 - ((int32_t)tmp_hi << 16)) >> 1);
// Store num in hi and low format
num_hi = (int16_t)(num >> 16);
num_low = (int16_t)((num - ((int32_t)num_hi << 16)) >> 1);
// num * (1/den) by 32 bit multiplication (result in Q28)
tmpW32 = num_hi * tmp_hi + (num_hi * tmp_low >> 15) +
(num_low * tmp_hi >> 15);
// Put result in Q31 (convert from Q28)
tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
return tmpW32;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/checks.h"
#include "rtc_base/sanitizer.h"
// TODO(Bjornv): Change the function parameter order to WebRTC code style.
// C version of WebRtcSpl_DownsampleFast() for generic platforms.
int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
size_t data_in_length,
int16_t* data_out,
size_t data_out_length,
const int16_t* __restrict coefficients,
size_t coefficients_length,
int factor,
size_t delay) {
int16_t* const original_data_out = data_out;
size_t i = 0;
size_t j = 0;
int32_t out_s32 = 0;
size_t endpos = delay + factor * (data_out_length - 1) + 1;
// Return error if any of the running conditions doesn't meet.
if (data_out_length == 0 || coefficients_length == 0
|| data_in_length < endpos) {
return -1;
}
rtc_MsanCheckInitialized(coefficients, sizeof(coefficients[0]),
coefficients_length);
for (i = delay; i < endpos; i += factor) {
out_s32 = 2048; // Round value, 0.5 in Q12.
for (j = 0; j < coefficients_length; j++) {
// Negative overflow is permitted here, because this is
// auto-regressive filters, and the state for each batch run is
// stored in the "negative" positions of the output vector.
rtc_MsanCheckInitialized(&data_in[(ptrdiff_t) i - (ptrdiff_t) j],
sizeof(data_in[0]), 1);
// out_s32 is in Q12 domain.
out_s32 += coefficients[j] * data_in[(ptrdiff_t) i - (ptrdiff_t) j];
}
out_s32 >>= 12; // Q0.
// Saturate and store the output.
*data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
}
RTC_DCHECK_EQ(original_data_out + data_out_length, data_out);
rtc_MsanCheckInitialized(original_data_out, sizeof(original_data_out[0]),
data_out_length);
return 0;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
#include <stdint.h>
// For ComplexFFT(), the maximum fft order is 10;
// WebRTC APM uses orders of only 7 and 8.
enum { kMaxFFTOrder = 10 };
struct RealFFT;
#ifdef __cplusplus
extern "C" {
#endif
struct RealFFT* WebRtcSpl_CreateRealFFT(int order);
void WebRtcSpl_FreeRealFFT(struct RealFFT* self);
// Compute an FFT for a real-valued signal of length of 2^order,
// where 1 < order <= MAX_FFT_ORDER. Transform length is determined by the
// specification structure, which must be initialized prior to calling the FFT
// function with WebRtcSpl_CreateRealFFT().
// The relationship between the input and output sequences can
// be expressed in terms of the DFT, i.e.:
// x[n] = (2^(-scalefactor)/N) . SUM[k=0,...,N-1] X[k].e^(jnk.2.pi/N)
// n=0,1,2,...N-1
// N=2^order.
// The conjugate-symmetric output sequence is represented using a CCS vector,
// which is of length N+2, and is organized as follows:
// Index: 0 1 2 3 4 5 . . . N-2 N-1 N N+1
// Component: R0 0 R1 I1 R2 I2 . . . R[N/2-1] I[N/2-1] R[N/2] 0
// where R[n] and I[n], respectively, denote the real and imaginary components
// for FFT bin 'n'. Bins are numbered from 0 to N/2, where N is the FFT length.
// Bin index 0 corresponds to the DC component, and bin index N/2 corresponds to
// the foldover frequency.
//
// Input Arguments:
// self - pointer to preallocated and initialized FFT specification structure.
// real_data_in - the input signal. For an ARM Neon platform, it must be
// aligned on a 32-byte boundary.
//
// Output Arguments:
// complex_data_out - the output complex signal with (2^order + 2) 16-bit
// elements. For an ARM Neon platform, it must be different
// from real_data_in, and aligned on a 32-byte boundary.
//
// Return Value:
// 0 - FFT calculation is successful.
// -1 - Error with bad arguments (null pointers).
int WebRtcSpl_RealForwardFFT(struct RealFFT* self,
const int16_t* real_data_in,
int16_t* complex_data_out);
// Compute the inverse FFT for a conjugate-symmetric input sequence of length of
// 2^order, where 1 < order <= MAX_FFT_ORDER. Transform length is determined by
// the specification structure, which must be initialized prior to calling the
// FFT function with WebRtcSpl_CreateRealFFT().
// For a transform of length M, the input sequence is represented using a packed
// CCS vector of length M+2, which is explained in the comments for
// WebRtcSpl_RealForwardFFTC above.
//
// Input Arguments:
// self - pointer to preallocated and initialized FFT specification structure.
// complex_data_in - the input complex signal with (2^order + 2) 16-bit
// elements. For an ARM Neon platform, it must be aligned on
// a 32-byte boundary.
//
// Output Arguments:
// real_data_out - the output real signal. For an ARM Neon platform, it must
// be different to complex_data_in, and aligned on a 32-byte
// boundary.
//
// Return Value:
// 0 or a positive number - a value that the elements in the `real_data_out`
// should be shifted left with in order to get
// correct physical values.
// -1 - Error with bad arguments (null pointers).
int WebRtcSpl_RealInverseFFT(struct RealFFT* self,
const int16_t* complex_data_in,
int16_t* real_data_out);
#ifdef __cplusplus
}
#endif
#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the implementation of functions
* WebRtcSpl_MaxAbsValueW16C()
* WebRtcSpl_MaxAbsValueW32C()
* WebRtcSpl_MaxValueW16C()
* WebRtcSpl_MaxValueW32C()
* WebRtcSpl_MinValueW16C()
* WebRtcSpl_MinValueW32C()
* WebRtcSpl_MaxAbsIndexW16()
* WebRtcSpl_MaxIndexW16()
* WebRtcSpl_MaxIndexW32()
* WebRtcSpl_MinIndexW16()
* WebRtcSpl_MinIndexW32()
*
*/
#include <stdlib.h>
#include <limits.h>
#include "rtc_base/checks.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
// TODO(bjorn/kma): Consolidate function pairs (e.g. combine
// WebRtcSpl_MaxAbsValueW16C and WebRtcSpl_MaxAbsIndexW16 into a single one.)
// TODO(kma): Move the next six functions into min_max_operations_c.c.
// Maximum absolute value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length) {
size_t i = 0;
int absolute = 0, maximum = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute = abs((int)vector[i]);
if (absolute > maximum) {
maximum = absolute;
}
}
// Guard the case for abs(-32768).
if (maximum > WEBRTC_SPL_WORD16_MAX) {
maximum = WEBRTC_SPL_WORD16_MAX;
}
return (int16_t)maximum;
}
// Maximum absolute value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length) {
// Use uint32_t for the local variables, to accommodate the return value
// of abs(0x80000000), which is 0x80000000.
uint32_t absolute = 0, maximum = 0;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute =
(vector[i] != INT_MIN) ? abs((int)vector[i]) : INT_MAX + (uint32_t)1;
if (absolute > maximum) {
maximum = absolute;
}
}
maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX);
return (int32_t)maximum;
}
// Maximum value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length) {
int16_t maximum = WEBRTC_SPL_WORD16_MIN;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum)
maximum = vector[i];
}
return maximum;
}
// Maximum value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length) {
int32_t maximum = WEBRTC_SPL_WORD32_MIN;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum)
maximum = vector[i];
}
return maximum;
}
// Minimum value of word16 vector. C version for generic platforms.
int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length) {
int16_t minimum = WEBRTC_SPL_WORD16_MAX;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum)
minimum = vector[i];
}
return minimum;
}
// Minimum value of word32 vector. C version for generic platforms.
int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length) {
int32_t minimum = WEBRTC_SPL_WORD32_MAX;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum)
minimum = vector[i];
}
return minimum;
}
// Index of maximum absolute value in a word16 vector.
size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length) {
// Use type int for local variables, to accomodate the value of abs(-32768).
size_t i = 0, index = 0;
int absolute = 0, maximum = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
absolute = abs((int)vector[i]);
if (absolute > maximum) {
maximum = absolute;
index = i;
}
}
return index;
}
int16_t WebRtcSpl_MaxAbsElementW16(const int16_t* vector, size_t length) {
int16_t min_val, max_val;
WebRtcSpl_MinMaxW16(vector, length, &min_val, &max_val);
if (min_val == max_val || min_val < -max_val) {
return min_val;
}
return max_val;
}
// Index of maximum value in a word16 vector.
size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length) {
size_t i = 0, index = 0;
int16_t maximum = WEBRTC_SPL_WORD16_MIN;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum) {
maximum = vector[i];
index = i;
}
}
return index;
}
// Index of maximum value in a word32 vector.
size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length) {
size_t i = 0, index = 0;
int32_t maximum = WEBRTC_SPL_WORD32_MIN;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] > maximum) {
maximum = vector[i];
index = i;
}
}
return index;
}
// Index of minimum value in a word16 vector.
size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length) {
size_t i = 0, index = 0;
int16_t minimum = WEBRTC_SPL_WORD16_MAX;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum) {
minimum = vector[i];
index = i;
}
}
return index;
}
// Index of minimum value in a word32 vector.
size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length) {
size_t i = 0, index = 0;
int32_t minimum = WEBRTC_SPL_WORD32_MAX;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum) {
minimum = vector[i];
index = i;
}
}
return index;
}
// Finds both the minimum and maximum elements in an array of 16-bit integers.
void WebRtcSpl_MinMaxW16(const int16_t* vector, size_t length,
int16_t* min_val, int16_t* max_val) {
#if defined(WEBRTC_HAS_NEON)
return WebRtcSpl_MinMaxW16Neon(vector, length, min_val, max_val);
#else
int16_t minimum = WEBRTC_SPL_WORD16_MAX;
int16_t maximum = WEBRTC_SPL_WORD16_MIN;
size_t i = 0;
RTC_DCHECK_GT(length, 0);
for (i = 0; i < length; i++) {
if (vector[i] < minimum)
minimum = vector[i];
if (vector[i] > maximum)
maximum = vector[i];
}
*min_val = minimum;
*max_val = maximum;
#endif
}

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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains implementations of the randomization functions
* WebRtcSpl_RandU()
* WebRtcSpl_RandN()
* WebRtcSpl_RandUArray()
*
* The description header can be found in signal_processing_library.h
*
*/
#include "common_audio/signal_processing/include/signal_processing_library.h"
static const uint32_t kMaxSeedUsed = 0x80000000;
static const int16_t kRandNTable[] = {
9178, -7260, 40, 10189, 4894, -3531, -13779, 14764,
-4008, -8884, -8990, 1008, 7368, 5184, 3251, -5817,
-9786, 5963, 1770, 8066, -7135, 10772, -2298, 1361,
6484, 2241, -8633, 792, 199, -3344, 6553, -10079,
-15040, 95, 11608, -12469, 14161, -4176, 2476, 6403,
13685, -16005, 6646, 2239, 10916, -3004, -602, -3141,
2142, 14144, -5829, 5305, 8209, 4713, 2697, -5112,
16092, -1210, -2891, -6631, -5360, -11878, -6781, -2739,
-6392, 536, 10923, 10872, 5059, -4748, -7770, 5477,
38, -1025, -2892, 1638, 6304, 14375, -11028, 1553,
-1565, 10762, -393, 4040, 5257, 12310, 6554, -4799,
4899, -6354, 1603, -1048, -2220, 8247, -186, -8944,
-12004, 2332, 4801, -4933, 6371, 131, 8614, -5927,
-8287, -22760, 4033, -15162, 3385, 3246, 3153, -5250,
3766, 784, 6494, -62, 3531, -1582, 15572, 662,
-3952, -330, -3196, 669, 7236, -2678, -6569, 23319,
-8645, -741, 14830, -15976, 4903, 315, -11342, 10311,
1858, -7777, 2145, 5436, 5677, -113, -10033, 826,
-1353, 17210, 7768, 986, -1471, 8291, -4982, 8207,
-14911, -6255, -2449, -11881, -7059, -11703, -4338, 8025,
7538, -2823, -12490, 9470, -1613, -2529, -10092, -7807,
9480, 6970, -12844, 5123, 3532, 4816, 4803, -8455,
-5045, 14032, -4378, -1643, 5756, -11041, -2732, -16618,
-6430, -18375, -3320, 6098, 5131, -4269, -8840, 2482,
-7048, 1547, -21890, -6505, -7414, -424, -11722, 7955,
1653, -17299, 1823, 473, -9232, 3337, 1111, 873,
4018, -8982, 9889, 3531, -11763, -3799, 7373, -4539,
3231, 7054, -8537, 7616, 6244, 16635, 447, -2915,
13967, 705, -2669, -1520, -1771, -16188, 5956, 5117,
6371, -9936, -1448, 2480, 5128, 7550, -8130, 5236,
8213, -6443, 7707, -1950, -13811, 7218, 7031, -3883,
67, 5731, -2874, 13480, -3743, 9298, -3280, 3552,
-4425, -18, -3785, -9988, -5357, 5477, -11794, 2117,
1416, -9935, 3376, 802, -5079, -8243, 12652, 66,
3653, -2368, 6781, -21895, -7227, 2487, 7839, -385,
6646, -7016, -4658, 5531, -1705, 834, 129, 3694,
-1343, 2238, -22640, -6417, -11139, 11301, -2945, -3494,
-5626, 185, -3615, -2041, -7972, -3106, -60, -23497,
-1566, 17064, 3519, 2518, 304, -6805, -10269, 2105,
1936, -426, -736, -8122, -1467, 4238, -6939, -13309,
360, 7402, -7970, 12576, 3287, 12194, -6289, -16006,
9171, 4042, -9193, 9123, -2512, 6388, -4734, -8739,
1028, -5406, -1696, 5889, -666, -4736, 4971, 3565,
9362, -6292, 3876, -3652, -19666, 7523, -4061, 391,
-11773, 7502, -3763, 4929, -9478, 13278, 2805, 4496,
7814, 16419, 12455, -14773, 2127, -2746, 3763, 4847,
3698, 6978, 4751, -6957, -3581, -45, 6252, 1513,
-4797, -7925, 11270, 16188, -2359, -5269, 9376, -10777,
7262, 20031, -6515, -2208, -5353, 8085, -1341, -1303,
7333, 5576, 3625, 5763, -7931, 9833, -3371, -10305,
6534, -13539, -9971, 997, 8464, -4064, -1495, 1857,
13624, 5458, 9490, -11086, -4524, 12022, -550, -198,
408, -8455, -7068, 10289, 9712, -3366, 9028, -7621,
-5243, 2362, 6909, 4672, -4933, -1799, 4709, -4563,
-62, -566, 1624, -7010, 14730, -17791, -3697, -2344,
-1741, 7099, -9509, -6855, -1989, 3495, -2289, 2031,
12784, 891, 14189, -3963, -5683, 421, -12575, 1724,
-12682, -5970, -8169, 3143, -1824, -5488, -5130, 8536,
12799, 794, 5738, 3459, -11689, -258, -3738, -3775,
-8742, 2333, 8312, -9383, 10331, 13119, 8398, 10644,
-19433, -6446, -16277, -11793, 16284, 9345, 15222, 15834,
2009, -7349, 130, -14547, 338, -5998, 3337, 21492,
2406, 7703, -951, 11196, -564, 3406, 2217, 4806,
2374, -5797, 11839, 8940, -11874, 18213, 2855, 10492
};
static uint32_t IncreaseSeed(uint32_t* seed) {
seed[0] = (seed[0] * ((int32_t)69069) + 1) & (kMaxSeedUsed - 1);
return seed[0];
}
int16_t WebRtcSpl_RandU(uint32_t* seed) {
return (int16_t)(IncreaseSeed(seed) >> 16);
}
int16_t WebRtcSpl_RandN(uint32_t* seed) {
return kRandNTable[IncreaseSeed(seed) >> 23];
}
// Creates an array of uniformly distributed variables.
int16_t WebRtcSpl_RandUArray(int16_t* vector,
int16_t vector_length,
uint32_t* seed) {
int i;
for (i = 0; i < vector_length; i++) {
vector[i] = WebRtcSpl_RandU(seed);
}
return vector_length;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "common_audio/signal_processing/include/real_fft.h"
#include <stdlib.h>
#include "common_audio/signal_processing/include/signal_processing_library.h"
struct RealFFT {
int order;
};
struct RealFFT* WebRtcSpl_CreateRealFFT(int order) {
struct RealFFT* self = NULL;
if (order > kMaxFFTOrder || order < 0) {
return NULL;
}
self = malloc(sizeof(struct RealFFT));
if (self == NULL) {
return NULL;
}
self->order = order;
return self;
}
void WebRtcSpl_FreeRealFFT(struct RealFFT* self) {
if (self != NULL) {
free(self);
}
}
// The C version FFT functions (i.e. WebRtcSpl_RealForwardFFT and
// WebRtcSpl_RealInverseFFT) are real-valued FFT wrappers for complex-valued
// FFT implementation in SPL.
int WebRtcSpl_RealForwardFFT(struct RealFFT* self,
const int16_t* real_data_in,
int16_t* complex_data_out) {
int i = 0;
int j = 0;
int result = 0;
int n = 1 << self->order;
// The complex-value FFT implementation needs a buffer to hold 2^order
// 16-bit COMPLEX numbers, for both time and frequency data.
int16_t complex_buffer[2 << kMaxFFTOrder];
// Insert zeros to the imaginary parts for complex forward FFT input.
for (i = 0, j = 0; i < n; i += 1, j += 2) {
complex_buffer[j] = real_data_in[i];
complex_buffer[j + 1] = 0;
};
WebRtcSpl_ComplexBitReverse(complex_buffer, self->order);
result = WebRtcSpl_ComplexFFT(complex_buffer, self->order, 1);
// For real FFT output, use only the first N + 2 elements from
// complex forward FFT.
memcpy(complex_data_out, complex_buffer, sizeof(int16_t) * (n + 2));
return result;
}
int WebRtcSpl_RealInverseFFT(struct RealFFT* self,
const int16_t* complex_data_in,
int16_t* real_data_out) {
int i = 0;
int j = 0;
int result = 0;
int n = 1 << self->order;
// Create the buffer specific to complex-valued FFT implementation.
int16_t complex_buffer[2 << kMaxFFTOrder];
// For n-point FFT, first copy the first n + 2 elements into complex
// FFT, then construct the remaining n - 2 elements by real FFT's
// conjugate-symmetric properties.
memcpy(complex_buffer, complex_data_in, sizeof(int16_t) * (n + 2));
for (i = n + 2; i < 2 * n; i += 2) {
complex_buffer[i] = complex_data_in[2 * n - i];
complex_buffer[i + 1] = -complex_data_in[2 * n - i + 1];
}
WebRtcSpl_ComplexBitReverse(complex_buffer, self->order);
result = WebRtcSpl_ComplexIFFT(complex_buffer, self->order, 1);
// Strip out the imaginary parts of the complex inverse FFT output.
for (i = 0, j = 0; i < n; i += 1, j += 2) {
real_data_out[i] = complex_buffer[j];
}
return result;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Some code came from common/rtcd.c in the WebM project.
#include "common_audio/signal_processing/include/signal_processing_library.h"
// TODO(bugs.webrtc.org/9553): These function pointers are useless. Refactor
// things so that we simply have a bunch of regular functions with different
// implementations for different platforms.
#if defined(WEBRTC_HAS_NEON)
const MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
const MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32Neon;
const MaxValueW16 WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16Neon;
const MaxValueW32 WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32Neon;
const MinValueW16 WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16Neon;
const MinValueW32 WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32Neon;
const CrossCorrelation WebRtcSpl_CrossCorrelation =
WebRtcSpl_CrossCorrelationNeon;
const DownsampleFast WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastNeon;
const ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
#elif defined(MIPS32_LE)
const MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16_mips;
const MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32 =
#ifdef MIPS_DSP_R1_LE
WebRtcSpl_MaxAbsValueW32_mips;
#else
WebRtcSpl_MaxAbsValueW32C;
#endif
const MaxValueW16 WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16_mips;
const MaxValueW32 WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32_mips;
const MinValueW16 WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16_mips;
const MinValueW32 WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32_mips;
const CrossCorrelation WebRtcSpl_CrossCorrelation =
WebRtcSpl_CrossCorrelation_mips;
const DownsampleFast WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFast_mips;
const ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound =
#ifdef MIPS_DSP_R1_LE
WebRtcSpl_ScaleAndAddVectorsWithRound_mips;
#else
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
#endif
#else
const MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
const MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
const MaxValueW16 WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16C;
const MaxValueW32 WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32C;
const MinValueW16 WebRtcSpl_MinValueW16 = WebRtcSpl_MinValueW16C;
const MinValueW32 WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32C;
const CrossCorrelation WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationC;
const DownsampleFast WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastC;
const ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound =
WebRtcSpl_ScaleAndAddVectorsWithRoundC;
#endif

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains implementations of the functions
* WebRtcSpl_VectorBitShiftW16()
* WebRtcSpl_VectorBitShiftW32()
* WebRtcSpl_VectorBitShiftW32ToW16()
* WebRtcSpl_ScaleVector()
* WebRtcSpl_ScaleVectorWithSat()
* WebRtcSpl_ScaleAndAddVectors()
* WebRtcSpl_ScaleAndAddVectorsWithRoundC()
*/
#include "common_audio/signal_processing/include/signal_processing_library.h"
void WebRtcSpl_VectorBitShiftW16(int16_t *res, size_t length,
const int16_t *in, int16_t right_shifts)
{
size_t i;
if (right_shifts > 0)
{
for (i = length; i > 0; i--)
{
(*res++) = ((*in++) >> right_shifts);
}
} else
{
for (i = length; i > 0; i--)
{
(*res++) = ((*in++) * (1 << (-right_shifts)));
}
}
}
void WebRtcSpl_VectorBitShiftW32(int32_t *out_vector,
size_t vector_length,
const int32_t *in_vector,
int16_t right_shifts)
{
size_t i;
if (right_shifts > 0)
{
for (i = vector_length; i > 0; i--)
{
(*out_vector++) = ((*in_vector++) >> right_shifts);
}
} else
{
for (i = vector_length; i > 0; i--)
{
(*out_vector++) = ((*in_vector++) << (-right_shifts));
}
}
}
void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, size_t length,
const int32_t* in, int right_shifts) {
size_t i;
int32_t tmp_w32;
if (right_shifts >= 0) {
for (i = length; i > 0; i--) {
tmp_w32 = (*in++) >> right_shifts;
(*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
}
} else {
int left_shifts = -right_shifts;
for (i = length; i > 0; i--) {
tmp_w32 = (*in++) << left_shifts;
(*out++) = WebRtcSpl_SatW32ToW16(tmp_w32);
}
}
}
void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector,
int16_t gain, size_t in_vector_length,
int16_t right_shifts)
{
// Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
size_t i;
const int16_t *inptr;
int16_t *outptr;
inptr = in_vector;
outptr = out_vector;
for (i = 0; i < in_vector_length; i++)
{
*outptr++ = (int16_t)((*inptr++ * gain) >> right_shifts);
}
}
void WebRtcSpl_ScaleVectorWithSat(const int16_t *in_vector, int16_t *out_vector,
int16_t gain, size_t in_vector_length,
int16_t right_shifts)
{
// Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
size_t i;
const int16_t *inptr;
int16_t *outptr;
inptr = in_vector;
outptr = out_vector;
for (i = 0; i < in_vector_length; i++) {
*outptr++ = WebRtcSpl_SatW32ToW16((*inptr++ * gain) >> right_shifts);
}
}
void WebRtcSpl_ScaleAndAddVectors(const int16_t *in1, int16_t gain1, int shift1,
const int16_t *in2, int16_t gain2, int shift2,
int16_t *out, size_t vector_length)
{
// Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
size_t i;
const int16_t *in1ptr;
const int16_t *in2ptr;
int16_t *outptr;
in1ptr = in1;
in2ptr = in2;
outptr = out;
for (i = 0; i < vector_length; i++)
{
*outptr++ = (int16_t)((gain1 * *in1ptr++) >> shift1) +
(int16_t)((gain2 * *in2ptr++) >> shift2);
}
}
// C version of WebRtcSpl_ScaleAndAddVectorsWithRound() for generic platforms.
int WebRtcSpl_ScaleAndAddVectorsWithRoundC(const int16_t* in_vector1,
int16_t in_vector1_scale,
const int16_t* in_vector2,
int16_t in_vector2_scale,
int right_shifts,
int16_t* out_vector,
size_t length) {
size_t i = 0;
int round_value = (1 << right_shifts) >> 1;
if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
length == 0 || right_shifts < 0) {
return -1;
}
for (i = 0; i < length; i++) {
out_vector[i] = (int16_t)((
in_vector1[i] * in_vector1_scale + in_vector2[i] * in_vector2_scale +
round_value) >> right_shifts);
}
return 0;
}

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Performs echo control (suppression) with fft routines in fixed-point.
#ifndef MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
#define MODULES_AUDIO_PROCESSING_AECM_AECM_CORE_H_
extern "C" {
#include "common_audio/ring_buffer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
}
#include "modules/audio_processing/aecm/aecm_defines.h"
struct RealFFT;
namespace webrtc {
#ifdef _MSC_VER // visual c++
#define ALIGN8_BEG __declspec(align(8))
#define ALIGN8_END
#else // gcc or icc
#define ALIGN8_BEG
#define ALIGN8_END __attribute__((aligned(8)))
#endif
typedef struct {
int16_t real;
int16_t imag;
} ComplexInt16;
typedef struct {
int farBufWritePos;
int farBufReadPos;
int knownDelay;
int lastKnownDelay;
int firstVAD; // Parameter to control poorly initialized channels
RingBuffer* farFrameBuf;
RingBuffer* nearNoisyFrameBuf;
RingBuffer* nearCleanFrameBuf;
RingBuffer* outFrameBuf;
int16_t farBuf[FAR_BUF_LEN];
int16_t mult;
uint32_t seed;
// Delay estimation variables
void* delay_estimator_farend;
void* delay_estimator;
uint16_t currentDelay;
// Far end history variables
// TODO(bjornv): Replace `far_history` with ring_buffer.
uint16_t far_history[PART_LEN1 * MAX_DELAY];
int far_history_pos;
int far_q_domains[MAX_DELAY];
int16_t nlpFlag;
int16_t fixedDelay;
uint32_t totCount;
int16_t dfaCleanQDomain;
int16_t dfaCleanQDomainOld;
int16_t dfaNoisyQDomain;
int16_t dfaNoisyQDomainOld;
int16_t nearLogEnergy[MAX_BUF_LEN];
int16_t farLogEnergy;
int16_t echoAdaptLogEnergy[MAX_BUF_LEN];
int16_t echoStoredLogEnergy[MAX_BUF_LEN];
// The extra 16 or 32 bytes in the following buffers are for alignment based
// Neon code.
// It's designed this way since the current GCC compiler can't align a
// buffer in 16 or 32 byte boundaries properly.
int16_t channelStored_buf[PART_LEN1 + 8];
int16_t channelAdapt16_buf[PART_LEN1 + 8];
int32_t channelAdapt32_buf[PART_LEN1 + 8];
int16_t xBuf_buf[PART_LEN2 + 16]; // farend
int16_t dBufClean_buf[PART_LEN2 + 16]; // nearend
int16_t dBufNoisy_buf[PART_LEN2 + 16]; // nearend
int16_t outBuf_buf[PART_LEN + 8];
// Pointers to the above buffers
int16_t* channelStored;
int16_t* channelAdapt16;
int32_t* channelAdapt32;
int16_t* xBuf;
int16_t* dBufClean;
int16_t* dBufNoisy;
int16_t* outBuf;
int32_t echoFilt[PART_LEN1];
int16_t nearFilt[PART_LEN1];
int32_t noiseEst[PART_LEN1];
int noiseEstTooLowCtr[PART_LEN1];
int noiseEstTooHighCtr[PART_LEN1];
int16_t noiseEstCtr;
int16_t cngMode;
int32_t mseAdaptOld;
int32_t mseStoredOld;
int32_t mseThreshold;
int16_t farEnergyMin;
int16_t farEnergyMax;
int16_t farEnergyMaxMin;
int16_t farEnergyVAD;
int16_t farEnergyMSE;
int currentVADValue;
int16_t vadUpdateCount;
int16_t startupState;
int16_t mseChannelCount;
int16_t supGain;
int16_t supGainOld;
int16_t supGainErrParamA;
int16_t supGainErrParamD;
int16_t supGainErrParamDiffAB;
int16_t supGainErrParamDiffBD;
struct RealFFT* real_fft;
#ifdef AEC_DEBUG
FILE* farFile;
FILE* nearFile;
FILE* outFile;
#endif
} AecmCore;
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_CreateCore()
//
// Allocates the memory needed by the AECM. The memory needs to be
// initialized separately using the WebRtcAecm_InitCore() function.
// Returns a pointer to the instance and a nullptr at failure.
AecmCore* WebRtcAecm_CreateCore();
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_InitCore(...)
//
// This function initializes the AECM instant created with
// WebRtcAecm_CreateCore()
// Input:
// - aecm : Pointer to the AECM instance
// - samplingFreq : Sampling Frequency
//
// Output:
// - aecm : Initialized instance
//
// Return value : 0 - Ok
// -1 - Error
//
int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_FreeCore(...)
//
// This function releases the memory allocated by WebRtcAecm_CreateCore()
// Input:
// - aecm : Pointer to the AECM instance
//
void WebRtcAecm_FreeCore(AecmCore* aecm);
int WebRtcAecm_Control(AecmCore* aecm, int delay, int nlpFlag);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_InitEchoPathCore(...)
//
// This function resets the echo channel adaptation with the specified channel.
// Input:
// - aecm : Pointer to the AECM instance
// - echo_path : Pointer to the data that should initialize the echo
// path
//
// Output:
// - aecm : Initialized instance
//
void WebRtcAecm_InitEchoPathCore(AecmCore* aecm, const int16_t* echo_path);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_ProcessFrame(...)
//
// This function processes frames and sends blocks to
// WebRtcAecm_ProcessBlock(...)
//
// Inputs:
// - aecm : Pointer to the AECM instance
// - farend : In buffer containing one frame of echo signal
// - nearendNoisy : In buffer containing one frame of nearend+echo signal
// without NS
// - nearendClean : In buffer containing one frame of nearend+echo signal
// with NS
//
// Output:
// - out : Out buffer, one frame of nearend signal :
//
//
int WebRtcAecm_ProcessFrame(AecmCore* aecm,
const int16_t* farend,
const int16_t* nearendNoisy,
const int16_t* nearendClean,
int16_t* out);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_ProcessBlock(...)
//
// This function is called for every block within one frame
// This function is called by WebRtcAecm_ProcessFrame(...)
//
// Inputs:
// - aecm : Pointer to the AECM instance
// - farend : In buffer containing one block of echo signal
// - nearendNoisy : In buffer containing one frame of nearend+echo signal
// without NS
// - nearendClean : In buffer containing one frame of nearend+echo signal
// with NS
//
// Output:
// - out : Out buffer, one block of nearend signal :
//
//
int WebRtcAecm_ProcessBlock(AecmCore* aecm,
const int16_t* farend,
const int16_t* nearendNoisy,
const int16_t* noisyClean,
int16_t* out);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_BufferFarFrame()
//
// Inserts a frame of data into farend buffer.
//
// Inputs:
// - aecm : Pointer to the AECM instance
// - farend : In buffer containing one frame of farend signal
// - farLen : Length of frame
//
void WebRtcAecm_BufferFarFrame(AecmCore* const aecm,
const int16_t* const farend,
int farLen);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_FetchFarFrame()
//
// Read the farend buffer to account for known delay
//
// Inputs:
// - aecm : Pointer to the AECM instance
// - farend : In buffer containing one frame of farend signal
// - farLen : Length of frame
// - knownDelay : known delay
//
void WebRtcAecm_FetchFarFrame(AecmCore* const aecm,
int16_t* const farend,
int farLen,
int knownDelay);
// All the functions below are intended to be private
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_UpdateFarHistory()
//
// Moves the pointer to the next entry and inserts `far_spectrum` and
// corresponding Q-domain in its buffer.
//
// Inputs:
// - self : Pointer to the delay estimation instance
// - far_spectrum : Pointer to the far end spectrum
// - far_q : Q-domain of far end spectrum
//
void WebRtcAecm_UpdateFarHistory(AecmCore* self,
uint16_t* far_spectrum,
int far_q);
////////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_AlignedFarend()
//
// Returns a pointer to the far end spectrum aligned to current near end
// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
// called before AlignedFarend(...). Otherwise, you get the pointer to the
// previous frame. The memory is only valid until the next call of
// WebRtc_DelayEstimatorProcessFix(...).
//
// Inputs:
// - self : Pointer to the AECM instance.
// - delay : Current delay estimate.
//
// Output:
// - far_q : The Q-domain of the aligned far end spectrum
//
// Return value:
// - far_spectrum : Pointer to the aligned far end spectrum
// NULL - Error
//
const uint16_t* WebRtcAecm_AlignedFarend(AecmCore* self, int* far_q, int delay);
///////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_CalcSuppressionGain()
//
// This function calculates the suppression gain that is used in the
// Wiener filter.
//
// Inputs:
// - aecm : Pointer to the AECM instance.
//
// Return value:
// - supGain : Suppression gain with which to scale the noise
// level (Q14).
//
int16_t WebRtcAecm_CalcSuppressionGain(AecmCore* const aecm);
///////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_CalcEnergies()
//
// This function calculates the log of energies for nearend, farend and
// estimated echoes. There is also an update of energy decision levels,
// i.e. internal VAD.
//
// Inputs:
// - aecm : Pointer to the AECM instance.
// - far_spectrum : Pointer to farend spectrum.
// - far_q : Q-domain of farend spectrum.
// - nearEner : Near end energy for current block in
// Q(aecm->dfaQDomain).
//
// Output:
// - echoEst : Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
//
void WebRtcAecm_CalcEnergies(AecmCore* aecm,
const uint16_t* far_spectrum,
int16_t far_q,
uint32_t nearEner,
int32_t* echoEst);
///////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_CalcStepSize()
//
// This function calculates the step size used in channel estimation
//
// Inputs:
// - aecm : Pointer to the AECM instance.
//
// Return value:
// - mu : Stepsize in log2(), i.e. number of shifts.
//
int16_t WebRtcAecm_CalcStepSize(AecmCore* const aecm);
///////////////////////////////////////////////////////////////////////////////
// WebRtcAecm_UpdateChannel(...)
//
// This function performs channel estimation.
// NLMS and decision on channel storage.
//
// Inputs:
// - aecm : Pointer to the AECM instance.
// - far_spectrum : Absolute value of the farend signal in Q(far_q)
// - far_q : Q-domain of the farend signal
// - dfa : Absolute value of the nearend signal
// (Q[aecm->dfaQDomain])
// - mu : NLMS step size.
// Input/Output:
// - echoEst : Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
//
void WebRtcAecm_UpdateChannel(AecmCore* aecm,
const uint16_t* far_spectrum,
int16_t far_q,
const uint16_t* const dfa,
int16_t mu,
int32_t* echoEst);
extern const int16_t WebRtcAecm_kCosTable[];
extern const int16_t WebRtcAecm_kSinTable[];
///////////////////////////////////////////////////////////////////////////////
// Some function pointers, for internal functions shared by ARM NEON and
// generic C code.
//
typedef void (*CalcLinearEnergies)(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echoEst,
uint32_t* far_energy,
uint32_t* echo_energy_adapt,
uint32_t* echo_energy_stored);
extern CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
typedef void (*StoreAdaptiveChannel)(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est);
extern StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
typedef void (*ResetAdaptiveChannel)(AecmCore* aecm);
extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
// For the above function pointers, functions for generic platforms are declared
// and defined as static in file aecm_core.c, while those for ARM Neon platforms
// are declared below and defined in file aecm_core_neon.c.
#if defined(WEBRTC_HAS_NEON)
void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est,
uint32_t* far_energy,
uint32_t* echo_energy_adapt,
uint32_t* echo_energy_stored);
void WebRtcAecm_StoreAdaptiveChannelNeon(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est);
void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore* aecm);
#endif
#if defined(MIPS32_LE)
void WebRtcAecm_CalcLinearEnergies_mips(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est,
uint32_t* far_energy,
uint32_t* echo_energy_adapt,
uint32_t* echo_energy_stored);
#if defined(MIPS_DSP_R1_LE)
void WebRtcAecm_StoreAdaptiveChannel_mips(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est);
void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore* aecm);
#endif
#endif
} // namespace webrtc
#endif

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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stddef.h>
#include <stdlib.h>
#include "modules/audio_processing/aecm/aecm_core.h"
extern "C" {
#include "common_audio/ring_buffer.h"
#include "common_audio/signal_processing/include/real_fft.h"
}
#include "modules/audio_processing/aecm/echo_control_mobile.h"
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
extern "C" {
#include "system_wrappers/include/cpu_features_wrapper.h"
}
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/sanitizer.h"
namespace webrtc {
namespace {
// Square root of Hanning window in Q14.
static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172, 3562, 3951,
4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019,
8364, 8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514,
11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189,
14384, 14571, 14749, 14918, 15079, 15231, 15373, 15506, 15631, 15746, 15851,
15947, 16034, 16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384};
#ifdef AECM_WITH_ABS_APPROX
// Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
static const uint16_t kAlpha1 = 32584;
// Q15 beta = 0.12967166976970 const Factor for magnitude approximation
static const uint16_t kBeta1 = 4249;
// Q15 alpha = 0.94234827210087 const Factor for magnitude approximation
static const uint16_t kAlpha2 = 30879;
// Q15 beta = 0.33787806009150 const Factor for magnitude approximation
static const uint16_t kBeta2 = 11072;
// Q15 alpha = 0.82247698684306 const Factor for magnitude approximation
static const uint16_t kAlpha3 = 26951;
// Q15 beta = 0.57762063060713 const Factor for magnitude approximation
static const uint16_t kBeta3 = 18927;
#endif
static const int16_t kNoiseEstQDomain = 15;
static const int16_t kNoiseEstIncCount = 5;
static void ComfortNoise(AecmCore* aecm,
const uint16_t* dfa,
ComplexInt16* out,
const int16_t* lambda) {
int16_t i;
int16_t tmp16;
int32_t tmp32;
int16_t randW16[PART_LEN];
int16_t uReal[PART_LEN1];
int16_t uImag[PART_LEN1];
int32_t outLShift32;
int16_t noiseRShift16[PART_LEN1];
int16_t shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
int16_t minTrackShift;
RTC_DCHECK_GE(shiftFromNearToNoise, 0);
RTC_DCHECK_LT(shiftFromNearToNoise, 16);
if (aecm->noiseEstCtr < 100) {
// Track the minimum more quickly initially.
aecm->noiseEstCtr++;
minTrackShift = 6;
} else {
minTrackShift = 9;
}
// Estimate noise power.
for (i = 0; i < PART_LEN1; i++) {
// Shift to the noise domain.
tmp32 = (int32_t)dfa[i];
outLShift32 = tmp32 << shiftFromNearToNoise;
if (outLShift32 < aecm->noiseEst[i]) {
// Reset "too low" counter
aecm->noiseEstTooLowCtr[i] = 0;
// Track the minimum.
if (aecm->noiseEst[i] < (1 << minTrackShift)) {
// For small values, decrease noiseEst[i] every
// `kNoiseEstIncCount` block. The regular approach below can not
// go further down due to truncation.
aecm->noiseEstTooHighCtr[i]++;
if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
aecm->noiseEst[i]--;
aecm->noiseEstTooHighCtr[i] = 0; // Reset the counter
}
} else {
aecm->noiseEst[i] -=
((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
}
} else {
// Reset "too high" counter
aecm->noiseEstTooHighCtr[i] = 0;
// Ramp slowly upwards until we hit the minimum again.
if ((aecm->noiseEst[i] >> 19) > 0) {
// Avoid overflow.
// Multiplication with 2049 will cause wrap around. Scale
// down first and then multiply
aecm->noiseEst[i] >>= 11;
aecm->noiseEst[i] *= 2049;
} else if ((aecm->noiseEst[i] >> 11) > 0) {
// Large enough for relative increase
aecm->noiseEst[i] *= 2049;
aecm->noiseEst[i] >>= 11;
} else {
// Make incremental increases based on size every
// `kNoiseEstIncCount` block
aecm->noiseEstTooLowCtr[i]++;
if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
}
}
}
}
for (i = 0; i < PART_LEN1; i++) {
tmp32 = aecm->noiseEst[i] >> shiftFromNearToNoise;
if (tmp32 > 32767) {
tmp32 = 32767;
aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
}
noiseRShift16[i] = (int16_t)tmp32;
tmp16 = ONE_Q14 - lambda[i];
noiseRShift16[i] = (int16_t)((tmp16 * noiseRShift16[i]) >> 14);
}
// Generate a uniform random array on [0 2^15-1].
WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
// Generate noise according to estimated energy.
uReal[0] = 0; // Reject LF noise.
uImag[0] = 0;
for (i = 1; i < PART_LEN1; i++) {
// Get a random index for the cos and sin tables over [0 359].
tmp16 = (int16_t)((359 * randW16[i - 1]) >> 15);
// Tables are in Q13.
uReal[i] =
(int16_t)((noiseRShift16[i] * WebRtcAecm_kCosTable[tmp16]) >> 13);
uImag[i] =
(int16_t)((-noiseRShift16[i] * WebRtcAecm_kSinTable[tmp16]) >> 13);
}
uImag[PART_LEN] = 0;
for (i = 0; i < PART_LEN1; i++) {
out[i].real = WebRtcSpl_AddSatW16(out[i].real, uReal[i]);
out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
}
}
static void WindowAndFFT(AecmCore* aecm,
int16_t* fft,
const int16_t* time_signal,
ComplexInt16* freq_signal,
int time_signal_scaling) {
int i = 0;
// FFT of signal
for (i = 0; i < PART_LEN; i++) {
// Window time domain signal and insert into real part of
// transformation array `fft`
int16_t scaled_time_signal = time_signal[i] * (1 << time_signal_scaling);
fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14);
scaled_time_signal = time_signal[i + PART_LEN] * (1 << time_signal_scaling);
fft[PART_LEN + i] = (int16_t)((scaled_time_signal *
WebRtcAecm_kSqrtHanning[PART_LEN - i]) >>
14);
}
// Do forward FFT, then take only the first PART_LEN complex samples,
// and change signs of the imaginary parts.
WebRtcSpl_RealForwardFFT(aecm->real_fft, fft, (int16_t*)freq_signal);
for (i = 0; i < PART_LEN; i++) {
freq_signal[i].imag = -freq_signal[i].imag;
}
}
static void InverseFFTAndWindow(AecmCore* aecm,
int16_t* fft,
ComplexInt16* efw,
int16_t* output,
const int16_t* nearendClean) {
int i, j, outCFFT;
int32_t tmp32no1;
// Reuse `efw` for the inverse FFT output after transferring
// the contents to `fft`.
int16_t* ifft_out = (int16_t*)efw;
// Synthesis
for (i = 1, j = 2; i < PART_LEN; i += 1, j += 2) {
fft[j] = efw[i].real;
fft[j + 1] = -efw[i].imag;
}
fft[0] = efw[0].real;
fft[1] = -efw[0].imag;
fft[PART_LEN2] = efw[PART_LEN].real;
fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
// Inverse FFT. Keep outCFFT to scale the samples in the next block.
outCFFT = WebRtcSpl_RealInverseFFT(aecm->real_fft, fft, ifft_out);
for (i = 0; i < PART_LEN; i++) {
ifft_out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
ifft_out[i], WebRtcAecm_kSqrtHanning[i], 14);
tmp32no1 = WEBRTC_SPL_SHIFT_W32((int32_t)ifft_out[i],
outCFFT - aecm->dfaCleanQDomain);
output[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
tmp32no1 + aecm->outBuf[i],
WEBRTC_SPL_WORD16_MIN);
tmp32no1 =
(ifft_out[PART_LEN + i] * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14;
tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
aecm->outBuf[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
WEBRTC_SPL_WORD16_MIN);
}
// Copy the current block to the old position
// (aecm->outBuf is shifted elsewhere)
memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN);
memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN,
sizeof(int16_t) * PART_LEN);
if (nearendClean != NULL) {
memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN,
sizeof(int16_t) * PART_LEN);
}
}
// Transforms a time domain signal into the frequency domain, outputting the
// complex valued signal, absolute value and sum of absolute values.
//
// time_signal [in] Pointer to time domain signal
// freq_signal_real [out] Pointer to real part of frequency domain array
// freq_signal_imag [out] Pointer to imaginary part of frequency domain
// array
// freq_signal_abs [out] Pointer to absolute value of frequency domain
// array
// freq_signal_sum_abs [out] Pointer to the sum of all absolute values in
// the frequency domain array
// return value The Q-domain of current frequency values
//
static int TimeToFrequencyDomain(AecmCore* aecm,
const int16_t* time_signal,
ComplexInt16* freq_signal,
uint16_t* freq_signal_abs,
uint32_t* freq_signal_sum_abs) {
int i = 0;
int time_signal_scaling = 0;
int32_t tmp32no1 = 0;
int32_t tmp32no2 = 0;
// In fft_buf, +16 for 32-byte alignment.
int16_t fft_buf[PART_LEN4 + 16];
int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
int16_t tmp16no1;
#ifndef WEBRTC_ARCH_ARM_V7
int16_t tmp16no2;
#endif
#ifdef AECM_WITH_ABS_APPROX
int16_t max_value = 0;
int16_t min_value = 0;
uint16_t alpha = 0;
uint16_t beta = 0;
#endif
#ifdef AECM_DYNAMIC_Q
tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
#endif
WindowAndFFT(aecm, fft, time_signal, freq_signal, time_signal_scaling);
// Extract imaginary and real part, calculate the magnitude for
// all frequency bins
freq_signal[0].imag = 0;
freq_signal[PART_LEN].imag = 0;
freq_signal_abs[0] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[0].real);
freq_signal_abs[PART_LEN] =
(uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[PART_LEN].real);
(*freq_signal_sum_abs) =
(uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
for (i = 1; i < PART_LEN; i++) {
if (freq_signal[i].real == 0) {
freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
} else if (freq_signal[i].imag == 0) {
freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
} else {
// Approximation for magnitude of complex fft output
// magn = sqrt(real^2 + imag^2)
// magn ~= alpha * max(`imag`,`real`) + beta * min(`imag`,`real`)
//
// The parameters alpha and beta are stored in Q15
#ifdef AECM_WITH_ABS_APPROX
tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
if (tmp16no1 > tmp16no2) {
max_value = tmp16no1;
min_value = tmp16no2;
} else {
max_value = tmp16no2;
min_value = tmp16no1;
}
// Magnitude in Q(-6)
if ((max_value >> 2) > min_value) {
alpha = kAlpha1;
beta = kBeta1;
} else if ((max_value >> 1) > min_value) {
alpha = kAlpha2;
beta = kBeta2;
} else {
alpha = kAlpha3;
beta = kBeta3;
}
tmp16no1 = (int16_t)((max_value * alpha) >> 15);
tmp16no2 = (int16_t)((min_value * beta) >> 15);
freq_signal_abs[i] = (uint16_t)tmp16no1 + (uint16_t)tmp16no2;
#else
#ifdef WEBRTC_ARCH_ARM_V7
__asm __volatile(
"smulbb %[tmp32no1], %[real], %[real]\n\t"
"smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
: [tmp32no1] "+&r"(tmp32no1), [tmp32no2] "=r"(tmp32no2)
: [real] "r"(freq_signal[i].real), [imag] "r"(freq_signal[i].imag));
#else
tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
tmp32no1 = tmp16no1 * tmp16no1;
tmp32no2 = tmp16no2 * tmp16no2;
tmp32no2 = WebRtcSpl_AddSatW32(tmp32no1, tmp32no2);
#endif // WEBRTC_ARCH_ARM_V7
tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
freq_signal_abs[i] = (uint16_t)tmp32no1;
#endif // AECM_WITH_ABS_APPROX
}
(*freq_signal_sum_abs) += (uint32_t)freq_signal_abs[i];
}
return time_signal_scaling;
}
} // namespace
int RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/8200
WebRtcAecm_ProcessBlock(AecmCore* aecm,
const int16_t* farend,
const int16_t* nearendNoisy,
const int16_t* nearendClean,
int16_t* output) {
int i;
uint32_t xfaSum;
uint32_t dfaNoisySum;
uint32_t dfaCleanSum;
uint32_t echoEst32Gained;
uint32_t tmpU32;
int32_t tmp32no1;
uint16_t xfa[PART_LEN1];
uint16_t dfaNoisy[PART_LEN1];
uint16_t dfaClean[PART_LEN1];
uint16_t* ptrDfaClean = dfaClean;
const uint16_t* far_spectrum_ptr = NULL;
// 32 byte aligned buffers (with +8 or +16).
// TODO(kma): define fft with ComplexInt16.
int16_t fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
int32_t echoEst32_buf[PART_LEN1 + 8];
int32_t dfw_buf[PART_LEN2 + 8];
int32_t efw_buf[PART_LEN2 + 8];
int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
int32_t* echoEst32 = (int32_t*)(((uintptr_t)echoEst32_buf + 31) & ~31);
ComplexInt16* dfw = (ComplexInt16*)(((uintptr_t)dfw_buf + 31) & ~31);
ComplexInt16* efw = (ComplexInt16*)(((uintptr_t)efw_buf + 31) & ~31);
int16_t hnl[PART_LEN1];
int16_t numPosCoef = 0;
int16_t nlpGain = ONE_Q14;
int delay;
int16_t tmp16no1;
int16_t tmp16no2;
int16_t mu;
int16_t supGain;
int16_t zeros32, zeros16;
int16_t zerosDBufNoisy, zerosDBufClean, zerosXBuf;
int far_q;
int16_t resolutionDiff, qDomainDiff, dfa_clean_q_domain_diff;
const int kMinPrefBand = 4;
const int kMaxPrefBand = 24;
int32_t avgHnl32 = 0;
// Determine startup state. There are three states:
// (0) the first CONV_LEN blocks
// (1) another CONV_LEN blocks
// (2) the rest
if (aecm->startupState < 2) {
aecm->startupState =
(aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
}
// END: Determine startup state
// Buffer near and far end signals
memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN);
memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN);
if (nearendClean != NULL) {
memcpy(aecm->dBufClean + PART_LEN, nearendClean,
sizeof(int16_t) * PART_LEN);
}
// Transform far end signal from time domain to frequency domain.
far_q = TimeToFrequencyDomain(aecm, aecm->xBuf, dfw, xfa, &xfaSum);
// Transform noisy near end signal from time domain to frequency domain.
zerosDBufNoisy =
TimeToFrequencyDomain(aecm, aecm->dBufNoisy, dfw, dfaNoisy, &dfaNoisySum);
aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy;
if (nearendClean == NULL) {
ptrDfaClean = dfaNoisy;
aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
dfaCleanSum = dfaNoisySum;
} else {
// Transform clean near end signal from time domain to frequency domain.
zerosDBufClean = TimeToFrequencyDomain(aecm, aecm->dBufClean, dfw, dfaClean,
&dfaCleanSum);
aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
aecm->dfaCleanQDomain = (int16_t)zerosDBufClean;
}
// Get the delay
// Save far-end history and estimate delay
WebRtcAecm_UpdateFarHistory(aecm, xfa, far_q);
if (WebRtc_AddFarSpectrumFix(aecm->delay_estimator_farend, xfa, PART_LEN1,
far_q) == -1) {
return -1;
}
delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator, dfaNoisy,
PART_LEN1, zerosDBufNoisy);
if (delay == -1) {
return -1;
} else if (delay == -2) {
// If the delay is unknown, we assume zero.
// NOTE: this will have to be adjusted if we ever add lookahead.
delay = 0;
}
if (aecm->fixedDelay >= 0) {
// Use fixed delay
delay = aecm->fixedDelay;
}
// Get aligned far end spectrum
far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay);
zerosXBuf = (int16_t)far_q;
if (far_spectrum_ptr == NULL) {
return -1;
}
// Calculate log(energy) and update energy threshold levels
WebRtcAecm_CalcEnergies(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisySum,
echoEst32);
// Calculate stepsize
mu = WebRtcAecm_CalcStepSize(aecm);
// Update counters
aecm->totCount++;
// This is the channel estimation algorithm.
// It is base on NLMS but has a variable step length,
// which was calculated above.
WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu,
echoEst32);
supGain = WebRtcAecm_CalcSuppressionGain(aecm);
// Calculate Wiener filter hnl[]
for (i = 0; i < PART_LEN1; i++) {
// Far end signal through channel estimate in Q8
// How much can we shift right to preserve resolution
tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
aecm->echoFilt[i] +=
rtc::dchecked_cast<int32_t>((int64_t{tmp32no1} * 50) >> 8);
zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
zeros16 = WebRtcSpl_NormW16(supGain) + 1;
if (zeros32 + zeros16 > 16) {
// Multiplication is safe
// Result in
// Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+
// aecm->xfaQDomainBuf[diff])
echoEst32Gained =
WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i], (uint16_t)supGain);
resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
} else {
tmp16no1 = 17 - zeros32 - zeros16;
resolutionDiff =
14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
if (zeros32 > tmp16no1) {
echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
supGain >> tmp16no1);
} else {
// Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
echoEst32Gained = (aecm->echoFilt[i] >> tmp16no1) * supGain;
}
}
zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
RTC_DCHECK_GE(zeros16, 0); // `zeros16` is a norm, hence non-negative.
dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
tmp16no1 = aecm->nearFilt[i] * (1 << zeros16);
qDomainDiff = zeros16 - dfa_clean_q_domain_diff;
tmp16no2 = ptrDfaClean[i] >> -qDomainDiff;
} else {
tmp16no1 = dfa_clean_q_domain_diff < 0
? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
: aecm->nearFilt[i] * (1 << dfa_clean_q_domain_diff);
qDomainDiff = 0;
tmp16no2 = ptrDfaClean[i];
}
tmp32no1 = (int32_t)(tmp16no2 - tmp16no1);
tmp16no2 = (int16_t)(tmp32no1 >> 4);
tmp16no2 += tmp16no1;
zeros16 = WebRtcSpl_NormW16(tmp16no2);
if ((tmp16no2) & (-qDomainDiff > zeros16)) {
aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
} else {
aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 * (1 << -qDomainDiff)
: tmp16no2 >> qDomainDiff;
}
// Wiener filter coefficients, resulting hnl in Q14
if (echoEst32Gained == 0) {
hnl[i] = ONE_Q14;
} else if (aecm->nearFilt[i] == 0) {
hnl[i] = 0;
} else {
// Multiply the suppression gain
// Rounding
echoEst32Gained += (uint32_t)(aecm->nearFilt[i] >> 1);
tmpU32 =
WebRtcSpl_DivU32U16(echoEst32Gained, (uint16_t)aecm->nearFilt[i]);
// Current resolution is
// Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN- max(0,17-zeros16- zeros32))
// Make sure we are in Q14
tmp32no1 = (int32_t)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
if (tmp32no1 > ONE_Q14) {
hnl[i] = 0;
} else if (tmp32no1 < 0) {
hnl[i] = ONE_Q14;
} else {
// 1-echoEst/dfa
hnl[i] = ONE_Q14 - (int16_t)tmp32no1;
if (hnl[i] < 0) {
hnl[i] = 0;
}
}
}
if (hnl[i]) {
numPosCoef++;
}
}
// Only in wideband. Prevent the gain in upper band from being larger than
// in lower band.
if (aecm->mult == 2) {
// TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
// speech distortion in double-talk.
for (i = 0; i < PART_LEN1; i++) {
hnl[i] = (int16_t)((hnl[i] * hnl[i]) >> 14);
}
for (i = kMinPrefBand; i <= kMaxPrefBand; i++) {
avgHnl32 += (int32_t)hnl[i];
}
RTC_DCHECK_GT(kMaxPrefBand - kMinPrefBand + 1, 0);
avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
for (i = kMaxPrefBand; i < PART_LEN1; i++) {
if (hnl[i] > (int16_t)avgHnl32) {
hnl[i] = (int16_t)avgHnl32;
}
}
}
// Calculate NLP gain, result is in Q14
if (aecm->nlpFlag) {
for (i = 0; i < PART_LEN1; i++) {
// Truncate values close to zero and one.
if (hnl[i] > NLP_COMP_HIGH) {
hnl[i] = ONE_Q14;
} else if (hnl[i] < NLP_COMP_LOW) {
hnl[i] = 0;
}
// Remove outliers
if (numPosCoef < 3) {
nlpGain = 0;
} else {
nlpGain = ONE_Q14;
}
// NLP
if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14)) {
hnl[i] = ONE_Q14;
} else {
hnl[i] = (int16_t)((hnl[i] * nlpGain) >> 14);
}
// multiply with Wiener coefficients
efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
hnl[i], 14));
efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
hnl[i], 14));
}
} else {
// multiply with Wiener coefficients
for (i = 0; i < PART_LEN1; i++) {
efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
hnl[i], 14));
efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
hnl[i], 14));
}
}
if (aecm->cngMode == AecmTrue) {
ComfortNoise(aecm, ptrDfaClean, efw, hnl);
}
InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
return 0;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <arm_neon.h>
#include "common_audio/signal_processing/include/real_fft.h"
#include "modules/audio_processing/aecm/aecm_core.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
// TODO(kma): Re-write the corresponding assembly file, the offset
// generating script and makefile, to replace these C functions.
static inline void AddLanes(uint32_t* ptr, uint32x4_t v) {
#if defined(WEBRTC_ARCH_ARM64)
*(ptr) = vaddvq_u32(v);
#else
uint32x2_t tmp_v;
tmp_v = vadd_u32(vget_low_u32(v), vget_high_u32(v));
tmp_v = vpadd_u32(tmp_v, tmp_v);
*(ptr) = vget_lane_u32(tmp_v, 0);
#endif
}
} // namespace
void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est,
uint32_t* far_energy,
uint32_t* echo_energy_adapt,
uint32_t* echo_energy_stored) {
int16_t* start_stored_p = aecm->channelStored;
int16_t* start_adapt_p = aecm->channelAdapt16;
int32_t* echo_est_p = echo_est;
const int16_t* end_stored_p = aecm->channelStored + PART_LEN;
const uint16_t* far_spectrum_p = far_spectrum;
int16x8_t store_v, adapt_v;
uint16x8_t spectrum_v;
uint32x4_t echo_est_v_low, echo_est_v_high;
uint32x4_t far_energy_v, echo_stored_v, echo_adapt_v;
far_energy_v = vdupq_n_u32(0);
echo_adapt_v = vdupq_n_u32(0);
echo_stored_v = vdupq_n_u32(0);
// Get energy for the delayed far end signal and estimated
// echo using both stored and adapted channels.
// The C code:
// for (i = 0; i < PART_LEN1; i++) {
// echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
// far_spectrum[i]);
// (*far_energy) += (uint32_t)(far_spectrum[i]);
// *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
// (*echo_energy_stored) += (uint32_t)echo_est[i];
// }
while (start_stored_p < end_stored_p) {
spectrum_v = vld1q_u16(far_spectrum_p);
adapt_v = vld1q_s16(start_adapt_p);
store_v = vld1q_s16(start_stored_p);
far_energy_v = vaddw_u16(far_energy_v, vget_low_u16(spectrum_v));
far_energy_v = vaddw_u16(far_energy_v, vget_high_u16(spectrum_v));
echo_est_v_low = vmull_u16(vreinterpret_u16_s16(vget_low_s16(store_v)),
vget_low_u16(spectrum_v));
echo_est_v_high = vmull_u16(vreinterpret_u16_s16(vget_high_s16(store_v)),
vget_high_u16(spectrum_v));
vst1q_s32(echo_est_p, vreinterpretq_s32_u32(echo_est_v_low));
vst1q_s32(echo_est_p + 4, vreinterpretq_s32_u32(echo_est_v_high));
echo_stored_v = vaddq_u32(echo_est_v_low, echo_stored_v);
echo_stored_v = vaddq_u32(echo_est_v_high, echo_stored_v);
echo_adapt_v =
vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_low_s16(adapt_v)),
vget_low_u16(spectrum_v));
echo_adapt_v =
vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_high_s16(adapt_v)),
vget_high_u16(spectrum_v));
start_stored_p += 8;
start_adapt_p += 8;
far_spectrum_p += 8;
echo_est_p += 8;
}
AddLanes(far_energy, far_energy_v);
AddLanes(echo_energy_stored, echo_stored_v);
AddLanes(echo_energy_adapt, echo_adapt_v);
echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
far_spectrum[PART_LEN]);
*echo_energy_stored += (uint32_t)echo_est[PART_LEN];
*far_energy += (uint32_t)far_spectrum[PART_LEN];
*echo_energy_adapt += aecm->channelAdapt16[PART_LEN] * far_spectrum[PART_LEN];
}
void WebRtcAecm_StoreAdaptiveChannelNeon(AecmCore* aecm,
const uint16_t* far_spectrum,
int32_t* echo_est) {
RTC_DCHECK_EQ(0, (uintptr_t)echo_est % 32);
RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
// This is C code of following optimized code.
// During startup we store the channel every block.
// memcpy(aecm->channelStored,
// aecm->channelAdapt16,
// sizeof(int16_t) * PART_LEN1);
// Recalculate echo estimate
// for (i = 0; i < PART_LEN; i += 4) {
// echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
// far_spectrum[i]);
// echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
// far_spectrum[i + 1]);
// echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
// far_spectrum[i + 2]);
// echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
// far_spectrum[i + 3]);
// }
// echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
// far_spectrum[i]);
const uint16_t* far_spectrum_p = far_spectrum;
int16_t* start_adapt_p = aecm->channelAdapt16;
int16_t* start_stored_p = aecm->channelStored;
const int16_t* end_stored_p = aecm->channelStored + PART_LEN;
int32_t* echo_est_p = echo_est;
uint16x8_t far_spectrum_v;
int16x8_t adapt_v;
uint32x4_t echo_est_v_low, echo_est_v_high;
while (start_stored_p < end_stored_p) {
far_spectrum_v = vld1q_u16(far_spectrum_p);
adapt_v = vld1q_s16(start_adapt_p);
vst1q_s16(start_stored_p, adapt_v);
echo_est_v_low = vmull_u16(vget_low_u16(far_spectrum_v),
vget_low_u16(vreinterpretq_u16_s16(adapt_v)));
echo_est_v_high = vmull_u16(vget_high_u16(far_spectrum_v),
vget_high_u16(vreinterpretq_u16_s16(adapt_v)));
vst1q_s32(echo_est_p, vreinterpretq_s32_u32(echo_est_v_low));
vst1q_s32(echo_est_p + 4, vreinterpretq_s32_u32(echo_est_v_high));
far_spectrum_p += 8;
start_adapt_p += 8;
start_stored_p += 8;
echo_est_p += 8;
}
aecm->channelStored[PART_LEN] = aecm->channelAdapt16[PART_LEN];
echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
far_spectrum[PART_LEN]);
}
void WebRtcAecm_ResetAdaptiveChannelNeon(AecmCore* aecm) {
RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelStored % 16);
RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt16 % 16);
RTC_DCHECK_EQ(0, (uintptr_t)aecm->channelAdapt32 % 32);
// The C code of following optimized code.
// for (i = 0; i < PART_LEN1; i++) {
// aecm->channelAdapt16[i] = aecm->channelStored[i];
// aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
// (int32_t)aecm->channelStored[i], 16);
// }
int16_t* start_stored_p = aecm->channelStored;
int16_t* start_adapt16_p = aecm->channelAdapt16;
int32_t* start_adapt32_p = aecm->channelAdapt32;
const int16_t* end_stored_p = start_stored_p + PART_LEN;
int16x8_t stored_v;
int32x4_t adapt32_v_low, adapt32_v_high;
while (start_stored_p < end_stored_p) {
stored_v = vld1q_s16(start_stored_p);
vst1q_s16(start_adapt16_p, stored_v);
adapt32_v_low = vshll_n_s16(vget_low_s16(stored_v), 16);
adapt32_v_high = vshll_n_s16(vget_high_s16(stored_v), 16);
vst1q_s32(start_adapt32_p, adapt32_v_low);
vst1q_s32(start_adapt32_p + 4, adapt32_v_high);
start_stored_p += 8;
start_adapt16_p += 8;
start_adapt32_p += 8;
}
aecm->channelAdapt16[PART_LEN] = aecm->channelStored[PART_LEN];
aecm->channelAdapt32[PART_LEN] = (int32_t)aecm->channelStored[PART_LEN] << 16;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
#define MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
#define AECM_DYNAMIC_Q /* Turn on/off dynamic Q-domain. */
/* Algorithm parameters */
#define FRAME_LEN 80 /* Total frame length, 10 ms. */
#define PART_LEN 64 /* Length of partition. */
#define PART_LEN_SHIFT 7 /* Length of (PART_LEN * 2) in base 2. */
#define PART_LEN1 (PART_LEN + 1) /* Unique fft coefficients. */
#define PART_LEN2 (PART_LEN << 1) /* Length of partition * 2. */
#define PART_LEN4 (PART_LEN << 2) /* Length of partition * 4. */
#define FAR_BUF_LEN PART_LEN4 /* Length of buffers. */
#define MAX_DELAY 100
/* Counter parameters */
#define CONV_LEN 512 /* Convergence length used at startup. */
#define CONV_LEN2 (CONV_LEN << 1) /* Used at startup. */
/* Energy parameters */
#define MAX_BUF_LEN 64 /* History length of energy signals. */
#define FAR_ENERGY_MIN 1025 /* Lowest Far energy level: At least 2 */
/* in energy. */
#define FAR_ENERGY_DIFF 929 /* Allowed difference between max */
/* and min. */
#define ENERGY_DEV_OFFSET 0 /* The energy error offset in Q8. */
#define ENERGY_DEV_TOL 400 /* The energy estimation tolerance (Q8). */
#define FAR_ENERGY_VAD_REGION 230 /* Far VAD tolerance region. */
/* Stepsize parameters */
#define MU_MIN 10 /* Min stepsize 2^-MU_MIN (far end energy */
/* dependent). */
#define MU_MAX 1 /* Max stepsize 2^-MU_MAX (far end energy */
/* dependent). */
#define MU_DIFF 9 /* MU_MIN - MU_MAX */
/* Channel parameters */
#define MIN_MSE_COUNT 20 /* Min number of consecutive blocks with enough */
/* far end energy to compare channel estimates. */
#define MIN_MSE_DIFF 29 /* The ratio between adapted and stored channel to */
/* accept a new storage (0.8 in Q-MSE_RESOLUTION). */
#define MSE_RESOLUTION 5 /* MSE parameter resolution. */
#define RESOLUTION_CHANNEL16 12 /* W16 Channel in Q-RESOLUTION_CHANNEL16. */
#define RESOLUTION_CHANNEL32 28 /* W32 Channel in Q-RESOLUTION_CHANNEL. */
#define CHANNEL_VAD 16 /* Minimum energy in frequency band */
/* to update channel. */
/* Suppression gain parameters: SUPGAIN parameters in Q-(RESOLUTION_SUPGAIN). */
#define RESOLUTION_SUPGAIN 8 /* Channel in Q-(RESOLUTION_SUPGAIN). */
#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN) /* Default. */
#define SUPGAIN_ERROR_PARAM_A 3072 /* Estimation error parameter */
/* (Maximum gain) (8 in Q8). */
#define SUPGAIN_ERROR_PARAM_B 1536 /* Estimation error parameter */
/* (Gain before going down). */
#define SUPGAIN_ERROR_PARAM_D SUPGAIN_DEFAULT /* Estimation error parameter */
/* (Should be the same as Default) (1 in Q8). */
#define SUPGAIN_EPC_DT 200 /* SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL */
/* Defines for "check delay estimation" */
#define CORR_WIDTH 31 /* Number of samples to correlate over. */
#define CORR_MAX 16 /* Maximum correlation offset. */
#define CORR_MAX_BUF 63
#define CORR_DEV 4
#define CORR_MAX_LEVEL 20
#define CORR_MAX_LOW 4
#define CORR_BUF_LEN (CORR_MAX << 1) + 1
/* Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN. */
#define ONE_Q14 (1 << 14)
/* NLP defines */
#define NLP_COMP_LOW 3277 /* 0.2 in Q14 */
#define NLP_COMP_HIGH ONE_Q14 /* 1 in Q14 */
#endif

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/aecm/echo_control_mobile.h"
#ifdef AEC_DEBUG
#include <stdio.h>
#endif
#include <stdlib.h>
#include <string.h>
extern "C" {
#include "common_audio/ring_buffer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_processing/aecm/aecm_defines.h"
}
#include "modules/audio_processing/aecm/aecm_core.h"
namespace webrtc {
namespace {
#define BUF_SIZE_FRAMES 50 // buffer size (frames)
// Maximum length of resampled signal. Must be an integer multiple of frames
// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
// The factor of 2 handles wb, and the + 1 is as a safety margin
#define MAX_RESAMP_LEN (5 * FRAME_LEN)
static const size_t kBufSizeSamp =
BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
static const int kSampMsNb = 8; // samples per ms in nb
// Target suppression levels for nlp modes
// log{0.001, 0.00001, 0.00000001}
static const int kInitCheck = 42;
typedef struct {
int sampFreq;
int scSampFreq;
short bufSizeStart;
int knownDelay;
// Stores the last frame added to the farend buffer
short farendOld[2][FRAME_LEN];
short initFlag; // indicates if AEC has been initialized
// Variables used for averaging far end buffer size
short counter;
short sum;
short firstVal;
short checkBufSizeCtr;
// Variables used for delay shifts
short msInSndCardBuf;
short filtDelay;
int timeForDelayChange;
int ECstartup;
int checkBuffSize;
int delayChange;
short lastDelayDiff;
int16_t echoMode;
#ifdef AEC_DEBUG
FILE* bufFile;
FILE* delayFile;
FILE* preCompFile;
FILE* postCompFile;
#endif // AEC_DEBUG
// Structures
RingBuffer* farendBuf;
AecmCore* aecmCore;
} AecMobile;
} // namespace
// Estimates delay to set the position of the farend buffer read pointer
// (controlled by knownDelay)
static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf);
// Stuffs the farend buffer if the estimated delay is too large
static int WebRtcAecm_DelayComp(AecMobile* aecm);
void* WebRtcAecm_Create() {
// Allocate zero-filled memory.
AecMobile* aecm = static_cast<AecMobile*>(calloc(1, sizeof(AecMobile)));
aecm->aecmCore = WebRtcAecm_CreateCore();
if (!aecm->aecmCore) {
WebRtcAecm_Free(aecm);
return NULL;
}
aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp, sizeof(int16_t));
if (!aecm->farendBuf) {
WebRtcAecm_Free(aecm);
return NULL;
}
#ifdef AEC_DEBUG
aecm->aecmCore->farFile = fopen("aecFar.pcm", "wb");
aecm->aecmCore->nearFile = fopen("aecNear.pcm", "wb");
aecm->aecmCore->outFile = fopen("aecOut.pcm", "wb");
// aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
aecm->bufFile = fopen("aecBuf.dat", "wb");
aecm->delayFile = fopen("aecDelay.dat", "wb");
aecm->preCompFile = fopen("preComp.pcm", "wb");
aecm->postCompFile = fopen("postComp.pcm", "wb");
#endif // AEC_DEBUG
return aecm;
}
void WebRtcAecm_Free(void* aecmInst) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
if (aecm == NULL) {
return;
}
#ifdef AEC_DEBUG
fclose(aecm->aecmCore->farFile);
fclose(aecm->aecmCore->nearFile);
fclose(aecm->aecmCore->outFile);
// fclose(aecm->aecmCore->outLpFile);
fclose(aecm->bufFile);
fclose(aecm->delayFile);
fclose(aecm->preCompFile);
fclose(aecm->postCompFile);
#endif // AEC_DEBUG
WebRtcAecm_FreeCore(aecm->aecmCore);
WebRtc_FreeBuffer(aecm->farendBuf);
free(aecm);
}
int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
AecmConfig aecConfig;
if (aecm == NULL) {
return -1;
}
if (sampFreq != 8000 && sampFreq != 16000) {
return AECM_BAD_PARAMETER_ERROR;
}
aecm->sampFreq = sampFreq;
// Initialize AECM core
if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1) {
return AECM_UNSPECIFIED_ERROR;
}
// Initialize farend buffer
WebRtc_InitBuffer(aecm->farendBuf);
aecm->initFlag = kInitCheck; // indicates that initialization has been done
aecm->delayChange = 1;
aecm->sum = 0;
aecm->counter = 0;
aecm->checkBuffSize = 1;
aecm->firstVal = 0;
aecm->ECstartup = 1;
aecm->bufSizeStart = 0;
aecm->checkBufSizeCtr = 0;
aecm->filtDelay = 0;
aecm->timeForDelayChange = 0;
aecm->knownDelay = 0;
aecm->lastDelayDiff = 0;
memset(&aecm->farendOld, 0, sizeof(aecm->farendOld));
// Default settings.
aecConfig.cngMode = AecmTrue;
aecConfig.echoMode = 3;
if (WebRtcAecm_set_config(aecm, aecConfig) == -1) {
return AECM_UNSPECIFIED_ERROR;
}
return 0;
}
// Returns any error that is caused when buffering the
// farend signal.
int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
const int16_t* farend,
size_t nrOfSamples) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
if (aecm == NULL)
return -1;
if (farend == NULL)
return AECM_NULL_POINTER_ERROR;
if (aecm->initFlag != kInitCheck)
return AECM_UNINITIALIZED_ERROR;
if (nrOfSamples != 80 && nrOfSamples != 160)
return AECM_BAD_PARAMETER_ERROR;
return 0;
}
int32_t WebRtcAecm_BufferFarend(void* aecmInst,
const int16_t* farend,
size_t nrOfSamples) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
const int32_t err =
WebRtcAecm_GetBufferFarendError(aecmInst, farend, nrOfSamples);
if (err != 0)
return err;
// TODO(unknown): Is this really a good idea?
if (!aecm->ECstartup) {
WebRtcAecm_DelayComp(aecm);
}
WebRtc_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
return 0;
}
int32_t WebRtcAecm_Process(void* aecmInst,
const int16_t* nearendNoisy,
const int16_t* nearendClean,
int16_t* out,
size_t nrOfSamples,
int16_t msInSndCardBuf) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
int32_t retVal = 0;
size_t i;
short nmbrOfFilledBuffers;
size_t nBlocks10ms;
size_t nFrames;
#ifdef AEC_DEBUG
short msInAECBuf;
#endif
if (aecm == NULL) {
return -1;
}
if (nearendNoisy == NULL) {
return AECM_NULL_POINTER_ERROR;
}
if (out == NULL) {
return AECM_NULL_POINTER_ERROR;
}
if (aecm->initFlag != kInitCheck) {
return AECM_UNINITIALIZED_ERROR;
}
if (nrOfSamples != 80 && nrOfSamples != 160) {
return AECM_BAD_PARAMETER_ERROR;
}
if (msInSndCardBuf < 0) {
msInSndCardBuf = 0;
retVal = AECM_BAD_PARAMETER_WARNING;
} else if (msInSndCardBuf > 500) {
msInSndCardBuf = 500;
retVal = AECM_BAD_PARAMETER_WARNING;
}
msInSndCardBuf += 10;
aecm->msInSndCardBuf = msInSndCardBuf;
nFrames = nrOfSamples / FRAME_LEN;
nBlocks10ms = nFrames / aecm->aecmCore->mult;
if (aecm->ECstartup) {
if (nearendClean == NULL) {
if (out != nearendNoisy) {
memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
}
} else if (out != nearendClean) {
memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
}
nmbrOfFilledBuffers =
(short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
// The AECM is in the start up mode
// AECM is disabled until the soundcard buffer and farend buffers are OK
// Mechanism to ensure that the soundcard buffer is reasonably stable.
if (aecm->checkBuffSize) {
aecm->checkBufSizeCtr++;
// Before we fill up the far end buffer we require the amount of data on
// the sound card to be stable (+/-8 ms) compared to the first value. This
// comparison is made during the following 4 consecutive frames. If it
// seems to be stable then we start to fill up the far end buffer.
if (aecm->counter == 0) {
aecm->firstVal = aecm->msInSndCardBuf;
aecm->sum = 0;
}
if (abs(aecm->firstVal - aecm->msInSndCardBuf) <
WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb)) {
aecm->sum += aecm->msInSndCardBuf;
aecm->counter++;
} else {
aecm->counter = 0;
}
if (aecm->counter * nBlocks10ms >= 6) {
// The farend buffer size is determined in blocks of 80 samples
// Use 75% of the average value of the soundcard buffer
aecm->bufSizeStart = WEBRTC_SPL_MIN(
(3 * aecm->sum * aecm->aecmCore->mult) / (aecm->counter * 40),
BUF_SIZE_FRAMES);
// buffersize has now been determined
aecm->checkBuffSize = 0;
}
if (aecm->checkBufSizeCtr * nBlocks10ms > 50) {
// for really bad sound cards, don't disable echocanceller for more than
// 0.5 sec
aecm->bufSizeStart = WEBRTC_SPL_MIN(
(3 * aecm->msInSndCardBuf * aecm->aecmCore->mult) / 40,
BUF_SIZE_FRAMES);
aecm->checkBuffSize = 0;
}
}
// if checkBuffSize changed in the if-statement above
if (!aecm->checkBuffSize) {
// soundcard buffer is now reasonably stable
// When the far end buffer is filled with approximately the same amount of
// data as the amount on the sound card we end the start up phase and
// start to cancel echoes.
if (nmbrOfFilledBuffers == aecm->bufSizeStart) {
aecm->ECstartup = 0; // Enable the AECM
} else if (nmbrOfFilledBuffers > aecm->bufSizeStart) {
WebRtc_MoveReadPtr(aecm->farendBuf,
(int)WebRtc_available_read(aecm->farendBuf) -
(int)aecm->bufSizeStart * FRAME_LEN);
aecm->ECstartup = 0;
}
}
} else {
// AECM is enabled
// Note only 1 block supported for nb and 2 blocks for wb
for (i = 0; i < nFrames; i++) {
int16_t farend[FRAME_LEN];
const int16_t* farend_ptr = NULL;
nmbrOfFilledBuffers =
(short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
// Check that there is data in the far end buffer
if (nmbrOfFilledBuffers > 0) {
// Get the next 80 samples from the farend buffer
WebRtc_ReadBuffer(aecm->farendBuf, (void**)&farend_ptr, farend,
FRAME_LEN);
// Always store the last frame for use when we run out of data
memcpy(&(aecm->farendOld[i][0]), farend_ptr, FRAME_LEN * sizeof(short));
} else {
// We have no data so we use the last played frame
memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
farend_ptr = farend;
}
// Call buffer delay estimator when all data is extracted,
// i,e. i = 0 for NB and i = 1 for WB
if ((i == 0 && aecm->sampFreq == 8000) ||
(i == 1 && aecm->sampFreq == 16000)) {
WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
}
// Call the AECM
/*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
&out[FRAME_LEN * i], aecm->knownDelay);*/
if (WebRtcAecm_ProcessFrame(
aecm->aecmCore, farend_ptr, &nearendNoisy[FRAME_LEN * i],
(nearendClean ? &nearendClean[FRAME_LEN * i] : NULL),
&out[FRAME_LEN * i]) == -1)
return -1;
}
}
#ifdef AEC_DEBUG
msInAECBuf = (short)WebRtc_available_read(aecm->farendBuf) /
(kSampMsNb * aecm->aecmCore->mult);
fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
#endif
return retVal;
}
int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
if (aecm == NULL) {
return -1;
}
if (aecm->initFlag != kInitCheck) {
return AECM_UNINITIALIZED_ERROR;
}
if (config.cngMode != AecmFalse && config.cngMode != AecmTrue) {
return AECM_BAD_PARAMETER_ERROR;
}
aecm->aecmCore->cngMode = config.cngMode;
if (config.echoMode < 0 || config.echoMode > 4) {
return AECM_BAD_PARAMETER_ERROR;
}
aecm->echoMode = config.echoMode;
if (aecm->echoMode == 0) {
aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
aecm->aecmCore->supGainErrParamDiffAB =
(SUPGAIN_ERROR_PARAM_A >> 3) - (SUPGAIN_ERROR_PARAM_B >> 3);
aecm->aecmCore->supGainErrParamDiffBD =
(SUPGAIN_ERROR_PARAM_B >> 3) - (SUPGAIN_ERROR_PARAM_D >> 3);
} else if (aecm->echoMode == 1) {
aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
aecm->aecmCore->supGainErrParamDiffAB =
(SUPGAIN_ERROR_PARAM_A >> 2) - (SUPGAIN_ERROR_PARAM_B >> 2);
aecm->aecmCore->supGainErrParamDiffBD =
(SUPGAIN_ERROR_PARAM_B >> 2) - (SUPGAIN_ERROR_PARAM_D >> 2);
} else if (aecm->echoMode == 2) {
aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
aecm->aecmCore->supGainErrParamDiffAB =
(SUPGAIN_ERROR_PARAM_A >> 1) - (SUPGAIN_ERROR_PARAM_B >> 1);
aecm->aecmCore->supGainErrParamDiffBD =
(SUPGAIN_ERROR_PARAM_B >> 1) - (SUPGAIN_ERROR_PARAM_D >> 1);
} else if (aecm->echoMode == 3) {
aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
aecm->aecmCore->supGainErrParamDiffAB =
SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
aecm->aecmCore->supGainErrParamDiffBD =
SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
} else if (aecm->echoMode == 4) {
aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
aecm->aecmCore->supGainErrParamDiffAB =
(SUPGAIN_ERROR_PARAM_A << 1) - (SUPGAIN_ERROR_PARAM_B << 1);
aecm->aecmCore->supGainErrParamDiffBD =
(SUPGAIN_ERROR_PARAM_B << 1) - (SUPGAIN_ERROR_PARAM_D << 1);
}
return 0;
}
int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
const void* echo_path,
size_t size_bytes) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
const int16_t* echo_path_ptr = static_cast<const int16_t*>(echo_path);
if (aecmInst == NULL) {
return -1;
}
if (echo_path == NULL) {
return AECM_NULL_POINTER_ERROR;
}
if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
// Input channel size does not match the size of AECM
return AECM_BAD_PARAMETER_ERROR;
}
if (aecm->initFlag != kInitCheck) {
return AECM_UNINITIALIZED_ERROR;
}
WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
return 0;
}
int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
void* echo_path,
size_t size_bytes) {
AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
int16_t* echo_path_ptr = static_cast<int16_t*>(echo_path);
if (aecmInst == NULL) {
return -1;
}
if (echo_path == NULL) {
return AECM_NULL_POINTER_ERROR;
}
if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
// Input channel size does not match the size of AECM
return AECM_BAD_PARAMETER_ERROR;
}
if (aecm->initFlag != kInitCheck) {
return AECM_UNINITIALIZED_ERROR;
}
memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
return 0;
}
size_t WebRtcAecm_echo_path_size_bytes() {
return (PART_LEN1 * sizeof(int16_t));
}
static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf) {
short delayNew, nSampSndCard;
short nSampFar = (short)WebRtc_available_read(aecm->farendBuf);
short diff;
nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
delayNew = nSampSndCard - nSampFar;
if (delayNew < FRAME_LEN) {
WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
delayNew += FRAME_LEN;
}
aecm->filtDelay =
WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
diff = aecm->filtDelay - aecm->knownDelay;
if (diff > 224) {
if (aecm->lastDelayDiff < 96) {
aecm->timeForDelayChange = 0;
} else {
aecm->timeForDelayChange++;
}
} else if (diff < 96 && aecm->knownDelay > 0) {
if (aecm->lastDelayDiff > 224) {
aecm->timeForDelayChange = 0;
} else {
aecm->timeForDelayChange++;
}
} else {
aecm->timeForDelayChange = 0;
}
aecm->lastDelayDiff = diff;
if (aecm->timeForDelayChange > 25) {
aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
}
return 0;
}
static int WebRtcAecm_DelayComp(AecMobile* aecm) {
int nSampFar = (int)WebRtc_available_read(aecm->farendBuf);
int nSampSndCard, delayNew, nSampAdd;
const int maxStuffSamp = 10 * FRAME_LEN;
nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
delayNew = nSampSndCard - nSampFar;
if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult) {
// The difference of the buffer sizes is larger than the maximum
// allowed known delay. Compensate by stuffing the buffer.
nSampAdd =
(int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar), FRAME_LEN));
nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
aecm->delayChange = 1; // the delay needs to be updated
}
return 0;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
#define MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
#include <stddef.h>
#include <stdint.h>
namespace webrtc {
enum { AecmFalse = 0, AecmTrue };
// Errors
#define AECM_UNSPECIFIED_ERROR 12000
#define AECM_UNSUPPORTED_FUNCTION_ERROR 12001
#define AECM_UNINITIALIZED_ERROR 12002
#define AECM_NULL_POINTER_ERROR 12003
#define AECM_BAD_PARAMETER_ERROR 12004
// Warnings
#define AECM_BAD_PARAMETER_WARNING 12100
typedef struct {
int16_t cngMode; // AECM_FALSE, AECM_TRUE (default)
int16_t echoMode; // 0, 1, 2, 3 (default), 4
} AecmConfig;
#ifdef __cplusplus
extern "C" {
#endif
/*
* Allocates the memory needed by the AECM. The memory needs to be
* initialized separately using the WebRtcAecm_Init() function.
* Returns a pointer to the instance and a nullptr at failure.
*/
void* WebRtcAecm_Create();
/*
* This function releases the memory allocated by WebRtcAecm_Create()
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
*/
void WebRtcAecm_Free(void* aecmInst);
/*
* Initializes an AECM instance.
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* int32_t sampFreq Sampling frequency of data
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq);
/*
* Inserts an 80 or 160 sample block of data into the farend buffer.
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* int16_t* farend In buffer containing one frame of
* farend signal
* int16_t nrOfSamples Number of samples in farend buffer
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_BufferFarend(void* aecmInst,
const int16_t* farend,
size_t nrOfSamples);
/*
* Reports any errors that would arise when buffering a farend buffer.
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* int16_t* farend In buffer containing one frame of
* farend signal
* int16_t nrOfSamples Number of samples in farend buffer
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
const int16_t* farend,
size_t nrOfSamples);
/*
* Runs the AECM on an 80 or 160 sample blocks of data.
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* int16_t* nearendNoisy In buffer containing one frame of
* reference nearend+echo signal. If
* noise reduction is active, provide
* the noisy signal here.
* int16_t* nearendClean In buffer containing one frame of
* nearend+echo signal. If noise
* reduction is active, provide the
* clean signal here. Otherwise pass a
* NULL pointer.
* int16_t nrOfSamples Number of samples in nearend buffer
* int16_t msInSndCardBuf Delay estimate for sound card and
* system buffers
*
* Outputs Description
* -------------------------------------------------------------------
* int16_t* out Out buffer, one frame of processed nearend
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_Process(void* aecmInst,
const int16_t* nearendNoisy,
const int16_t* nearendClean,
int16_t* out,
size_t nrOfSamples,
int16_t msInSndCardBuf);
/*
* This function enables the user to set certain parameters on-the-fly
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* AecmConfig config Config instance that contains all
* properties to be set
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config);
/*
* This function enables the user to set the echo path on-the-fly.
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* void* echo_path Pointer to the echo path to be set
* size_t size_bytes Size in bytes of the echo path
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
const void* echo_path,
size_t size_bytes);
/*
* This function enables the user to get the currently used echo path
* on-the-fly
*
* Inputs Description
* -------------------------------------------------------------------
* void* aecmInst Pointer to the AECM instance
* void* echo_path Pointer to echo path
* size_t size_bytes Size in bytes of the echo path
*
* Outputs Description
* -------------------------------------------------------------------
* int32_t return 0: OK
* 1200-12004,12100: error/warning
*/
int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
void* echo_path,
size_t size_bytes);
/*
* This function enables the user to get the echo path size in bytes
*
* Outputs Description
* -------------------------------------------------------------------
* size_t return Size in bytes
*/
size_t WebRtcAecm_echo_path_size_bytes();
#ifdef __cplusplus
}
#endif
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/utility/delay_estimator.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
// Number of right shifts for scaling is linearly depending on number of bits in
// the far-end binary spectrum.
static const int kShiftsAtZero = 13; // Right shifts at zero binary spectrum.
static const int kShiftsLinearSlope = 3;
static const int32_t kProbabilityOffset = 1024; // 2 in Q9.
static const int32_t kProbabilityLowerLimit = 8704; // 17 in Q9.
static const int32_t kProbabilityMinSpread = 2816; // 5.5 in Q9.
// Robust validation settings
static const float kHistogramMax = 3000.f;
static const float kLastHistogramMax = 250.f;
static const float kMinHistogramThreshold = 1.5f;
static const int kMinRequiredHits = 10;
static const int kMaxHitsWhenPossiblyNonCausal = 10;
static const int kMaxHitsWhenPossiblyCausal = 1000;
static const float kQ14Scaling = 1.f / (1 << 14); // Scaling by 2^14 to get Q0.
static const float kFractionSlope = 0.05f;
static const float kMinFractionWhenPossiblyCausal = 0.5f;
static const float kMinFractionWhenPossiblyNonCausal = 0.25f;
} // namespace
// Counts and returns number of bits of a 32-bit word.
static int BitCount(uint32_t u32) {
uint32_t tmp =
u32 - ((u32 >> 1) & 033333333333) - ((u32 >> 2) & 011111111111);
tmp = ((tmp + (tmp >> 3)) & 030707070707);
tmp = (tmp + (tmp >> 6));
tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
return ((int)tmp);
}
// Compares the `binary_vector` with all rows of the `binary_matrix` and counts
// per row the number of times they have the same value.
//
// Inputs:
// - binary_vector : binary "vector" stored in a long
// - binary_matrix : binary "matrix" stored as a vector of long
// - matrix_size : size of binary "matrix"
//
// Output:
// - bit_counts : "Vector" stored as a long, containing for each
// row the number of times the matrix row and the
// input vector have the same value
//
static void BitCountComparison(uint32_t binary_vector,
const uint32_t* binary_matrix,
int matrix_size,
int32_t* bit_counts) {
int n = 0;
// Compare `binary_vector` with all rows of the `binary_matrix`
for (; n < matrix_size; n++) {
bit_counts[n] = (int32_t)BitCount(binary_vector ^ binary_matrix[n]);
}
}
// Collects necessary statistics for the HistogramBasedValidation(). This
// function has to be called prior to calling HistogramBasedValidation(). The
// statistics updated and used by the HistogramBasedValidation() are:
// 1. the number of `candidate_hits`, which states for how long we have had the
// same `candidate_delay`
// 2. the `histogram` of candidate delays over time. This histogram is
// weighted with respect to a reliability measure and time-varying to cope
// with possible delay shifts.
// For further description see commented code.
//
// Inputs:
// - candidate_delay : The delay to validate.
// - valley_depth_q14 : The cost function has a valley/minimum at the
// `candidate_delay` location. `valley_depth_q14` is the
// cost function difference between the minimum and
// maximum locations. The value is in the Q14 domain.
// - valley_level_q14 : Is the cost function value at the minimum, in Q14.
static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self,
int candidate_delay,
int32_t valley_depth_q14,
int32_t valley_level_q14) {
const float valley_depth = valley_depth_q14 * kQ14Scaling;
float decrease_in_last_set = valley_depth;
const int max_hits_for_slow_change = (candidate_delay < self->last_delay)
? kMaxHitsWhenPossiblyNonCausal
: kMaxHitsWhenPossiblyCausal;
int i = 0;
RTC_DCHECK_EQ(self->history_size, self->farend->history_size);
// Reset `candidate_hits` if we have a new candidate.
if (candidate_delay != self->last_candidate_delay) {
self->candidate_hits = 0;
self->last_candidate_delay = candidate_delay;
}
self->candidate_hits++;
// The `histogram` is updated differently across the bins.
// 1. The `candidate_delay` histogram bin is increased with the
// `valley_depth`, which is a simple measure of how reliable the
// `candidate_delay` is. The histogram is not increased above
// `kHistogramMax`.
self->histogram[candidate_delay] += valley_depth;
if (self->histogram[candidate_delay] > kHistogramMax) {
self->histogram[candidate_delay] = kHistogramMax;
}
// 2. The histogram bins in the neighborhood of `candidate_delay` are
// unaffected. The neighborhood is defined as x + {-2, -1, 0, 1}.
// 3. The histogram bins in the neighborhood of `last_delay` are decreased
// with `decrease_in_last_set`. This value equals the difference between
// the cost function values at the locations `candidate_delay` and
// `last_delay` until we reach `max_hits_for_slow_change` consecutive hits
// at the `candidate_delay`. If we exceed this amount of hits the
// `candidate_delay` is a "potential" candidate and we start decreasing
// these histogram bins more rapidly with `valley_depth`.
if (self->candidate_hits < max_hits_for_slow_change) {
decrease_in_last_set =
(self->mean_bit_counts[self->compare_delay] - valley_level_q14) *
kQ14Scaling;
}
// 4. All other bins are decreased with `valley_depth`.
// TODO(bjornv): Investigate how to make this loop more efficient. Split up
// the loop? Remove parts that doesn't add too much.
for (i = 0; i < self->history_size; ++i) {
int is_in_last_set = (i >= self->last_delay - 2) &&
(i <= self->last_delay + 1) && (i != candidate_delay);
int is_in_candidate_set =
(i >= candidate_delay - 2) && (i <= candidate_delay + 1);
self->histogram[i] -=
decrease_in_last_set * is_in_last_set +
valley_depth * (!is_in_last_set && !is_in_candidate_set);
// 5. No histogram bin can go below 0.
if (self->histogram[i] < 0) {
self->histogram[i] = 0;
}
}
}
// Validates the `candidate_delay`, estimated in WebRtc_ProcessBinarySpectrum(),
// based on a mix of counting concurring hits with a modified histogram
// of recent delay estimates. In brief a candidate is valid (returns 1) if it
// is the most likely according to the histogram. There are a couple of
// exceptions that are worth mentioning:
// 1. If the `candidate_delay` < `last_delay` it can be that we are in a
// non-causal state, breaking a possible echo control algorithm. Hence, we
// open up for a quicker change by allowing the change even if the
// `candidate_delay` is not the most likely one according to the histogram.
// 2. There's a minimum number of hits (kMinRequiredHits) and the histogram
// value has to reached a minimum (kMinHistogramThreshold) to be valid.
// 3. The action is also depending on the filter length used for echo control.
// If the delay difference is larger than what the filter can capture, we
// also move quicker towards a change.
// For further description see commented code.
//
// Input:
// - candidate_delay : The delay to validate.
//
// Return value:
// - is_histogram_valid : 1 - The `candidate_delay` is valid.
// 0 - Otherwise.
static int HistogramBasedValidation(const BinaryDelayEstimator* self,
int candidate_delay) {
float fraction = 1.f;
float histogram_threshold = self->histogram[self->compare_delay];
const int delay_difference = candidate_delay - self->last_delay;
int is_histogram_valid = 0;
// The histogram based validation of `candidate_delay` is done by comparing
// the `histogram` at bin `candidate_delay` with a `histogram_threshold`.
// This `histogram_threshold` equals a `fraction` of the `histogram` at bin
// `last_delay`. The `fraction` is a piecewise linear function of the
// `delay_difference` between the `candidate_delay` and the `last_delay`
// allowing for a quicker move if
// i) a potential echo control filter can not handle these large differences.
// ii) keeping `last_delay` instead of updating to `candidate_delay` could
// force an echo control into a non-causal state.
// We further require the histogram to have reached a minimum value of
// `kMinHistogramThreshold`. In addition, we also require the number of
// `candidate_hits` to be more than `kMinRequiredHits` to remove spurious
// values.
// Calculate a comparison histogram value (`histogram_threshold`) that is
// depending on the distance between the `candidate_delay` and `last_delay`.
// TODO(bjornv): How much can we gain by turning the fraction calculation
// into tables?
if (delay_difference > self->allowed_offset) {
fraction = 1.f - kFractionSlope * (delay_difference - self->allowed_offset);
fraction = (fraction > kMinFractionWhenPossiblyCausal
? fraction
: kMinFractionWhenPossiblyCausal);
} else if (delay_difference < 0) {
fraction =
kMinFractionWhenPossiblyNonCausal - kFractionSlope * delay_difference;
fraction = (fraction > 1.f ? 1.f : fraction);
}
histogram_threshold *= fraction;
histogram_threshold =
(histogram_threshold > kMinHistogramThreshold ? histogram_threshold
: kMinHistogramThreshold);
is_histogram_valid =
(self->histogram[candidate_delay] >= histogram_threshold) &&
(self->candidate_hits > kMinRequiredHits);
return is_histogram_valid;
}
// Performs a robust validation of the `candidate_delay` estimated in
// WebRtc_ProcessBinarySpectrum(). The algorithm takes the
// `is_instantaneous_valid` and the `is_histogram_valid` and combines them
// into a robust validation. The HistogramBasedValidation() has to be called
// prior to this call.
// For further description on how the combination is done, see commented code.
//
// Inputs:
// - candidate_delay : The delay to validate.
// - is_instantaneous_valid : The instantaneous validation performed in
// WebRtc_ProcessBinarySpectrum().
// - is_histogram_valid : The histogram based validation.
//
// Return value:
// - is_robust : 1 - The candidate_delay is valid according to a
// combination of the two inputs.
// : 0 - Otherwise.
static int RobustValidation(const BinaryDelayEstimator* self,
int candidate_delay,
int is_instantaneous_valid,
int is_histogram_valid) {
int is_robust = 0;
// The final robust validation is based on the two algorithms; 1) the
// `is_instantaneous_valid` and 2) the histogram based with result stored in
// `is_histogram_valid`.
// i) Before we actually have a valid estimate (`last_delay` == -2), we say
// a candidate is valid if either algorithm states so
// (`is_instantaneous_valid` OR `is_histogram_valid`).
is_robust =
(self->last_delay < 0) && (is_instantaneous_valid || is_histogram_valid);
// ii) Otherwise, we need both algorithms to be certain
// (`is_instantaneous_valid` AND `is_histogram_valid`)
is_robust |= is_instantaneous_valid && is_histogram_valid;
// iii) With one exception, i.e., the histogram based algorithm can overrule
// the instantaneous one if `is_histogram_valid` = 1 and the histogram
// is significantly strong.
is_robust |= is_histogram_valid &&
(self->histogram[candidate_delay] > self->last_delay_histogram);
return is_robust;
}
void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
if (self == NULL) {
return;
}
free(self->binary_far_history);
self->binary_far_history = NULL;
free(self->far_bit_counts);
self->far_bit_counts = NULL;
free(self);
}
BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend(
int history_size) {
BinaryDelayEstimatorFarend* self = NULL;
if (history_size > 1) {
// Sanity conditions fulfilled.
self = static_cast<BinaryDelayEstimatorFarend*>(
malloc(sizeof(BinaryDelayEstimatorFarend)));
}
if (self == NULL) {
return NULL;
}
self->history_size = 0;
self->binary_far_history = NULL;
self->far_bit_counts = NULL;
if (WebRtc_AllocateFarendBufferMemory(self, history_size) == 0) {
WebRtc_FreeBinaryDelayEstimatorFarend(self);
self = NULL;
}
return self;
}
int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
int history_size) {
RTC_DCHECK(self);
// (Re-)Allocate memory for history buffers.
self->binary_far_history = static_cast<uint32_t*>(
realloc(self->binary_far_history,
history_size * sizeof(*self->binary_far_history)));
self->far_bit_counts = static_cast<int*>(realloc(
self->far_bit_counts, history_size * sizeof(*self->far_bit_counts)));
if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) {
history_size = 0;
}
// Fill with zeros if we have expanded the buffers.
if (history_size > self->history_size) {
int size_diff = history_size - self->history_size;
memset(&self->binary_far_history[self->history_size], 0,
sizeof(*self->binary_far_history) * size_diff);
memset(&self->far_bit_counts[self->history_size], 0,
sizeof(*self->far_bit_counts) * size_diff);
}
self->history_size = history_size;
return self->history_size;
}
void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
RTC_DCHECK(self);
memset(self->binary_far_history, 0, sizeof(uint32_t) * self->history_size);
memset(self->far_bit_counts, 0, sizeof(int) * self->history_size);
}
void WebRtc_SoftResetBinaryDelayEstimatorFarend(
BinaryDelayEstimatorFarend* self,
int delay_shift) {
int abs_shift = abs(delay_shift);
int shift_size = 0;
int dest_index = 0;
int src_index = 0;
int padding_index = 0;
RTC_DCHECK(self);
shift_size = self->history_size - abs_shift;
RTC_DCHECK_GT(shift_size, 0);
if (delay_shift == 0) {
return;
} else if (delay_shift > 0) {
dest_index = abs_shift;
} else if (delay_shift < 0) {
src_index = abs_shift;
padding_index = shift_size;
}
// Shift and zero pad buffers.
memmove(&self->binary_far_history[dest_index],
&self->binary_far_history[src_index],
sizeof(*self->binary_far_history) * shift_size);
memset(&self->binary_far_history[padding_index], 0,
sizeof(*self->binary_far_history) * abs_shift);
memmove(&self->far_bit_counts[dest_index], &self->far_bit_counts[src_index],
sizeof(*self->far_bit_counts) * shift_size);
memset(&self->far_bit_counts[padding_index], 0,
sizeof(*self->far_bit_counts) * abs_shift);
}
void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle,
uint32_t binary_far_spectrum) {
RTC_DCHECK(handle);
// Shift binary spectrum history and insert current `binary_far_spectrum`.
memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
(handle->history_size - 1) * sizeof(uint32_t));
handle->binary_far_history[0] = binary_far_spectrum;
// Shift history of far-end binary spectrum bit counts and insert bit count
// of current `binary_far_spectrum`.
memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
(handle->history_size - 1) * sizeof(int));
handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
}
void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) {
if (self == NULL) {
return;
}
free(self->mean_bit_counts);
self->mean_bit_counts = NULL;
free(self->bit_counts);
self->bit_counts = NULL;
free(self->binary_near_history);
self->binary_near_history = NULL;
free(self->histogram);
self->histogram = NULL;
// BinaryDelayEstimator does not have ownership of `farend`, hence we do not
// free the memory here. That should be handled separately by the user.
self->farend = NULL;
free(self);
}
BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
BinaryDelayEstimatorFarend* farend,
int max_lookahead) {
BinaryDelayEstimator* self = NULL;
if ((farend != NULL) && (max_lookahead >= 0)) {
// Sanity conditions fulfilled.
self = static_cast<BinaryDelayEstimator*>(
malloc(sizeof(BinaryDelayEstimator)));
}
if (self == NULL) {
return NULL;
}
self->farend = farend;
self->near_history_size = max_lookahead + 1;
self->history_size = 0;
self->robust_validation_enabled = 0; // Disabled by default.
self->allowed_offset = 0;
self->lookahead = max_lookahead;
// Allocate memory for spectrum and history buffers.
self->mean_bit_counts = NULL;
self->bit_counts = NULL;
self->histogram = NULL;
self->binary_near_history = static_cast<uint32_t*>(
malloc((max_lookahead + 1) * sizeof(*self->binary_near_history)));
if (self->binary_near_history == NULL ||
WebRtc_AllocateHistoryBufferMemory(self, farend->history_size) == 0) {
WebRtc_FreeBinaryDelayEstimator(self);
self = NULL;
}
return self;
}
int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
int history_size) {
BinaryDelayEstimatorFarend* far = self->farend;
// (Re-)Allocate memory for spectrum and history buffers.
if (history_size != far->history_size) {
// Only update far-end buffers if we need.
history_size = WebRtc_AllocateFarendBufferMemory(far, history_size);
}
// The extra array element in `mean_bit_counts` and `histogram` is a dummy
// element only used while `last_delay` == -2, i.e., before we have a valid
// estimate.
self->mean_bit_counts = static_cast<int32_t*>(
realloc(self->mean_bit_counts,
(history_size + 1) * sizeof(*self->mean_bit_counts)));
self->bit_counts = static_cast<int32_t*>(
realloc(self->bit_counts, history_size * sizeof(*self->bit_counts)));
self->histogram = static_cast<float*>(
realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram)));
if ((self->mean_bit_counts == NULL) || (self->bit_counts == NULL) ||
(self->histogram == NULL)) {
history_size = 0;
}
// Fill with zeros if we have expanded the buffers.
if (history_size > self->history_size) {
int size_diff = history_size - self->history_size;
memset(&self->mean_bit_counts[self->history_size], 0,
sizeof(*self->mean_bit_counts) * size_diff);
memset(&self->bit_counts[self->history_size], 0,
sizeof(*self->bit_counts) * size_diff);
memset(&self->histogram[self->history_size], 0,
sizeof(*self->histogram) * size_diff);
}
self->history_size = history_size;
return self->history_size;
}
void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) {
int i = 0;
RTC_DCHECK(self);
memset(self->bit_counts, 0, sizeof(int32_t) * self->history_size);
memset(self->binary_near_history, 0,
sizeof(uint32_t) * self->near_history_size);
for (i = 0; i <= self->history_size; ++i) {
self->mean_bit_counts[i] = (20 << 9); // 20 in Q9.
self->histogram[i] = 0.f;
}
self->minimum_probability = kMaxBitCountsQ9; // 32 in Q9.
self->last_delay_probability = (int)kMaxBitCountsQ9; // 32 in Q9.
// Default return value if we're unable to estimate. -1 is used for errors.
self->last_delay = -2;
self->last_candidate_delay = -2;
self->compare_delay = self->history_size;
self->candidate_hits = 0;
self->last_delay_histogram = 0.f;
}
int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self,
int delay_shift) {
int lookahead = 0;
RTC_DCHECK(self);
lookahead = self->lookahead;
self->lookahead -= delay_shift;
if (self->lookahead < 0) {
self->lookahead = 0;
}
if (self->lookahead > self->near_history_size - 1) {
self->lookahead = self->near_history_size - 1;
}
return lookahead - self->lookahead;
}
int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
uint32_t binary_near_spectrum) {
int i = 0;
int candidate_delay = -1;
int valid_candidate = 0;
int32_t value_best_candidate = kMaxBitCountsQ9;
int32_t value_worst_candidate = 0;
int32_t valley_depth = 0;
RTC_DCHECK(self);
if (self->farend->history_size != self->history_size) {
// Non matching history sizes.
return -1;
}
if (self->near_history_size > 1) {
// If we apply lookahead, shift near-end binary spectrum history. Insert
// current `binary_near_spectrum` and pull out the delayed one.
memmove(&(self->binary_near_history[1]), &(self->binary_near_history[0]),
(self->near_history_size - 1) * sizeof(uint32_t));
self->binary_near_history[0] = binary_near_spectrum;
binary_near_spectrum = self->binary_near_history[self->lookahead];
}
// Compare with delayed spectra and store the `bit_counts` for each delay.
BitCountComparison(binary_near_spectrum, self->farend->binary_far_history,
self->history_size, self->bit_counts);
// Update `mean_bit_counts`, which is the smoothed version of `bit_counts`.
for (i = 0; i < self->history_size; i++) {
// `bit_counts` is constrained to [0, 32], meaning we can smooth with a
// factor up to 2^26. We use Q9.
int32_t bit_count = (self->bit_counts[i] << 9); // Q9.
// Update `mean_bit_counts` only when far-end signal has something to
// contribute. If `far_bit_counts` is zero the far-end signal is weak and
// we likely have a poor echo condition, hence don't update.
if (self->farend->far_bit_counts[i] > 0) {
// Make number of right shifts piecewise linear w.r.t. `far_bit_counts`.
int shifts = kShiftsAtZero;
shifts -= (kShiftsLinearSlope * self->farend->far_bit_counts[i]) >> 4;
WebRtc_MeanEstimatorFix(bit_count, shifts, &(self->mean_bit_counts[i]));
}
}
// Find `candidate_delay`, `value_best_candidate` and `value_worst_candidate`
// of `mean_bit_counts`.
for (i = 0; i < self->history_size; i++) {
if (self->mean_bit_counts[i] < value_best_candidate) {
value_best_candidate = self->mean_bit_counts[i];
candidate_delay = i;
}
if (self->mean_bit_counts[i] > value_worst_candidate) {
value_worst_candidate = self->mean_bit_counts[i];
}
}
valley_depth = value_worst_candidate - value_best_candidate;
// The `value_best_candidate` is a good indicator on the probability of
// `candidate_delay` being an accurate delay (a small `value_best_candidate`
// means a good binary match). In the following sections we make a decision
// whether to update `last_delay` or not.
// 1) If the difference bit counts between the best and the worst delay
// candidates is too small we consider the situation to be unreliable and
// don't update `last_delay`.
// 2) If the situation is reliable we update `last_delay` if the value of the
// best candidate delay has a value less than
// i) an adaptive threshold `minimum_probability`, or
// ii) this corresponding value `last_delay_probability`, but updated at
// this time instant.
// Update `minimum_probability`.
if ((self->minimum_probability > kProbabilityLowerLimit) &&
(valley_depth > kProbabilityMinSpread)) {
// The "hard" threshold can't be lower than 17 (in Q9).
// The valley in the curve also has to be distinct, i.e., the
// difference between `value_worst_candidate` and `value_best_candidate` has
// to be large enough.
int32_t threshold = value_best_candidate + kProbabilityOffset;
if (threshold < kProbabilityLowerLimit) {
threshold = kProbabilityLowerLimit;
}
if (self->minimum_probability > threshold) {
self->minimum_probability = threshold;
}
}
// Update `last_delay_probability`.
// We use a Markov type model, i.e., a slowly increasing level over time.
self->last_delay_probability++;
// Validate `candidate_delay`. We have a reliable instantaneous delay
// estimate if
// 1) The valley is distinct enough (`valley_depth` > `kProbabilityOffset`)
// and
// 2) The depth of the valley is deep enough
// (`value_best_candidate` < `minimum_probability`)
// and deeper than the best estimate so far
// (`value_best_candidate` < `last_delay_probability`)
valid_candidate = ((valley_depth > kProbabilityOffset) &&
((value_best_candidate < self->minimum_probability) ||
(value_best_candidate < self->last_delay_probability)));
// Check for nonstationary farend signal.
const bool non_stationary_farend =
std::any_of(self->farend->far_bit_counts,
self->farend->far_bit_counts + self->history_size,
[](int a) { return a > 0; });
if (non_stationary_farend) {
// Only update the validation statistics when the farend is nonstationary
// as the underlying estimates are otherwise frozen.
UpdateRobustValidationStatistics(self, candidate_delay, valley_depth,
value_best_candidate);
}
if (self->robust_validation_enabled) {
int is_histogram_valid = HistogramBasedValidation(self, candidate_delay);
valid_candidate = RobustValidation(self, candidate_delay, valid_candidate,
is_histogram_valid);
}
// Only update the delay estimate when the farend is nonstationary and when
// a valid delay candidate is available.
if (non_stationary_farend && valid_candidate) {
if (candidate_delay != self->last_delay) {
self->last_delay_histogram =
(self->histogram[candidate_delay] > kLastHistogramMax
? kLastHistogramMax
: self->histogram[candidate_delay]);
// Adjust the histogram if we made a change to `last_delay`, though it was
// not the most likely one according to the histogram.
if (self->histogram[candidate_delay] <
self->histogram[self->compare_delay]) {
self->histogram[self->compare_delay] = self->histogram[candidate_delay];
}
}
self->last_delay = candidate_delay;
if (value_best_candidate < self->last_delay_probability) {
self->last_delay_probability = value_best_candidate;
}
self->compare_delay = self->last_delay;
}
return self->last_delay;
}
int WebRtc_binary_last_delay(BinaryDelayEstimator* self) {
RTC_DCHECK(self);
return self->last_delay;
}
float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self) {
float quality = 0;
RTC_DCHECK(self);
if (self->robust_validation_enabled) {
// Simply a linear function of the histogram height at delay estimate.
quality = self->histogram[self->compare_delay] / kHistogramMax;
} else {
// Note that `last_delay_probability` states how deep the minimum of the
// cost function is, so it is rather an error probability.
quality = (float)(kMaxBitCountsQ9 - self->last_delay_probability) /
kMaxBitCountsQ9;
if (quality < 0) {
quality = 0;
}
}
return quality;
}
void WebRtc_MeanEstimatorFix(int32_t new_value,
int factor,
int32_t* mean_value) {
int32_t diff = new_value - *mean_value;
// mean_new = mean_value + ((new_value - mean_value) >> factor);
if (diff < 0) {
diff = -((-diff) >> factor);
} else {
diff = (diff >> factor);
}
*mean_value += diff;
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Performs delay estimation on binary converted spectra.
// The return value is 0 - OK and -1 - Error, unless otherwise stated.
#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
#include <stdint.h>
namespace webrtc {
static const int32_t kMaxBitCountsQ9 = (32 << 9); // 32 matching bits in Q9.
typedef struct {
// Pointer to bit counts.
int* far_bit_counts;
// Binary history variables.
uint32_t* binary_far_history;
int history_size;
} BinaryDelayEstimatorFarend;
typedef struct {
// Pointer to bit counts.
int32_t* mean_bit_counts;
// Array only used locally in ProcessBinarySpectrum() but whose size is
// determined at run-time.
int32_t* bit_counts;
// Binary history variables.
uint32_t* binary_near_history;
int near_history_size;
int history_size;
// Delay estimation variables.
int32_t minimum_probability;
int last_delay_probability;
// Delay memory.
int last_delay;
// Robust validation
int robust_validation_enabled;
int allowed_offset;
int last_candidate_delay;
int compare_delay;
int candidate_hits;
float* histogram;
float last_delay_histogram;
// For dynamically changing the lookahead when using SoftReset...().
int lookahead;
// Far-end binary spectrum history buffer etc.
BinaryDelayEstimatorFarend* farend;
} BinaryDelayEstimator;
// Releases the memory allocated by
// WebRtc_CreateBinaryDelayEstimatorFarend(...).
// Input:
// - self : Pointer to the binary delay estimation far-end
// instance which is the return value of
// WebRtc_CreateBinaryDelayEstimatorFarend().
//
void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self);
// Allocates the memory needed by the far-end part of the binary delay
// estimation. The memory needs to be initialized separately through
// WebRtc_InitBinaryDelayEstimatorFarend(...).
//
// Inputs:
// - history_size : Size of the far-end binary spectrum history.
//
// Return value:
// - BinaryDelayEstimatorFarend*
// : Created `handle`. If the memory can't be allocated
// or if any of the input parameters are invalid NULL
// is returned.
//
BinaryDelayEstimatorFarend* WebRtc_CreateBinaryDelayEstimatorFarend(
int history_size);
// Re-allocates the buffers.
//
// Inputs:
// - self : Pointer to the binary estimation far-end instance
// which is the return value of
// WebRtc_CreateBinaryDelayEstimatorFarend().
// - history_size : Size of the far-end binary spectrum history.
//
// Return value:
// - history_size : The history size allocated.
int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
int history_size);
// Initializes the delay estimation far-end instance created with
// WebRtc_CreateBinaryDelayEstimatorFarend(...).
//
// Input:
// - self : Pointer to the delay estimation far-end instance.
//
// Output:
// - self : Initialized far-end instance.
//
void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self);
// Soft resets the delay estimation far-end instance created with
// WebRtc_CreateBinaryDelayEstimatorFarend(...).
//
// Input:
// - delay_shift : The amount of blocks to shift history buffers.
//
void WebRtc_SoftResetBinaryDelayEstimatorFarend(
BinaryDelayEstimatorFarend* self,
int delay_shift);
// Adds the binary far-end spectrum to the internal far-end history buffer. This
// spectrum is used as reference when calculating the delay using
// WebRtc_ProcessBinarySpectrum().
//
// Inputs:
// - self : Pointer to the delay estimation far-end
// instance.
// - binary_far_spectrum : Far-end binary spectrum.
//
// Output:
// - self : Updated far-end instance.
//
void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* self,
uint32_t binary_far_spectrum);
// Releases the memory allocated by WebRtc_CreateBinaryDelayEstimator(...).
//
// Note that BinaryDelayEstimator utilizes BinaryDelayEstimatorFarend, but does
// not take ownership of it, hence the BinaryDelayEstimator has to be torn down
// before the far-end.
//
// Input:
// - self : Pointer to the binary delay estimation instance
// which is the return value of
// WebRtc_CreateBinaryDelayEstimator().
//
void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self);
// Allocates the memory needed by the binary delay estimation. The memory needs
// to be initialized separately through WebRtc_InitBinaryDelayEstimator(...).
//
// See WebRtc_CreateDelayEstimator(..) in delay_estimator_wrapper.c for detailed
// description.
BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
BinaryDelayEstimatorFarend* farend,
int max_lookahead);
// Re-allocates `history_size` dependent buffers. The far-end buffers will be
// updated at the same time if needed.
//
// Input:
// - self : Pointer to the binary estimation instance which is
// the return value of
// WebRtc_CreateBinaryDelayEstimator().
// - history_size : Size of the history buffers.
//
// Return value:
// - history_size : The history size allocated.
int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
int history_size);
// Initializes the delay estimation instance created with
// WebRtc_CreateBinaryDelayEstimator(...).
//
// Input:
// - self : Pointer to the delay estimation instance.
//
// Output:
// - self : Initialized instance.
//
void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self);
// Soft resets the delay estimation instance created with
// WebRtc_CreateBinaryDelayEstimator(...).
//
// Input:
// - delay_shift : The amount of blocks to shift history buffers.
//
// Return value:
// - actual_shifts : The actual number of shifts performed.
//
int WebRtc_SoftResetBinaryDelayEstimator(BinaryDelayEstimator* self,
int delay_shift);
// Estimates and returns the delay between the binary far-end and binary near-
// end spectra. It is assumed the binary far-end spectrum has been added using
// WebRtc_AddBinaryFarSpectrum() prior to this call. The value will be offset by
// the lookahead (i.e. the lookahead should be subtracted from the returned
// value).
//
// Inputs:
// - self : Pointer to the delay estimation instance.
// - binary_near_spectrum : Near-end binary spectrum of the current block.
//
// Output:
// - self : Updated instance.
//
// Return value:
// - delay : >= 0 - Calculated delay value.
// -2 - Insufficient data for estimation.
//
int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
uint32_t binary_near_spectrum);
// Returns the last calculated delay updated by the function
// WebRtc_ProcessBinarySpectrum(...).
//
// Input:
// - self : Pointer to the delay estimation instance.
//
// Return value:
// - delay : >= 0 - Last calculated delay value
// -2 - Insufficient data for estimation.
//
int WebRtc_binary_last_delay(BinaryDelayEstimator* self);
// Returns the estimation quality of the last calculated delay updated by the
// function WebRtc_ProcessBinarySpectrum(...). The estimation quality is a value
// in the interval [0, 1]. The higher the value, the better the quality.
//
// Return value:
// - delay_quality : >= 0 - Estimation quality of last calculated
// delay value.
float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self);
// Updates the `mean_value` recursively with a step size of 2^-`factor`. This
// function is used internally in the Binary Delay Estimator as well as the
// Fixed point wrapper.
//
// Inputs:
// - new_value : The new value the mean should be updated with.
// - factor : The step size, in number of right shifts.
//
// Input/Output:
// - mean_value : Pointer to the mean value.
//
void WebRtc_MeanEstimatorFix(int32_t new_value,
int factor,
int32_t* mean_value);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Header file including the delay estimator handle used for testing.
#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
#include "modules/audio_processing/utility/delay_estimator.h"
namespace webrtc {
typedef union {
float float_;
int32_t int32_;
} SpectrumType;
typedef struct {
// Pointers to mean values of spectrum.
SpectrumType* mean_far_spectrum;
// `mean_far_spectrum` initialization indicator.
int far_spectrum_initialized;
int spectrum_size;
// Far-end part of binary spectrum based delay estimation.
BinaryDelayEstimatorFarend* binary_farend;
} DelayEstimatorFarend;
typedef struct {
// Pointers to mean values of spectrum.
SpectrumType* mean_near_spectrum;
// `mean_near_spectrum` initialization indicator.
int near_spectrum_initialized;
int spectrum_size;
// Binary spectrum based delay estimator
BinaryDelayEstimator* binary_handle;
} DelayEstimator;
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/utility/delay_estimator_wrapper.h"
#include <stdlib.h>
#include <string.h>
#include "modules/audio_processing/utility/delay_estimator.h"
#include "modules/audio_processing/utility/delay_estimator_internal.h"
#include "rtc_base/checks.h"
namespace webrtc {
// Only bit `kBandFirst` through bit `kBandLast` are processed and
// `kBandFirst` - `kBandLast` must be < 32.
constexpr int kBandFirst = 12;
constexpr int kBandLast = 43;
static __inline uint32_t SetBit(uint32_t in, int pos) {
uint32_t mask = (1 << pos);
uint32_t out = (in | mask);
return out;
}
// Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(),
// but for float.
//
// Inputs:
// - new_value : New additional value.
// - scale : Scale for smoothing (should be less than 1.0).
//
// Input/Output:
// - mean_value : Pointer to the mean value for updating.
//
static void MeanEstimatorFloat(float new_value,
float scale,
float* mean_value) {
RTC_DCHECK_LT(scale, 1.0f);
*mean_value += (new_value - *mean_value) * scale;
}
// Computes the binary spectrum by comparing the input `spectrum` with a
// `threshold_spectrum`. Float and fixed point versions.
//
// Inputs:
// - spectrum : Spectrum of which the binary spectrum should be
// calculated.
// - threshold_spectrum : Threshold spectrum with which the input
// spectrum is compared.
// Return:
// - out : Binary spectrum.
//
static uint32_t BinarySpectrumFix(const uint16_t* spectrum,
SpectrumType* threshold_spectrum,
int q_domain,
int* threshold_initialized) {
int i = kBandFirst;
uint32_t out = 0;
RTC_DCHECK_LT(q_domain, 16);
if (!(*threshold_initialized)) {
// Set the `threshold_spectrum` to half the input `spectrum` as starting
// value. This speeds up the convergence.
for (i = kBandFirst; i <= kBandLast; i++) {
if (spectrum[i] > 0) {
// Convert input spectrum from Q(`q_domain`) to Q15.
int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
*threshold_initialized = 1;
}
}
}
for (i = kBandFirst; i <= kBandLast; i++) {
// Convert input spectrum from Q(`q_domain`) to Q15.
int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
// Update the `threshold_spectrum`.
WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
// Convert `spectrum` at current frequency bin to a binary value.
if (spectrum_q15 > threshold_spectrum[i].int32_) {
out = SetBit(out, i - kBandFirst);
}
}
return out;
}
static uint32_t BinarySpectrumFloat(const float* spectrum,
SpectrumType* threshold_spectrum,
int* threshold_initialized) {
int i = kBandFirst;
uint32_t out = 0;
const float kScale = 1 / 64.0;
if (!(*threshold_initialized)) {
// Set the `threshold_spectrum` to half the input `spectrum` as starting
// value. This speeds up the convergence.
for (i = kBandFirst; i <= kBandLast; i++) {
if (spectrum[i] > 0.0f) {
threshold_spectrum[i].float_ = (spectrum[i] / 2);
*threshold_initialized = 1;
}
}
}
for (i = kBandFirst; i <= kBandLast; i++) {
// Update the `threshold_spectrum`.
MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
// Convert `spectrum` at current frequency bin to a binary value.
if (spectrum[i] > threshold_spectrum[i].float_) {
out = SetBit(out, i - kBandFirst);
}
}
return out;
}
void WebRtc_FreeDelayEstimatorFarend(void* handle) {
DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
if (handle == NULL) {
return;
}
free(self->mean_far_spectrum);
self->mean_far_spectrum = NULL;
WebRtc_FreeBinaryDelayEstimatorFarend(self->binary_farend);
self->binary_farend = NULL;
free(self);
}
void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
DelayEstimatorFarend* self = NULL;
// Check if the sub band used in the delay estimation is small enough to fit
// the binary spectra in a uint32_t.
static_assert(kBandLast - kBandFirst < 32, "");
if (spectrum_size >= kBandLast) {
self = static_cast<DelayEstimatorFarend*>(
malloc(sizeof(DelayEstimatorFarend)));
}
if (self != NULL) {
int memory_fail = 0;
// Allocate memory for the binary far-end spectrum handling.
self->binary_farend = WebRtc_CreateBinaryDelayEstimatorFarend(history_size);
memory_fail |= (self->binary_farend == NULL);
// Allocate memory for spectrum buffers.
self->mean_far_spectrum = static_cast<SpectrumType*>(
malloc(spectrum_size * sizeof(SpectrumType)));
memory_fail |= (self->mean_far_spectrum == NULL);
self->spectrum_size = spectrum_size;
if (memory_fail) {
WebRtc_FreeDelayEstimatorFarend(self);
self = NULL;
}
}
return self;
}
int WebRtc_InitDelayEstimatorFarend(void* handle) {
DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
if (self == NULL) {
return -1;
}
// Initialize far-end part of binary delay estimator.
WebRtc_InitBinaryDelayEstimatorFarend(self->binary_farend);
// Set averaged far and near end spectra to zero.
memset(self->mean_far_spectrum, 0,
sizeof(SpectrumType) * self->spectrum_size);
// Reset initialization indicators.
self->far_spectrum_initialized = 0;
return 0;
}
void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift) {
DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
RTC_DCHECK(self);
WebRtc_SoftResetBinaryDelayEstimatorFarend(self->binary_farend, delay_shift);
}
int WebRtc_AddFarSpectrumFix(void* handle,
const uint16_t* far_spectrum,
int spectrum_size,
int far_q) {
DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
uint32_t binary_spectrum = 0;
if (self == NULL) {
return -1;
}
if (far_spectrum == NULL) {
// Empty far end spectrum.
return -1;
}
if (spectrum_size != self->spectrum_size) {
// Data sizes don't match.
return -1;
}
if (far_q > 15) {
// If `far_q` is larger than 15 we cannot guarantee no wrap around.
return -1;
}
// Get binary spectrum.
binary_spectrum = BinarySpectrumFix(far_spectrum, self->mean_far_spectrum,
far_q, &(self->far_spectrum_initialized));
WebRtc_AddBinaryFarSpectrum(self->binary_farend, binary_spectrum);
return 0;
}
int WebRtc_AddFarSpectrumFloat(void* handle,
const float* far_spectrum,
int spectrum_size) {
DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
uint32_t binary_spectrum = 0;
if (self == NULL) {
return -1;
}
if (far_spectrum == NULL) {
// Empty far end spectrum.
return -1;
}
if (spectrum_size != self->spectrum_size) {
// Data sizes don't match.
return -1;
}
// Get binary spectrum.
binary_spectrum = BinarySpectrumFloat(far_spectrum, self->mean_far_spectrum,
&(self->far_spectrum_initialized));
WebRtc_AddBinaryFarSpectrum(self->binary_farend, binary_spectrum);
return 0;
}
void WebRtc_FreeDelayEstimator(void* handle) {
DelayEstimator* self = (DelayEstimator*)handle;
if (handle == NULL) {
return;
}
free(self->mean_near_spectrum);
self->mean_near_spectrum = NULL;
WebRtc_FreeBinaryDelayEstimator(self->binary_handle);
self->binary_handle = NULL;
free(self);
}
void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
DelayEstimator* self = NULL;
DelayEstimatorFarend* farend = (DelayEstimatorFarend*)farend_handle;
if (farend_handle != NULL) {
self = static_cast<DelayEstimator*>(malloc(sizeof(DelayEstimator)));
}
if (self != NULL) {
int memory_fail = 0;
// Allocate memory for the farend spectrum handling.
self->binary_handle =
WebRtc_CreateBinaryDelayEstimator(farend->binary_farend, max_lookahead);
memory_fail |= (self->binary_handle == NULL);
// Allocate memory for spectrum buffers.
self->mean_near_spectrum = static_cast<SpectrumType*>(
malloc(farend->spectrum_size * sizeof(SpectrumType)));
memory_fail |= (self->mean_near_spectrum == NULL);
self->spectrum_size = farend->spectrum_size;
if (memory_fail) {
WebRtc_FreeDelayEstimator(self);
self = NULL;
}
}
return self;
}
int WebRtc_InitDelayEstimator(void* handle) {
DelayEstimator* self = (DelayEstimator*)handle;
if (self == NULL) {
return -1;
}
// Initialize binary delay estimator.
WebRtc_InitBinaryDelayEstimator(self->binary_handle);
// Set averaged far and near end spectra to zero.
memset(self->mean_near_spectrum, 0,
sizeof(SpectrumType) * self->spectrum_size);
// Reset initialization indicators.
self->near_spectrum_initialized = 0;
return 0;
}
int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) {
DelayEstimator* self = (DelayEstimator*)handle;
RTC_DCHECK(self);
return WebRtc_SoftResetBinaryDelayEstimator(self->binary_handle, delay_shift);
}
int WebRtc_set_history_size(void* handle, int history_size) {
DelayEstimator* self = static_cast<DelayEstimator*>(handle);
if ((self == NULL) || (history_size <= 1)) {
return -1;
}
return WebRtc_AllocateHistoryBufferMemory(self->binary_handle, history_size);
}
int WebRtc_history_size(const void* handle) {
const DelayEstimator* self = static_cast<const DelayEstimator*>(handle);
if (self == NULL) {
return -1;
}
if (self->binary_handle->farend->history_size !=
self->binary_handle->history_size) {
// Non matching history sizes.
return -1;
}
return self->binary_handle->history_size;
}
int WebRtc_set_lookahead(void* handle, int lookahead) {
DelayEstimator* self = (DelayEstimator*)handle;
RTC_DCHECK(self);
RTC_DCHECK(self->binary_handle);
if ((lookahead > self->binary_handle->near_history_size - 1) ||
(lookahead < 0)) {
return -1;
}
self->binary_handle->lookahead = lookahead;
return self->binary_handle->lookahead;
}
int WebRtc_lookahead(void* handle) {
DelayEstimator* self = (DelayEstimator*)handle;
RTC_DCHECK(self);
RTC_DCHECK(self->binary_handle);
return self->binary_handle->lookahead;
}
int WebRtc_set_allowed_offset(void* handle, int allowed_offset) {
DelayEstimator* self = (DelayEstimator*)handle;
if ((self == NULL) || (allowed_offset < 0)) {
return -1;
}
self->binary_handle->allowed_offset = allowed_offset;
return 0;
}
int WebRtc_get_allowed_offset(const void* handle) {
const DelayEstimator* self = (const DelayEstimator*)handle;
if (self == NULL) {
return -1;
}
return self->binary_handle->allowed_offset;
}
int WebRtc_enable_robust_validation(void* handle, int enable) {
DelayEstimator* self = (DelayEstimator*)handle;
if (self == NULL) {
return -1;
}
if ((enable < 0) || (enable > 1)) {
return -1;
}
RTC_DCHECK(self->binary_handle);
self->binary_handle->robust_validation_enabled = enable;
return 0;
}
int WebRtc_is_robust_validation_enabled(const void* handle) {
const DelayEstimator* self = (const DelayEstimator*)handle;
if (self == NULL) {
return -1;
}
return self->binary_handle->robust_validation_enabled;
}
int WebRtc_DelayEstimatorProcessFix(void* handle,
const uint16_t* near_spectrum,
int spectrum_size,
int near_q) {
DelayEstimator* self = (DelayEstimator*)handle;
uint32_t binary_spectrum = 0;
if (self == NULL) {
return -1;
}
if (near_spectrum == NULL) {
// Empty near end spectrum.
return -1;
}
if (spectrum_size != self->spectrum_size) {
// Data sizes don't match.
return -1;
}
if (near_q > 15) {
// If `near_q` is larger than 15 we cannot guarantee no wrap around.
return -1;
}
// Get binary spectra.
binary_spectrum =
BinarySpectrumFix(near_spectrum, self->mean_near_spectrum, near_q,
&(self->near_spectrum_initialized));
return WebRtc_ProcessBinarySpectrum(self->binary_handle, binary_spectrum);
}
int WebRtc_DelayEstimatorProcessFloat(void* handle,
const float* near_spectrum,
int spectrum_size) {
DelayEstimator* self = (DelayEstimator*)handle;
uint32_t binary_spectrum = 0;
if (self == NULL) {
return -1;
}
if (near_spectrum == NULL) {
// Empty near end spectrum.
return -1;
}
if (spectrum_size != self->spectrum_size) {
// Data sizes don't match.
return -1;
}
// Get binary spectrum.
binary_spectrum = BinarySpectrumFloat(near_spectrum, self->mean_near_spectrum,
&(self->near_spectrum_initialized));
return WebRtc_ProcessBinarySpectrum(self->binary_handle, binary_spectrum);
}
int WebRtc_last_delay(void* handle) {
DelayEstimator* self = (DelayEstimator*)handle;
if (self == NULL) {
return -1;
}
return WebRtc_binary_last_delay(self->binary_handle);
}
float WebRtc_last_delay_quality(void* handle) {
DelayEstimator* self = (DelayEstimator*)handle;
RTC_DCHECK(self);
return WebRtc_binary_last_delay_quality(self->binary_handle);
}
} // namespace webrtc

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// Performs delay estimation on block by block basis.
// The return value is 0 - OK and -1 - Error, unless otherwise stated.
#ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
#define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
#include <stdint.h>
namespace webrtc {
// Releases the memory allocated by WebRtc_CreateDelayEstimatorFarend(...)
void WebRtc_FreeDelayEstimatorFarend(void* handle);
// Allocates the memory needed by the far-end part of the delay estimation. The
// memory needs to be initialized separately through
// WebRtc_InitDelayEstimatorFarend(...).
//
// Inputs:
// - spectrum_size : Size of the spectrum used both in far-end and
// near-end. Used to allocate memory for spectrum
// specific buffers.
// - history_size : The far-end history buffer size. A change in buffer
// size can be forced with WebRtc_set_history_size().
// Note that the maximum delay which can be estimated is
// determined together with WebRtc_set_lookahead().
//
// Return value:
// - void* : Created `handle`. If the memory can't be allocated or
// if any of the input parameters are invalid NULL is
// returned.
void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size);
// Initializes the far-end part of the delay estimation instance returned by
// WebRtc_CreateDelayEstimatorFarend(...)
int WebRtc_InitDelayEstimatorFarend(void* handle);
// Soft resets the far-end part of the delay estimation instance returned by
// WebRtc_CreateDelayEstimatorFarend(...).
// Input:
// - delay_shift : The amount of blocks to shift history buffers.
void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift);
// Adds the far-end spectrum to the far-end history buffer. This spectrum is
// used as reference when calculating the delay using
// WebRtc_ProcessSpectrum().
//
// Inputs:
// - far_spectrum : Far-end spectrum.
// - spectrum_size : The size of the data arrays (same for both far- and
// near-end).
// - far_q : The Q-domain of the far-end data.
//
// Output:
// - handle : Updated far-end instance.
//
int WebRtc_AddFarSpectrumFix(void* handle,
const uint16_t* far_spectrum,
int spectrum_size,
int far_q);
// See WebRtc_AddFarSpectrumFix() for description.
int WebRtc_AddFarSpectrumFloat(void* handle,
const float* far_spectrum,
int spectrum_size);
// Releases the memory allocated by WebRtc_CreateDelayEstimator(...)
void WebRtc_FreeDelayEstimator(void* handle);
// Allocates the memory needed by the delay estimation. The memory needs to be
// initialized separately through WebRtc_InitDelayEstimator(...).
//
// Inputs:
// - farend_handle : Pointer to the far-end part of the delay estimation
// instance created prior to this call using
// WebRtc_CreateDelayEstimatorFarend().
//
// Note that WebRtc_CreateDelayEstimator does not take
// ownership of `farend_handle`, which has to be torn
// down properly after this instance.
//
// - max_lookahead : Maximum amount of non-causal lookahead allowed. The
// actual amount of lookahead used can be controlled by
// WebRtc_set_lookahead(...). The default `lookahead` is
// set to `max_lookahead` at create time. Use
// WebRtc_set_lookahead(...) before start if a different
// value is desired.
//
// Using lookahead can detect cases in which a near-end
// signal occurs before the corresponding far-end signal.
// It will delay the estimate for the current block by an
// equal amount, and the returned values will be offset
// by it.
//
// A value of zero is the typical no-lookahead case.
// This also represents the minimum delay which can be
// estimated.
//
// Note that the effective range of delay estimates is
// [-`lookahead`,... ,`history_size`-`lookahead`)
// where `history_size` is set through
// WebRtc_set_history_size().
//
// Return value:
// - void* : Created `handle`. If the memory can't be allocated or
// if any of the input parameters are invalid NULL is
// returned.
void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead);
// Initializes the delay estimation instance returned by
// WebRtc_CreateDelayEstimator(...)
int WebRtc_InitDelayEstimator(void* handle);
// Soft resets the delay estimation instance returned by
// WebRtc_CreateDelayEstimator(...)
// Input:
// - delay_shift : The amount of blocks to shift history buffers.
//
// Return value:
// - actual_shifts : The actual number of shifts performed.
int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift);
// Sets the effective `history_size` used. Valid values from 2. We simply need
// at least two delays to compare to perform an estimate. If `history_size` is
// changed, buffers are reallocated filling in with zeros if necessary.
// Note that changing the `history_size` affects both buffers in far-end and
// near-end. Hence it is important to change all DelayEstimators that use the
// same reference far-end, to the same `history_size` value.
// Inputs:
// - handle : Pointer to the delay estimation instance.
// - history_size : Effective history size to be used.
// Return value:
// - new_history_size : The new history size used. If the memory was not able
// to be allocated 0 is returned.
int WebRtc_set_history_size(void* handle, int history_size);
// Returns the history_size currently used.
// Input:
// - handle : Pointer to the delay estimation instance.
int WebRtc_history_size(const void* handle);
// Sets the amount of `lookahead` to use. Valid values are [0, max_lookahead]
// where `max_lookahead` was set at create time through
// WebRtc_CreateDelayEstimator(...).
//
// Input:
// - handle : Pointer to the delay estimation instance.
// - lookahead : The amount of lookahead to be used.
//
// Return value:
// - new_lookahead : The actual amount of lookahead set, unless `handle` is
// a NULL pointer or `lookahead` is invalid, for which an
// error is returned.
int WebRtc_set_lookahead(void* handle, int lookahead);
// Returns the amount of lookahead we currently use.
// Input:
// - handle : Pointer to the delay estimation instance.
int WebRtc_lookahead(void* handle);
// Sets the `allowed_offset` used in the robust validation scheme. If the
// delay estimator is used in an echo control component, this parameter is
// related to the filter length. In principle `allowed_offset` should be set to
// the echo control filter length minus the expected echo duration, i.e., the
// delay offset the echo control can handle without quality regression. The
// default value, used if not set manually, is zero. Note that `allowed_offset`
// has to be non-negative.
// Inputs:
// - handle : Pointer to the delay estimation instance.
// - allowed_offset : The amount of delay offset, measured in partitions,
// the echo control filter can handle.
int WebRtc_set_allowed_offset(void* handle, int allowed_offset);
// Returns the `allowed_offset` in number of partitions.
int WebRtc_get_allowed_offset(const void* handle);
// Enables/Disables a robust validation functionality in the delay estimation.
// This is by default set to disabled at create time. The state is preserved
// over a reset.
// Inputs:
// - handle : Pointer to the delay estimation instance.
// - enable : Enable (1) or disable (0) this feature.
int WebRtc_enable_robust_validation(void* handle, int enable);
// Returns 1 if robust validation is enabled and 0 if disabled.
int WebRtc_is_robust_validation_enabled(const void* handle);
// Estimates and returns the delay between the far-end and near-end blocks. The
// value will be offset by the lookahead (i.e. the lookahead should be
// subtracted from the returned value).
// Inputs:
// - handle : Pointer to the delay estimation instance.
// - near_spectrum : Pointer to the near-end spectrum data of the current
// block.
// - spectrum_size : The size of the data arrays (same for both far- and
// near-end).
// - near_q : The Q-domain of the near-end data.
//
// Output:
// - handle : Updated instance.
//
// Return value:
// - delay : >= 0 - Calculated delay value.
// -1 - Error.
// -2 - Insufficient data for estimation.
int WebRtc_DelayEstimatorProcessFix(void* handle,
const uint16_t* near_spectrum,
int spectrum_size,
int near_q);
// See WebRtc_DelayEstimatorProcessFix() for description.
int WebRtc_DelayEstimatorProcessFloat(void* handle,
const float* near_spectrum,
int spectrum_size);
// Returns the last calculated delay updated by the function
// WebRtc_DelayEstimatorProcess(...).
//
// Input:
// - handle : Pointer to the delay estimation instance.
//
// Return value:
// - delay : >= 0 - Last calculated delay value.
// -1 - Error.
// -2 - Insufficient data for estimation.
int WebRtc_last_delay(void* handle);
// Returns the estimation quality/probability of the last calculated delay
// updated by the function WebRtc_DelayEstimatorProcess(...). The estimation
// quality is a value in the interval [0, 1]. The higher the value, the better
// the quality.
//
// Return value:
// - delay_quality : >= 0 - Estimation quality of last calculated delay.
float WebRtc_last_delay_quality(void* handle);
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef RTC_BASE_SANITIZER_H_
#define RTC_BASE_SANITIZER_H_
#include <stddef.h> // For size_t.
#ifdef __cplusplus
#include "absl/meta/type_traits.h"
#endif
#if defined(__has_feature)
#if __has_feature(address_sanitizer)
#define RTC_HAS_ASAN 1
#endif
#if __has_feature(memory_sanitizer)
#define RTC_HAS_MSAN 1
#endif
#endif
#ifndef RTC_HAS_ASAN
#define RTC_HAS_ASAN 0
#endif
#ifndef RTC_HAS_MSAN
#define RTC_HAS_MSAN 0
#endif
#if RTC_HAS_ASAN
#include <sanitizer/asan_interface.h>
#endif
#if RTC_HAS_MSAN
#include <sanitizer/msan_interface.h>
#endif
#ifdef __has_attribute
#if __has_attribute(no_sanitize)
#define RTC_NO_SANITIZE(what) __attribute__((no_sanitize(what)))
#endif
#endif
#ifndef RTC_NO_SANITIZE
#define RTC_NO_SANITIZE(what)
#endif
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being unaddressable, so that reads and writes are not allowed. ASan may
// narrow the range to the nearest alignment boundaries.
static inline void rtc_AsanPoison(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
}
// Ask ASan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being addressable, so that reads and writes are allowed. ASan may widen
// the range to the nearest alignment boundaries.
static inline void rtc_AsanUnpoison(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_ASAN
ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
#endif
}
// Ask MSan to mark the memory range [ptr, ptr + element_size * num_elements)
// as being uninitialized.
static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_poison(ptr, element_size * num_elements);
#endif
}
// Force an MSan check (if any bits in the memory range [ptr, ptr +
// element_size * num_elements) are uninitialized the call will crash with an
// MSan report).
static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
size_t element_size,
size_t num_elements) {
#if RTC_HAS_MSAN
__msan_check_mem_is_initialized(ptr, element_size * num_elements);
#endif
}
#ifdef __cplusplus
namespace rtc {
namespace sanitizer_impl {
template <typename T>
constexpr bool IsTriviallyCopyable() {
return static_cast<bool>(absl::is_trivially_copy_constructible<T>::value &&
(absl::is_trivially_copy_assignable<T>::value ||
!std::is_copy_assignable<T>::value) &&
absl::is_trivially_destructible<T>::value);
}
} // namespace sanitizer_impl
template <typename T>
inline void AsanPoison(const T& mem) {
rtc_AsanPoison(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline void AsanUnpoison(const T& mem) {
rtc_AsanUnpoison(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline void MsanMarkUninitialized(const T& mem) {
rtc_MsanMarkUninitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
}
template <typename T>
inline T MsanUninitialized(T t) {
#if RTC_HAS_MSAN
// TODO(bugs.webrtc.org/8762): Switch to std::is_trivially_copyable when it
// becomes available in downstream projects.
static_assert(sanitizer_impl::IsTriviallyCopyable<T>(), "");
#endif
rtc_MsanMarkUninitialized(&t, sizeof(T), 1);
return t;
}
template <typename T>
inline void MsanCheckInitialized(const T& mem) {
rtc_MsanCheckInitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
}
} // namespace rtc
#endif // __cplusplus
#endif // RTC_BASE_SANITIZER_H_