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-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c103
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/auto_correlation.c65
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse.c108
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_arm.S119
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_mips.c176
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_fft.c299
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_fft_mips.c328
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/complex_fft_tables.h132
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/copy_set_operations.c82
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/cross_correlation.c30
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/cross_correlation_mips.c104
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/cross_correlation_neon.c88
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/division_operations.c140
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.cc34
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.h40
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/downsample_fast.c65
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/downsample_fast_mips.c169
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/downsample_fast_neon.c217
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/energy.c39
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/filter_ar.c95
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c47
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S218
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c140
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/filter_ma_fast_q12.c55
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/get_hanning_window.c77
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/get_scaling_square.c46
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/ilbc_specific_functions.c90
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/include/real_fft.h96
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/include/signal_processing_library.h1635
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/include/spl_inl.h155
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_armv7.h138
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_mips.h204
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/levinson_durbin.c249
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/lpc_to_refl_coef.c56
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/min_max_operations.c256
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/min_max_operations_mips.c375
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/min_max_operations_neon.c333
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/randomization_functions.c115
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/real_fft.c102
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/real_fft_unittest.cc98
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/refl_coef_to_lpc.c59
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample.c505
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_48khz.c186
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_by_2.c183
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.c689
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.h60
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_by_2_mips.c292
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/resample_fractional.c239
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/signal_processing_unittest.cc668
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/spl_init.c69
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/spl_inl.c24
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/spl_sqrt.c194
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/splitting_filter.c211
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c35
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations.c165
-rw-r--r--third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations_mips.c57
56 files changed, 10554 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c b/third_party/libwebrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
new file mode 100644
index 0000000000..a3ec24f5da
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c
@@ -0,0 +1,103 @@
+/*
+ * 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_AutoCorrToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_AutoCorrToReflCoef(const int32_t *R, int use_order, int16_t *K)
+{
+ int i, n;
+ int16_t tmp;
+ const int32_t *rptr;
+ int32_t L_num, L_den;
+ int16_t *acfptr, *pptr, *wptr, *p1ptr, *w1ptr, ACF[WEBRTC_SPL_MAX_LPC_ORDER],
+ P[WEBRTC_SPL_MAX_LPC_ORDER], W[WEBRTC_SPL_MAX_LPC_ORDER];
+
+ // Initialize loop and pointers.
+ acfptr = ACF;
+ rptr = R;
+ pptr = P;
+ p1ptr = &P[1];
+ w1ptr = &W[1];
+ wptr = w1ptr;
+
+ // First loop; n=0. Determine shifting.
+ tmp = WebRtcSpl_NormW32(*R);
+ *acfptr = (int16_t)((*rptr++ << tmp) >> 16);
+ *pptr++ = *acfptr++;
+
+ // Initialize ACF, P and W.
+ for (i = 1; i <= use_order; i++)
+ {
+ *acfptr = (int16_t)((*rptr++ << tmp) >> 16);
+ *wptr++ = *acfptr;
+ *pptr++ = *acfptr++;
+ }
+
+ // Compute reflection coefficients.
+ for (n = 1; n <= use_order; n++, K++)
+ {
+ tmp = WEBRTC_SPL_ABS_W16(*p1ptr);
+ if (*P < tmp)
+ {
+ for (i = n; i <= use_order; i++)
+ *K++ = 0;
+
+ return;
+ }
+
+ // Division: WebRtcSpl_div(tmp, *P)
+ *K = 0;
+ if (tmp != 0)
+ {
+ L_num = tmp;
+ L_den = *P;
+ i = 15;
+ while (i--)
+ {
+ (*K) <<= 1;
+ L_num <<= 1;
+ if (L_num >= L_den)
+ {
+ L_num -= L_den;
+ (*K)++;
+ }
+ }
+ if (*p1ptr > 0)
+ *K = -*K;
+ }
+
+ // Last iteration; don't do Schur recursion.
+ if (n == use_order)
+ return;
+
+ // Schur recursion.
+ pptr = P;
+ wptr = w1ptr;
+ tmp = (int16_t)(((int32_t)*p1ptr * (int32_t)*K + 16384) >> 15);
+ *pptr = WebRtcSpl_AddSatW16(*pptr, tmp);
+ pptr++;
+ for (i = 1; i <= use_order - n; i++)
+ {
+ tmp = (int16_t)(((int32_t)*wptr * (int32_t)*K + 16384) >> 15);
+ *pptr = WebRtcSpl_AddSatW16(*(pptr + 1), tmp);
+ pptr++;
+ tmp = (int16_t)(((int32_t)*pptr * (int32_t)*K + 16384) >> 15);
+ *wptr = WebRtcSpl_AddSatW16(*wptr, tmp);
+ wptr++;
+ }
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/auto_correlation.c b/third_party/libwebrtc/common_audio/signal_processing/auto_correlation.c
new file mode 100644
index 0000000000..1455820e8f
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/auto_correlation.c
@@ -0,0 +1,65 @@
+/*
+ * 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"
+
+size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
+ size_t in_vector_length,
+ size_t order,
+ int32_t* result,
+ int* scale) {
+ int32_t sum = 0;
+ size_t i = 0, j = 0;
+ int16_t smax = 0;
+ int scaling = 0;
+
+ RTC_DCHECK_LE(order, in_vector_length);
+
+ // Find the maximum absolute value of the samples.
+ smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
+
+ // In order to avoid overflow when computing the sum we should scale the
+ // samples so that (in_vector_length * smax * smax) will not overflow.
+ if (smax == 0) {
+ scaling = 0;
+ } else {
+ // Number of bits in the sum loop.
+ int nbits = WebRtcSpl_GetSizeInBits((uint32_t)in_vector_length);
+ // Number of bits to normalize smax.
+ int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+ if (t > nbits) {
+ scaling = 0;
+ } else {
+ scaling = nbits - t;
+ }
+ }
+
+ // Perform the actual correlation calculation.
+ for (i = 0; i < order + 1; i++) {
+ sum = 0;
+ /* Unroll the loop to improve performance. */
+ for (j = 0; i + j + 3 < in_vector_length; j += 4) {
+ sum += (in_vector[j + 0] * in_vector[i + j + 0]) >> scaling;
+ sum += (in_vector[j + 1] * in_vector[i + j + 1]) >> scaling;
+ sum += (in_vector[j + 2] * in_vector[i + j + 2]) >> scaling;
+ sum += (in_vector[j + 3] * in_vector[i + j + 3]) >> scaling;
+ }
+ for (; j < in_vector_length - i; j++) {
+ sum += (in_vector[j] * in_vector[i + j]) >> scaling;
+ }
+ *result++ = sum;
+ }
+
+ *scale = scaling;
+ return order + 1;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse.c b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse.c
new file mode 100644
index 0000000000..1c82cff50f
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse.c
@@ -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;
+ }
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_arm.S b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
new file mode 100644
index 0000000000..be8e181aa7
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
@@ -0,0 +1,119 @@
+@
+@ 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 function WebRtcSpl_ComplexBitReverse(), optimized
+@ for ARMv5 platforms.
+@ Reference C code is in file complex_bit_reverse.c. Bit-exact.
+
+#include "rtc_base/system/asm_defines.h"
+
+GLOBAL_FUNCTION WebRtcSpl_ComplexBitReverse
+.align 2
+DEFINE_FUNCTION WebRtcSpl_ComplexBitReverse
+ push {r4-r7}
+
+ cmp r1, #7
+ adr r3, index_7 @ Table pointer.
+ mov r4, #112 @ Number of interations.
+ beq PRE_LOOP_STAGES_7_OR_8
+
+ cmp r1, #8
+ adr r3, index_8 @ Table pointer.
+ mov r4, #240 @ Number of interations.
+ beq PRE_LOOP_STAGES_7_OR_8
+
+ mov r3, #1 @ Initialize m.
+ mov r1, r3, asl r1 @ n = 1 << stages;
+ subs r6, r1, #1 @ nn = n - 1;
+ ble END
+
+ mov r5, r0 @ &complex_data
+ mov r4, #0 @ ml
+
+LOOP_GENERIC:
+ rsb r12, r4, r6 @ l > nn - mr
+ mov r2, r1 @ n
+
+LOOP_SHIFT:
+ asr r2, #1 @ l >>= 1;
+ cmp r2, r12
+ bgt LOOP_SHIFT
+
+ sub r12, r2, #1
+ and r4, r12, r4
+ add r4, r2 @ mr = (mr & (l - 1)) + l;
+ cmp r4, r3 @ mr <= m ?
+ ble UPDATE_REGISTERS
+
+ mov r12, r4, asl #2
+ ldr r7, [r5, #4] @ complex_data[2 * m, 2 * m + 1].
+ @ Offset 4 due to m incrementing from 1.
+ ldr r2, [r0, r12] @ complex_data[2 * mr, 2 * mr + 1].
+ str r7, [r0, r12]
+ str r2, [r5, #4]
+
+UPDATE_REGISTERS:
+ add r3, r3, #1
+ add r5, #4
+ cmp r3, r1
+ bne LOOP_GENERIC
+
+ b END
+
+PRE_LOOP_STAGES_7_OR_8:
+ add r4, r3, r4, asl #1
+
+LOOP_STAGES_7_OR_8:
+ ldrsh r2, [r3], #2 @ index[m]
+ ldrsh r5, [r3], #2 @ index[m + 1]
+ ldr r1, [r0, r2] @ complex_data[index[m], index[m] + 1]
+ ldr r12, [r0, r5] @ complex_data[index[m + 1], index[m + 1] + 1]
+ cmp r3, r4
+ str r1, [r0, r5]
+ str r12, [r0, r2]
+ bne LOOP_STAGES_7_OR_8
+
+END:
+ pop {r4-r7}
+ bx lr
+
+@ The index tables. Note the values are doubles of the actual indexes for 16-bit
+@ elements, different from the generic C code. It actually provides byte offsets
+@ for the indexes.
+
+.align 2
+index_7: @ Indexes for stages == 7.
+ .short 4, 256, 8, 128, 12, 384, 16, 64, 20, 320, 24, 192, 28, 448, 36, 288
+ .short 40, 160, 44, 416, 48, 96, 52, 352, 56, 224, 60, 480, 68, 272, 72, 144
+ .short 76, 400, 84, 336, 88, 208, 92, 464, 100, 304, 104, 176, 108, 432, 116
+ .short 368, 120, 240, 124, 496, 132, 264, 140, 392, 148, 328, 152, 200, 156
+ .short 456, 164, 296, 172, 424, 180, 360, 184, 232, 188, 488, 196, 280, 204
+ .short 408, 212, 344, 220, 472, 228, 312, 236, 440, 244, 376, 252, 504, 268
+ .short 388, 276, 324, 284, 452, 300, 420, 308, 356, 316, 484, 332, 404, 348
+ .short 468, 364, 436, 380, 500, 412, 460, 444, 492
+
+index_8: @ Indexes for stages == 8.
+ .short 4, 512, 8, 256, 12, 768, 16, 128, 20, 640, 24, 384, 28, 896, 32, 64
+ .short 36, 576, 40, 320, 44, 832, 48, 192, 52, 704, 56, 448, 60, 960, 68, 544
+ .short 72, 288, 76, 800, 80, 160, 84, 672, 88, 416, 92, 928, 100, 608, 104
+ .short 352, 108, 864, 112, 224, 116, 736, 120, 480, 124, 992, 132, 528, 136
+ .short 272, 140, 784, 148, 656, 152, 400, 156, 912, 164, 592, 168, 336, 172
+ .short 848, 176, 208, 180, 720, 184, 464, 188, 976, 196, 560, 200, 304, 204
+ .short 816, 212, 688, 216, 432, 220, 944, 228, 624, 232, 368, 236, 880, 244
+ .short 752, 248, 496, 252, 1008, 260, 520, 268, 776, 276, 648, 280, 392, 284
+ .short 904, 292, 584, 296, 328, 300, 840, 308, 712, 312, 456, 316, 968, 324
+ .short 552, 332, 808, 340, 680, 344, 424, 348, 936, 356, 616, 364, 872, 372
+ .short 744, 376, 488, 380, 1000, 388, 536, 396, 792, 404, 664, 412, 920, 420
+ .short 600, 428, 856, 436, 728, 440, 472, 444, 984, 452, 568, 460, 824, 468
+ .short 696, 476, 952, 484, 632, 492, 888, 500, 760, 508, 1016, 524, 772, 532
+ .short 644, 540, 900, 548, 580, 556, 836, 564, 708, 572, 964, 588, 804, 596
+ .short 676, 604, 932, 620, 868, 628, 740, 636, 996, 652, 788, 668, 916, 684
+ .short 852, 692, 724, 700, 980, 716, 820, 732, 948, 748, 884, 764, 1012, 796
+ .short 908, 812, 844, 828, 972, 860, 940, 892, 1004, 956, 988
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_mips.c b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_mips.c
new file mode 100644
index 0000000000..9007b19cf6
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_bit_reverse_mips.c
@@ -0,0 +1,176 @@
+/*
+ * 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 "common_audio/signal_processing/include/signal_processing_library.h"
+
+static int16_t coefTable_7[] = {
+ 4, 256, 8, 128, 12, 384, 16, 64,
+ 20, 320, 24, 192, 28, 448, 36, 288,
+ 40, 160, 44, 416, 48, 96, 52, 352,
+ 56, 224, 60, 480, 68, 272, 72, 144,
+ 76, 400, 84, 336, 88, 208, 92, 464,
+ 100, 304, 104, 176, 108, 432, 116, 368,
+ 120, 240, 124, 496, 132, 264, 140, 392,
+ 148, 328, 152, 200, 156, 456, 164, 296,
+ 172, 424, 180, 360, 184, 232, 188, 488,
+ 196, 280, 204, 408, 212, 344, 220, 472,
+ 228, 312, 236, 440, 244, 376, 252, 504,
+ 268, 388, 276, 324, 284, 452, 300, 420,
+ 308, 356, 316, 484, 332, 404, 348, 468,
+ 364, 436, 380, 500, 412, 460, 444, 492
+};
+
+static int16_t coefTable_8[] = {
+ 4, 512, 8, 256, 12, 768, 16, 128,
+ 20, 640, 24, 384, 28, 896, 32, 64,
+ 36, 576, 40, 320, 44, 832, 48, 192,
+ 52, 704, 56, 448, 60, 960, 68, 544,
+ 72, 288, 76, 800, 80, 160, 84, 672,
+ 88, 416, 92, 928, 100, 608, 104, 352,
+ 108, 864, 112, 224, 116, 736, 120, 480,
+ 124, 992, 132, 528, 136, 272, 140, 784,
+ 148, 656, 152, 400, 156, 912, 164, 592,
+ 168, 336, 172, 848, 176, 208, 180, 720,
+ 184, 464, 188, 976, 196, 560, 200, 304,
+ 204, 816, 212, 688, 216, 432, 220, 944,
+ 228, 624, 232, 368, 236, 880, 244, 752,
+ 248, 496, 252, 1008, 260, 520, 268, 776,
+ 276, 648, 280, 392, 284, 904, 292, 584,
+ 296, 328, 300, 840, 308, 712, 312, 456,
+ 316, 968, 324, 552, 332, 808, 340, 680,
+ 344, 424, 348, 936, 356, 616, 364, 872,
+ 372, 744, 376, 488, 380, 1000, 388, 536,
+ 396, 792, 404, 664, 412, 920, 420, 600,
+ 428, 856, 436, 728, 440, 472, 444, 984,
+ 452, 568, 460, 824, 468, 696, 476, 952,
+ 484, 632, 492, 888, 500, 760, 508, 1016,
+ 524, 772, 532, 644, 540, 900, 548, 580,
+ 556, 836, 564, 708, 572, 964, 588, 804,
+ 596, 676, 604, 932, 620, 868, 628, 740,
+ 636, 996, 652, 788, 668, 916, 684, 852,
+ 692, 724, 700, 980, 716, 820, 732, 948,
+ 748, 884, 764, 1012, 796, 908, 812, 844,
+ 828, 972, 860, 940, 892, 1004, 956, 988
+};
+
+void WebRtcSpl_ComplexBitReverse(int16_t frfi[], int stages) {
+ int l;
+ int16_t tr, ti;
+ int32_t tmp1, tmp2, tmp3, tmp4;
+ int32_t* ptr_i;
+ int32_t* ptr_j;
+
+ if (stages == 8) {
+ int16_t* pcoeftable_8 = coefTable_8;
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "addiu %[l], $zero, 120 \n\t"
+ "1: \n\t"
+ "addiu %[l], %[l], -4 \n\t"
+ "lh %[tr], 0(%[pcoeftable_8]) \n\t"
+ "lh %[ti], 2(%[pcoeftable_8]) \n\t"
+ "lh %[tmp3], 4(%[pcoeftable_8]) \n\t"
+ "lh %[tmp4], 6(%[pcoeftable_8]) \n\t"
+ "addu %[ptr_i], %[frfi], %[tr] \n\t"
+ "addu %[ptr_j], %[frfi], %[ti] \n\t"
+ "addu %[tr], %[frfi], %[tmp3] \n\t"
+ "addu %[ti], %[frfi], %[tmp4] \n\t"
+ "ulw %[tmp1], 0(%[ptr_i]) \n\t"
+ "ulw %[tmp2], 0(%[ptr_j]) \n\t"
+ "ulw %[tmp3], 0(%[tr]) \n\t"
+ "ulw %[tmp4], 0(%[ti]) \n\t"
+ "usw %[tmp1], 0(%[ptr_j]) \n\t"
+ "usw %[tmp2], 0(%[ptr_i]) \n\t"
+ "usw %[tmp4], 0(%[tr]) \n\t"
+ "usw %[tmp3], 0(%[ti]) \n\t"
+ "lh %[tmp1], 8(%[pcoeftable_8]) \n\t"
+ "lh %[tmp2], 10(%[pcoeftable_8]) \n\t"
+ "lh %[tr], 12(%[pcoeftable_8]) \n\t"
+ "lh %[ti], 14(%[pcoeftable_8]) \n\t"
+ "addu %[ptr_i], %[frfi], %[tmp1] \n\t"
+ "addu %[ptr_j], %[frfi], %[tmp2] \n\t"
+ "addu %[tr], %[frfi], %[tr] \n\t"
+ "addu %[ti], %[frfi], %[ti] \n\t"
+ "ulw %[tmp1], 0(%[ptr_i]) \n\t"
+ "ulw %[tmp2], 0(%[ptr_j]) \n\t"
+ "ulw %[tmp3], 0(%[tr]) \n\t"
+ "ulw %[tmp4], 0(%[ti]) \n\t"
+ "usw %[tmp1], 0(%[ptr_j]) \n\t"
+ "usw %[tmp2], 0(%[ptr_i]) \n\t"
+ "usw %[tmp4], 0(%[tr]) \n\t"
+ "usw %[tmp3], 0(%[ti]) \n\t"
+ "bgtz %[l], 1b \n\t"
+ " addiu %[pcoeftable_8], %[pcoeftable_8], 16 \n\t"
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [ptr_i] "=&r" (ptr_i),
+ [ptr_j] "=&r" (ptr_j), [tr] "=&r" (tr), [l] "=&r" (l),
+ [tmp3] "=&r" (tmp3), [pcoeftable_8] "+r" (pcoeftable_8),
+ [ti] "=&r" (ti), [tmp4] "=&r" (tmp4)
+ : [frfi] "r" (frfi)
+ : "memory"
+ );
+ } else if (stages == 7) {
+ int16_t* pcoeftable_7 = coefTable_7;
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "addiu %[l], $zero, 56 \n\t"
+ "1: \n\t"
+ "addiu %[l], %[l], -4 \n\t"
+ "lh %[tr], 0(%[pcoeftable_7]) \n\t"
+ "lh %[ti], 2(%[pcoeftable_7]) \n\t"
+ "lh %[tmp3], 4(%[pcoeftable_7]) \n\t"
+ "lh %[tmp4], 6(%[pcoeftable_7]) \n\t"
+ "addu %[ptr_i], %[frfi], %[tr] \n\t"
+ "addu %[ptr_j], %[frfi], %[ti] \n\t"
+ "addu %[tr], %[frfi], %[tmp3] \n\t"
+ "addu %[ti], %[frfi], %[tmp4] \n\t"
+ "ulw %[tmp1], 0(%[ptr_i]) \n\t"
+ "ulw %[tmp2], 0(%[ptr_j]) \n\t"
+ "ulw %[tmp3], 0(%[tr]) \n\t"
+ "ulw %[tmp4], 0(%[ti]) \n\t"
+ "usw %[tmp1], 0(%[ptr_j]) \n\t"
+ "usw %[tmp2], 0(%[ptr_i]) \n\t"
+ "usw %[tmp4], 0(%[tr]) \n\t"
+ "usw %[tmp3], 0(%[ti]) \n\t"
+ "lh %[tmp1], 8(%[pcoeftable_7]) \n\t"
+ "lh %[tmp2], 10(%[pcoeftable_7]) \n\t"
+ "lh %[tr], 12(%[pcoeftable_7]) \n\t"
+ "lh %[ti], 14(%[pcoeftable_7]) \n\t"
+ "addu %[ptr_i], %[frfi], %[tmp1] \n\t"
+ "addu %[ptr_j], %[frfi], %[tmp2] \n\t"
+ "addu %[tr], %[frfi], %[tr] \n\t"
+ "addu %[ti], %[frfi], %[ti] \n\t"
+ "ulw %[tmp1], 0(%[ptr_i]) \n\t"
+ "ulw %[tmp2], 0(%[ptr_j]) \n\t"
+ "ulw %[tmp3], 0(%[tr]) \n\t"
+ "ulw %[tmp4], 0(%[ti]) \n\t"
+ "usw %[tmp1], 0(%[ptr_j]) \n\t"
+ "usw %[tmp2], 0(%[ptr_i]) \n\t"
+ "usw %[tmp4], 0(%[tr]) \n\t"
+ "usw %[tmp3], 0(%[ti]) \n\t"
+ "bgtz %[l], 1b \n\t"
+ " addiu %[pcoeftable_7], %[pcoeftable_7], 16 \n\t"
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [ptr_i] "=&r" (ptr_i),
+ [ptr_j] "=&r" (ptr_j), [ti] "=&r" (ti), [tr] "=&r" (tr),
+ [l] "=&r" (l), [pcoeftable_7] "+r" (pcoeftable_7),
+ [tmp3] "=&r" (tmp3), [tmp4] "=&r" (tmp4)
+ : [frfi] "r" (frfi)
+ : "memory"
+ );
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_fft.c b/third_party/libwebrtc/common_audio/signal_processing/complex_fft.c
new file mode 100644
index 0000000000..ddc9a97b59
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_fft.c
@@ -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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_fft_mips.c b/third_party/libwebrtc/common_audio/signal_processing/complex_fft_mips.c
new file mode 100644
index 0000000000..27071f8b39
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_fft_mips.c
@@ -0,0 +1,328 @@
+/*
+ * 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 "common_audio/signal_processing/complex_fft_tables.h"
+#include "common_audio/signal_processing/include/signal_processing_library.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 = 0;
+ int l = 0;
+ int k = 0;
+ int istep = 0;
+ int n = 0;
+ int m = 0;
+ int32_t wr = 0, wi = 0;
+ int32_t tmp1 = 0;
+ int32_t tmp2 = 0;
+ int32_t tmp3 = 0;
+ int32_t tmp4 = 0;
+ int32_t tmp5 = 0;
+ int32_t tmp6 = 0;
+ int32_t tmp = 0;
+ int16_t* ptr_j = NULL;
+ int16_t* ptr_i = NULL;
+
+ n = 1 << stages;
+ if (n > 1024) {
+ return -1;
+ }
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "addiu %[k], $zero, 10 \n\t"
+ "addiu %[l], $zero, 1 \n\t"
+ "3: \n\t"
+ "sll %[istep], %[l], 1 \n\t"
+ "move %[m], $zero \n\t"
+ "sll %[tmp], %[l], 2 \n\t"
+ "move %[i], $zero \n\t"
+ "2: \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "sllv %[tmp3], %[m], %[k] \n\t"
+ "addiu %[tmp2], %[tmp3], 512 \n\t"
+ "addiu %[m], %[m], 1 \n\t"
+ "lhx %[wi], %[tmp3](%[kSinTable1024]) \n\t"
+ "lhx %[wr], %[tmp2](%[kSinTable1024]) \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "sllv %[tmp3], %[m], %[k] \n\t"
+ "addu %[ptr_j], %[tmp3], %[kSinTable1024] \n\t"
+ "addiu %[ptr_i], %[ptr_j], 512 \n\t"
+ "addiu %[m], %[m], 1 \n\t"
+ "lh %[wi], 0(%[ptr_j]) \n\t"
+ "lh %[wr], 0(%[ptr_i]) \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "1: \n\t"
+ "sll %[tmp1], %[i], 2 \n\t"
+ "addu %[ptr_i], %[frfi], %[tmp1] \n\t"
+ "addu %[ptr_j], %[ptr_i], %[tmp] \n\t"
+ "lh %[tmp6], 0(%[ptr_i]) \n\t"
+ "lh %[tmp5], 2(%[ptr_i]) \n\t"
+ "lh %[tmp3], 0(%[ptr_j]) \n\t"
+ "lh %[tmp4], 2(%[ptr_j]) \n\t"
+ "addu %[i], %[i], %[istep] \n\t"
+#if defined(MIPS_DSP_R2_LE)
+ "mult %[wr], %[tmp3] \n\t"
+ "madd %[wi], %[tmp4] \n\t"
+ "mult $ac1, %[wr], %[tmp4] \n\t"
+ "msub $ac1, %[wi], %[tmp3] \n\t"
+ "mflo %[tmp1] \n\t"
+ "mflo %[tmp2], $ac1 \n\t"
+ "sll %[tmp6], %[tmp6], 14 \n\t"
+ "sll %[tmp5], %[tmp5], 14 \n\t"
+ "shra_r.w %[tmp1], %[tmp1], 1 \n\t"
+ "shra_r.w %[tmp2], %[tmp2], 1 \n\t"
+ "subu %[tmp4], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp1], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp6], %[tmp5], %[tmp2] \n\t"
+ "subu %[tmp5], %[tmp5], %[tmp2] \n\t"
+ "shra_r.w %[tmp1], %[tmp1], 15 \n\t"
+ "shra_r.w %[tmp6], %[tmp6], 15 \n\t"
+ "shra_r.w %[tmp4], %[tmp4], 15 \n\t"
+ "shra_r.w %[tmp5], %[tmp5], 15 \n\t"
+#else // #if defined(MIPS_DSP_R2_LE)
+ "mul %[tmp2], %[wr], %[tmp4] \n\t"
+ "mul %[tmp1], %[wr], %[tmp3] \n\t"
+ "mul %[tmp4], %[wi], %[tmp4] \n\t"
+ "mul %[tmp3], %[wi], %[tmp3] \n\t"
+ "sll %[tmp6], %[tmp6], 14 \n\t"
+ "sll %[tmp5], %[tmp5], 14 \n\t"
+ "addiu %[tmp6], %[tmp6], 16384 \n\t"
+ "addiu %[tmp5], %[tmp5], 16384 \n\t"
+ "addu %[tmp1], %[tmp1], %[tmp4] \n\t"
+ "subu %[tmp2], %[tmp2], %[tmp3] \n\t"
+ "addiu %[tmp1], %[tmp1], 1 \n\t"
+ "addiu %[tmp2], %[tmp2], 1 \n\t"
+ "sra %[tmp1], %[tmp1], 1 \n\t"
+ "sra %[tmp2], %[tmp2], 1 \n\t"
+ "subu %[tmp4], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp1], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp6], %[tmp5], %[tmp2] \n\t"
+ "subu %[tmp5], %[tmp5], %[tmp2] \n\t"
+ "sra %[tmp4], %[tmp4], 15 \n\t"
+ "sra %[tmp1], %[tmp1], 15 \n\t"
+ "sra %[tmp6], %[tmp6], 15 \n\t"
+ "sra %[tmp5], %[tmp5], 15 \n\t"
+#endif // #if defined(MIPS_DSP_R2_LE)
+ "sh %[tmp1], 0(%[ptr_i]) \n\t"
+ "sh %[tmp6], 2(%[ptr_i]) \n\t"
+ "sh %[tmp4], 0(%[ptr_j]) \n\t"
+ "blt %[i], %[n], 1b \n\t"
+ " sh %[tmp5], 2(%[ptr_j]) \n\t"
+ "blt %[m], %[l], 2b \n\t"
+ " addu %[i], $zero, %[m] \n\t"
+ "move %[l], %[istep] \n\t"
+ "blt %[l], %[n], 3b \n\t"
+ " addiu %[k], %[k], -1 \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
+ [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6),
+ [ptr_i] "=&r" (ptr_i), [i] "=&r" (i), [wi] "=&r" (wi), [wr] "=&r" (wr),
+ [m] "=&r" (m), [istep] "=&r" (istep), [l] "=&r" (l), [k] "=&r" (k),
+ [ptr_j] "=&r" (ptr_j), [tmp] "=&r" (tmp)
+ : [n] "r" (n), [frfi] "r" (frfi), [kSinTable1024] "r" (kSinTable1024)
+ : "hi", "lo", "memory"
+#if defined(MIPS_DSP_R2_LE)
+ , "$ac1hi", "$ac1lo"
+#endif // #if defined(MIPS_DSP_R2_LE)
+ );
+
+ return 0;
+}
+
+int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode) {
+ int i = 0, l = 0, k = 0;
+ int istep = 0, n = 0, m = 0;
+ int scale = 0, shift = 0;
+ int32_t wr = 0, wi = 0;
+ int32_t tmp1 = 0, tmp2 = 0, tmp3 = 0, tmp4 = 0;
+ int32_t tmp5 = 0, tmp6 = 0, tmp = 0, tempMax = 0, round2 = 0;
+ int16_t* ptr_j = NULL;
+ int16_t* ptr_i = NULL;
+
+ n = 1 << stages;
+ if (n > 1024) {
+ return -1;
+ }
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "addiu %[k], $zero, 10 \n\t"
+ "addiu %[l], $zero, 1 \n\t"
+ "move %[scale], $zero \n\t"
+ "3: \n\t"
+ "addiu %[shift], $zero, 14 \n\t"
+ "addiu %[round2], $zero, 8192 \n\t"
+ "move %[ptr_i], %[frfi] \n\t"
+ "move %[tempMax], $zero \n\t"
+ "addu %[i], %[n], %[n] \n\t"
+ "5: \n\t"
+ "lh %[tmp1], 0(%[ptr_i]) \n\t"
+ "lh %[tmp2], 2(%[ptr_i]) \n\t"
+ "lh %[tmp3], 4(%[ptr_i]) \n\t"
+ "lh %[tmp4], 6(%[ptr_i]) \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "absq_s.w %[tmp1], %[tmp1] \n\t"
+ "absq_s.w %[tmp2], %[tmp2] \n\t"
+ "absq_s.w %[tmp3], %[tmp3] \n\t"
+ "absq_s.w %[tmp4], %[tmp4] \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "slt %[tmp5], %[tmp1], $zero \n\t"
+ "subu %[tmp6], $zero, %[tmp1] \n\t"
+ "movn %[tmp1], %[tmp6], %[tmp5] \n\t"
+ "slt %[tmp5], %[tmp2], $zero \n\t"
+ "subu %[tmp6], $zero, %[tmp2] \n\t"
+ "movn %[tmp2], %[tmp6], %[tmp5] \n\t"
+ "slt %[tmp5], %[tmp3], $zero \n\t"
+ "subu %[tmp6], $zero, %[tmp3] \n\t"
+ "movn %[tmp3], %[tmp6], %[tmp5] \n\t"
+ "slt %[tmp5], %[tmp4], $zero \n\t"
+ "subu %[tmp6], $zero, %[tmp4] \n\t"
+ "movn %[tmp4], %[tmp6], %[tmp5] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "slt %[tmp5], %[tempMax], %[tmp1] \n\t"
+ "movn %[tempMax], %[tmp1], %[tmp5] \n\t"
+ "addiu %[i], %[i], -4 \n\t"
+ "slt %[tmp5], %[tempMax], %[tmp2] \n\t"
+ "movn %[tempMax], %[tmp2], %[tmp5] \n\t"
+ "slt %[tmp5], %[tempMax], %[tmp3] \n\t"
+ "movn %[tempMax], %[tmp3], %[tmp5] \n\t"
+ "slt %[tmp5], %[tempMax], %[tmp4] \n\t"
+ "movn %[tempMax], %[tmp4], %[tmp5] \n\t"
+ "bgtz %[i], 5b \n\t"
+ " addiu %[ptr_i], %[ptr_i], 8 \n\t"
+ "addiu %[tmp1], $zero, 13573 \n\t"
+ "addiu %[tmp2], $zero, 27146 \n\t"
+#if !defined(MIPS32_R2_LE)
+ "sll %[tempMax], %[tempMax], 16 \n\t"
+ "sra %[tempMax], %[tempMax], 16 \n\t"
+#else // #if !defined(MIPS32_R2_LE)
+ "seh %[tempMax] \n\t"
+#endif // #if !defined(MIPS32_R2_LE)
+ "slt %[tmp1], %[tmp1], %[tempMax] \n\t"
+ "slt %[tmp2], %[tmp2], %[tempMax] \n\t"
+ "addu %[tmp1], %[tmp1], %[tmp2] \n\t"
+ "addu %[shift], %[shift], %[tmp1] \n\t"
+ "addu %[scale], %[scale], %[tmp1] \n\t"
+ "sllv %[round2], %[round2], %[tmp1] \n\t"
+ "sll %[istep], %[l], 1 \n\t"
+ "move %[m], $zero \n\t"
+ "sll %[tmp], %[l], 2 \n\t"
+ "2: \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "sllv %[tmp3], %[m], %[k] \n\t"
+ "addiu %[tmp2], %[tmp3], 512 \n\t"
+ "addiu %[m], %[m], 1 \n\t"
+ "lhx %[wi], %[tmp3](%[kSinTable1024]) \n\t"
+ "lhx %[wr], %[tmp2](%[kSinTable1024]) \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "sllv %[tmp3], %[m], %[k] \n\t"
+ "addu %[ptr_j], %[tmp3], %[kSinTable1024] \n\t"
+ "addiu %[ptr_i], %[ptr_j], 512 \n\t"
+ "addiu %[m], %[m], 1 \n\t"
+ "lh %[wi], 0(%[ptr_j]) \n\t"
+ "lh %[wr], 0(%[ptr_i]) \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "1: \n\t"
+ "sll %[tmp1], %[i], 2 \n\t"
+ "addu %[ptr_i], %[frfi], %[tmp1] \n\t"
+ "addu %[ptr_j], %[ptr_i], %[tmp] \n\t"
+ "lh %[tmp3], 0(%[ptr_j]) \n\t"
+ "lh %[tmp4], 2(%[ptr_j]) \n\t"
+ "lh %[tmp6], 0(%[ptr_i]) \n\t"
+ "lh %[tmp5], 2(%[ptr_i]) \n\t"
+ "addu %[i], %[i], %[istep] \n\t"
+#if defined(MIPS_DSP_R2_LE)
+ "mult %[wr], %[tmp3] \n\t"
+ "msub %[wi], %[tmp4] \n\t"
+ "mult $ac1, %[wr], %[tmp4] \n\t"
+ "madd $ac1, %[wi], %[tmp3] \n\t"
+ "mflo %[tmp1] \n\t"
+ "mflo %[tmp2], $ac1 \n\t"
+ "sll %[tmp6], %[tmp6], 14 \n\t"
+ "sll %[tmp5], %[tmp5], 14 \n\t"
+ "shra_r.w %[tmp1], %[tmp1], 1 \n\t"
+ "shra_r.w %[tmp2], %[tmp2], 1 \n\t"
+ "addu %[tmp6], %[tmp6], %[round2] \n\t"
+ "addu %[tmp5], %[tmp5], %[round2] \n\t"
+ "subu %[tmp4], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp1], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp6], %[tmp5], %[tmp2] \n\t"
+ "subu %[tmp5], %[tmp5], %[tmp2] \n\t"
+ "srav %[tmp4], %[tmp4], %[shift] \n\t"
+ "srav %[tmp1], %[tmp1], %[shift] \n\t"
+ "srav %[tmp6], %[tmp6], %[shift] \n\t"
+ "srav %[tmp5], %[tmp5], %[shift] \n\t"
+#else // #if defined(MIPS_DSP_R2_LE)
+ "mul %[tmp1], %[wr], %[tmp3] \n\t"
+ "mul %[tmp2], %[wr], %[tmp4] \n\t"
+ "mul %[tmp4], %[wi], %[tmp4] \n\t"
+ "mul %[tmp3], %[wi], %[tmp3] \n\t"
+ "sll %[tmp6], %[tmp6], 14 \n\t"
+ "sll %[tmp5], %[tmp5], 14 \n\t"
+ "sub %[tmp1], %[tmp1], %[tmp4] \n\t"
+ "addu %[tmp2], %[tmp2], %[tmp3] \n\t"
+ "addiu %[tmp1], %[tmp1], 1 \n\t"
+ "addiu %[tmp2], %[tmp2], 1 \n\t"
+ "sra %[tmp2], %[tmp2], 1 \n\t"
+ "sra %[tmp1], %[tmp1], 1 \n\t"
+ "addu %[tmp6], %[tmp6], %[round2] \n\t"
+ "addu %[tmp5], %[tmp5], %[round2] \n\t"
+ "subu %[tmp4], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp1], %[tmp6], %[tmp1] \n\t"
+ "addu %[tmp6], %[tmp5], %[tmp2] \n\t"
+ "subu %[tmp5], %[tmp5], %[tmp2] \n\t"
+ "sra %[tmp4], %[tmp4], %[shift] \n\t"
+ "sra %[tmp1], %[tmp1], %[shift] \n\t"
+ "sra %[tmp6], %[tmp6], %[shift] \n\t"
+ "sra %[tmp5], %[tmp5], %[shift] \n\t"
+#endif // #if defined(MIPS_DSP_R2_LE)
+ "sh %[tmp1], 0(%[ptr_i]) \n\t"
+ "sh %[tmp6], 2(%[ptr_i]) \n\t"
+ "sh %[tmp4], 0(%[ptr_j]) \n\t"
+ "blt %[i], %[n], 1b \n\t"
+ " sh %[tmp5], 2(%[ptr_j]) \n\t"
+ "blt %[m], %[l], 2b \n\t"
+ " addu %[i], $zero, %[m] \n\t"
+ "move %[l], %[istep] \n\t"
+ "blt %[l], %[n], 3b \n\t"
+ " addiu %[k], %[k], -1 \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
+ [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6),
+ [ptr_i] "=&r" (ptr_i), [i] "=&r" (i), [m] "=&r" (m), [tmp] "=&r" (tmp),
+ [istep] "=&r" (istep), [wi] "=&r" (wi), [wr] "=&r" (wr), [l] "=&r" (l),
+ [k] "=&r" (k), [round2] "=&r" (round2), [ptr_j] "=&r" (ptr_j),
+ [shift] "=&r" (shift), [scale] "=&r" (scale), [tempMax] "=&r" (tempMax)
+ : [n] "r" (n), [frfi] "r" (frfi), [kSinTable1024] "r" (kSinTable1024)
+ : "hi", "lo", "memory"
+#if defined(MIPS_DSP_R2_LE)
+ , "$ac1hi", "$ac1lo"
+#endif // #if defined(MIPS_DSP_R2_LE)
+ );
+
+ return scale;
+
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/complex_fft_tables.h b/third_party/libwebrtc/common_audio/signal_processing/complex_fft_tables.h
new file mode 100644
index 0000000000..90fac072d2
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/complex_fft_tables.h
@@ -0,0 +1,132 @@
+/*
+ * 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_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/copy_set_operations.c b/third_party/libwebrtc/common_audio/signal_processing/copy_set_operations.c
new file mode 100644
index 0000000000..ae709d40f0
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/copy_set_operations.c
@@ -0,0 +1,82 @@
+/*
+ * 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 implementation of functions
+ * WebRtcSpl_MemSetW16()
+ * WebRtcSpl_MemSetW32()
+ * WebRtcSpl_MemCpyReversedOrder()
+ * WebRtcSpl_CopyFromEndW16()
+ * WebRtcSpl_ZerosArrayW16()
+ * WebRtcSpl_ZerosArrayW32()
+ *
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+
+void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, size_t length)
+{
+ size_t j;
+ int16_t *arrptr = ptr;
+
+ for (j = length; j > 0; j--)
+ {
+ *arrptr++ = set_value;
+ }
+}
+
+void WebRtcSpl_MemSetW32(int32_t *ptr, int32_t set_value, size_t length)
+{
+ size_t j;
+ int32_t *arrptr = ptr;
+
+ for (j = length; j > 0; j--)
+ {
+ *arrptr++ = set_value;
+ }
+}
+
+void WebRtcSpl_MemCpyReversedOrder(int16_t* dest,
+ int16_t* source,
+ size_t length)
+{
+ size_t j;
+ int16_t* destPtr = dest;
+ int16_t* sourcePtr = source;
+
+ for (j = 0; j < length; j++)
+ {
+ *destPtr-- = *sourcePtr++;
+ }
+}
+
+void WebRtcSpl_CopyFromEndW16(const int16_t *vector_in,
+ size_t length,
+ size_t samples,
+ int16_t *vector_out)
+{
+ // Copy the last <samples> of the input vector to vector_out
+ WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
+}
+
+void WebRtcSpl_ZerosArrayW16(int16_t *vector, size_t length)
+{
+ WebRtcSpl_MemSetW16(vector, 0, length);
+}
+
+void WebRtcSpl_ZerosArrayW32(int32_t *vector, size_t length)
+{
+ WebRtcSpl_MemSetW32(vector, 0, length);
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/cross_correlation.c b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation.c
new file mode 100644
index 0000000000..c6267c92c2
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation.c
@@ -0,0 +1,30 @@
+/*
+ * 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;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_mips.c b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_mips.c
new file mode 100644
index 0000000000..c395101900
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_mips.c
@@ -0,0 +1,104 @@
+/*
+ * 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 "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_CrossCorrelation_mips(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) {
+
+ int32_t t0 = 0, t1 = 0, t2 = 0, t3 = 0, sum = 0;
+ int16_t *pseq2 = NULL;
+ int16_t *pseq1 = NULL;
+ int16_t *pseq1_0 = (int16_t*)&seq1[0];
+ int16_t *pseq2_0 = (int16_t*)&seq2[0];
+ int k = 0;
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "sll %[step_seq2], %[step_seq2], 1 \n\t"
+ "andi %[t0], %[dim_seq], 1 \n\t"
+ "bgtz %[t0], 3f \n\t"
+ " nop \n\t"
+ "1: \n\t"
+ "move %[pseq1], %[pseq1_0] \n\t"
+ "move %[pseq2], %[pseq2_0] \n\t"
+ "sra %[k], %[dim_seq], 1 \n\t"
+ "addiu %[dim_cc], %[dim_cc], -1 \n\t"
+ "xor %[sum], %[sum], %[sum] \n\t"
+ "2: \n\t"
+ "lh %[t0], 0(%[pseq1]) \n\t"
+ "lh %[t1], 0(%[pseq2]) \n\t"
+ "lh %[t2], 2(%[pseq1]) \n\t"
+ "lh %[t3], 2(%[pseq2]) \n\t"
+ "mul %[t0], %[t0], %[t1] \n\t"
+ "addiu %[k], %[k], -1 \n\t"
+ "mul %[t2], %[t2], %[t3] \n\t"
+ "addiu %[pseq1], %[pseq1], 4 \n\t"
+ "addiu %[pseq2], %[pseq2], 4 \n\t"
+ "srav %[t0], %[t0], %[right_shifts] \n\t"
+ "addu %[sum], %[sum], %[t0] \n\t"
+ "srav %[t2], %[t2], %[right_shifts] \n\t"
+ "bgtz %[k], 2b \n\t"
+ " addu %[sum], %[sum], %[t2] \n\t"
+ "addu %[pseq2_0], %[pseq2_0], %[step_seq2] \n\t"
+ "sw %[sum], 0(%[cc]) \n\t"
+ "bgtz %[dim_cc], 1b \n\t"
+ " addiu %[cc], %[cc], 4 \n\t"
+ "b 6f \n\t"
+ " nop \n\t"
+ "3: \n\t"
+ "move %[pseq1], %[pseq1_0] \n\t"
+ "move %[pseq2], %[pseq2_0] \n\t"
+ "sra %[k], %[dim_seq], 1 \n\t"
+ "addiu %[dim_cc], %[dim_cc], -1 \n\t"
+ "beqz %[k], 5f \n\t"
+ " xor %[sum], %[sum], %[sum] \n\t"
+ "4: \n\t"
+ "lh %[t0], 0(%[pseq1]) \n\t"
+ "lh %[t1], 0(%[pseq2]) \n\t"
+ "lh %[t2], 2(%[pseq1]) \n\t"
+ "lh %[t3], 2(%[pseq2]) \n\t"
+ "mul %[t0], %[t0], %[t1] \n\t"
+ "addiu %[k], %[k], -1 \n\t"
+ "mul %[t2], %[t2], %[t3] \n\t"
+ "addiu %[pseq1], %[pseq1], 4 \n\t"
+ "addiu %[pseq2], %[pseq2], 4 \n\t"
+ "srav %[t0], %[t0], %[right_shifts] \n\t"
+ "addu %[sum], %[sum], %[t0] \n\t"
+ "srav %[t2], %[t2], %[right_shifts] \n\t"
+ "bgtz %[k], 4b \n\t"
+ " addu %[sum], %[sum], %[t2] \n\t"
+ "5: \n\t"
+ "lh %[t0], 0(%[pseq1]) \n\t"
+ "lh %[t1], 0(%[pseq2]) \n\t"
+ "mul %[t0], %[t0], %[t1] \n\t"
+ "srav %[t0], %[t0], %[right_shifts] \n\t"
+ "addu %[sum], %[sum], %[t0] \n\t"
+ "addu %[pseq2_0], %[pseq2_0], %[step_seq2] \n\t"
+ "sw %[sum], 0(%[cc]) \n\t"
+ "bgtz %[dim_cc], 3b \n\t"
+ " addiu %[cc], %[cc], 4 \n\t"
+ "6: \n\t"
+ ".set pop \n\t"
+ : [step_seq2] "+r" (step_seq2), [t0] "=&r" (t0), [t1] "=&r" (t1),
+ [t2] "=&r" (t2), [t3] "=&r" (t3), [pseq1] "=&r" (pseq1),
+ [pseq2] "=&r" (pseq2), [pseq1_0] "+r" (pseq1_0), [pseq2_0] "+r" (pseq2_0),
+ [k] "=&r" (k), [dim_cc] "+r" (dim_cross_correlation), [sum] "=&r" (sum),
+ [cc] "+r" (cross_correlation)
+ : [dim_seq] "r" (dim_seq), [right_shifts] "r" (right_shifts)
+ : "hi", "lo", "memory"
+ );
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_neon.c b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_neon.c
new file mode 100644
index 0000000000..f2afbdf9f5
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/cross_correlation_neon.c
@@ -0,0 +1,88 @@
+/*
+ * Copyright (c) 2014 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/system/arch.h"
+
+#include <arm_neon.h>
+
+static inline void DotProductWithScaleNeon(int32_t* cross_correlation,
+ const int16_t* vector1,
+ const int16_t* vector2,
+ size_t length,
+ int scaling) {
+ size_t i = 0;
+ size_t len1 = length >> 3;
+ size_t len2 = length & 7;
+ int64x2_t sum0 = vdupq_n_s64(0);
+ int64x2_t sum1 = vdupq_n_s64(0);
+
+ for (i = len1; i > 0; i -= 1) {
+ int16x8_t seq1_16x8 = vld1q_s16(vector1);
+ int16x8_t seq2_16x8 = vld1q_s16(vector2);
+#if defined(WEBRTC_ARCH_ARM64)
+ int32x4_t tmp0 = vmull_s16(vget_low_s16(seq1_16x8),
+ vget_low_s16(seq2_16x8));
+ int32x4_t tmp1 = vmull_high_s16(seq1_16x8, seq2_16x8);
+#else
+ int32x4_t tmp0 = vmull_s16(vget_low_s16(seq1_16x8),
+ vget_low_s16(seq2_16x8));
+ int32x4_t tmp1 = vmull_s16(vget_high_s16(seq1_16x8),
+ vget_high_s16(seq2_16x8));
+#endif
+ sum0 = vpadalq_s32(sum0, tmp0);
+ sum1 = vpadalq_s32(sum1, tmp1);
+ vector1 += 8;
+ vector2 += 8;
+ }
+
+ // Calculate the rest of the samples.
+ int64_t sum_res = 0;
+ for (i = len2; i > 0; i -= 1) {
+ sum_res += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
+ vector1++;
+ vector2++;
+ }
+
+ sum0 = vaddq_s64(sum0, sum1);
+#if defined(WEBRTC_ARCH_ARM64)
+ int64_t sum2 = vaddvq_s64(sum0);
+ *cross_correlation = (int32_t)((sum2 + sum_res) >> scaling);
+#else
+ int64x1_t shift = vdup_n_s64(-scaling);
+ int64x1_t sum2 = vadd_s64(vget_low_s64(sum0), vget_high_s64(sum0));
+ sum2 = vadd_s64(sum2, vdup_n_s64(sum_res));
+ sum2 = vshl_s64(sum2, shift);
+ vst1_lane_s32(cross_correlation, vreinterpret_s32_s64(sum2), 0);
+#endif
+}
+
+/* NEON version of WebRtcSpl_CrossCorrelation() for ARM32/64 platforms. */
+void WebRtcSpl_CrossCorrelationNeon(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;
+
+ for (i = 0; i < dim_cross_correlation; i++) {
+ const int16_t* seq1_ptr = seq1;
+ const int16_t* seq2_ptr = seq2 + (step_seq2 * i);
+
+ DotProductWithScaleNeon(cross_correlation,
+ seq1_ptr,
+ seq2_ptr,
+ dim_seq,
+ right_shifts);
+ cross_correlation++;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/division_operations.c b/third_party/libwebrtc/common_audio/signal_processing/division_operations.c
new file mode 100644
index 0000000000..4764ddfccd
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/division_operations.c
@@ -0,0 +1,140 @@
+/*
+ * 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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.cc b/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.cc
new file mode 100644
index 0000000000..00799dae02
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.cc
@@ -0,0 +1,34 @@
+/*
+ * 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/dot_product_with_scale.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
+ const int16_t* vector2,
+ size_t length,
+ int scaling) {
+ int64_t sum = 0;
+ size_t i = 0;
+
+ /* Unroll the loop to improve performance. */
+ for (i = 0; i + 3 < length; i += 4) {
+ sum += (vector1[i + 0] * vector2[i + 0]) >> scaling;
+ sum += (vector1[i + 1] * vector2[i + 1]) >> scaling;
+ sum += (vector1[i + 2] * vector2[i + 2]) >> scaling;
+ sum += (vector1[i + 3] * vector2[i + 3]) >> scaling;
+ }
+ for (; i < length; i++) {
+ sum += (vector1[i] * vector2[i]) >> scaling;
+ }
+
+ return rtc::saturated_cast<int32_t>(sum);
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.h b/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.h
new file mode 100644
index 0000000000..9f0d922aaf
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/dot_product_with_scale.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (c) 2017 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_DOT_PRODUCT_WITH_SCALE_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Calculates the dot product between two (int16_t) vectors.
+//
+// Input:
+// - vector1 : Vector 1
+// - vector2 : Vector 2
+// - vector_length : Number of samples used in the dot product
+// - scaling : The number of right bit shifts to apply on each term
+// during calculation to avoid overflow, i.e., the
+// output will be in Q(-`scaling`)
+//
+// Return value : The dot product in Q(-scaling)
+int32_t WebRtcSpl_DotProductWithScale(const int16_t* vector1,
+ const int16_t* vector2,
+ size_t length,
+ int scaling);
+
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/downsample_fast.c b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast.c
new file mode 100644
index 0000000000..80fdc58a49
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast.c
@@ -0,0 +1,65 @@
+/*
+ * 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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_mips.c b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_mips.c
new file mode 100644
index 0000000000..0f3f3a069f
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_mips.c
@@ -0,0 +1,169 @@
+/*
+ * 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 "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Version of WebRtcSpl_DownsampleFast() for MIPS platforms.
+int WebRtcSpl_DownsampleFast_mips(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) {
+ int i;
+ int j;
+ int k;
+ int32_t out_s32 = 0;
+ size_t endpos = delay + factor * (data_out_length - 1) + 1;
+
+ int32_t tmp1, tmp2, tmp3, tmp4, factor_2;
+ int16_t* p_coefficients;
+ int16_t* p_data_in;
+ int16_t* p_data_in_0 = (int16_t*)&data_in[delay];
+ int16_t* p_coefficients_0 = (int16_t*)&coefficients[0];
+#if !defined(MIPS_DSP_R1_LE)
+ int32_t max_16 = 0x7FFF;
+ int32_t min_16 = 0xFFFF8000;
+#endif // #if !defined(MIPS_DSP_R1_LE)
+
+ // 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;
+ }
+#if defined(MIPS_DSP_R2_LE)
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "subu %[i], %[endpos], %[delay] \n\t"
+ "sll %[factor_2], %[factor], 1 \n\t"
+ "1: \n\t"
+ "move %[p_data_in], %[p_data_in_0] \n\t"
+ "mult $zero, $zero \n\t"
+ "move %[p_coefs], %[p_coefs_0] \n\t"
+ "sra %[j], %[coef_length], 2 \n\t"
+ "beq %[j], $zero, 3f \n\t"
+ " andi %[k], %[coef_length], 3 \n\t"
+ "2: \n\t"
+ "lwl %[tmp1], 1(%[p_data_in]) \n\t"
+ "lwl %[tmp2], 3(%[p_coefs]) \n\t"
+ "lwl %[tmp3], -3(%[p_data_in]) \n\t"
+ "lwl %[tmp4], 7(%[p_coefs]) \n\t"
+ "lwr %[tmp1], -2(%[p_data_in]) \n\t"
+ "lwr %[tmp2], 0(%[p_coefs]) \n\t"
+ "lwr %[tmp3], -6(%[p_data_in]) \n\t"
+ "lwr %[tmp4], 4(%[p_coefs]) \n\t"
+ "packrl.ph %[tmp1], %[tmp1], %[tmp1] \n\t"
+ "packrl.ph %[tmp3], %[tmp3], %[tmp3] \n\t"
+ "dpa.w.ph $ac0, %[tmp1], %[tmp2] \n\t"
+ "dpa.w.ph $ac0, %[tmp3], %[tmp4] \n\t"
+ "addiu %[j], %[j], -1 \n\t"
+ "addiu %[p_data_in], %[p_data_in], -8 \n\t"
+ "bgtz %[j], 2b \n\t"
+ " addiu %[p_coefs], %[p_coefs], 8 \n\t"
+ "3: \n\t"
+ "beq %[k], $zero, 5f \n\t"
+ " nop \n\t"
+ "4: \n\t"
+ "lhu %[tmp1], 0(%[p_data_in]) \n\t"
+ "lhu %[tmp2], 0(%[p_coefs]) \n\t"
+ "addiu %[p_data_in], %[p_data_in], -2 \n\t"
+ "addiu %[k], %[k], -1 \n\t"
+ "dpa.w.ph $ac0, %[tmp1], %[tmp2] \n\t"
+ "bgtz %[k], 4b \n\t"
+ " addiu %[p_coefs], %[p_coefs], 2 \n\t"
+ "5: \n\t"
+ "extr_r.w %[out_s32], $ac0, 12 \n\t"
+ "addu %[p_data_in_0], %[p_data_in_0], %[factor_2] \n\t"
+ "subu %[i], %[i], %[factor] \n\t"
+ "shll_s.w %[out_s32], %[out_s32], 16 \n\t"
+ "sra %[out_s32], %[out_s32], 16 \n\t"
+ "sh %[out_s32], 0(%[data_out]) \n\t"
+ "bgtz %[i], 1b \n\t"
+ " addiu %[data_out], %[data_out], 2 \n\t"
+ ".set pop \n\t"
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
+ [tmp4] "=&r" (tmp4), [p_data_in] "=&r" (p_data_in),
+ [p_data_in_0] "+r" (p_data_in_0), [p_coefs] "=&r" (p_coefficients),
+ [j] "=&r" (j), [out_s32] "=&r" (out_s32), [factor_2] "=&r" (factor_2),
+ [i] "=&r" (i), [k] "=&r" (k)
+ : [coef_length] "r" (coefficients_length), [data_out] "r" (data_out),
+ [p_coefs_0] "r" (p_coefficients_0), [endpos] "r" (endpos),
+ [delay] "r" (delay), [factor] "r" (factor)
+ : "memory", "hi", "lo"
+ );
+#else // #if defined(MIPS_DSP_R2_LE)
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "sll %[factor_2], %[factor], 1 \n\t"
+ "subu %[i], %[endpos], %[delay] \n\t"
+ "1: \n\t"
+ "move %[p_data_in], %[p_data_in_0] \n\t"
+ "addiu %[out_s32], $zero, 2048 \n\t"
+ "move %[p_coefs], %[p_coefs_0] \n\t"
+ "sra %[j], %[coef_length], 1 \n\t"
+ "beq %[j], $zero, 3f \n\t"
+ " andi %[k], %[coef_length], 1 \n\t"
+ "2: \n\t"
+ "lh %[tmp1], 0(%[p_data_in]) \n\t"
+ "lh %[tmp2], 0(%[p_coefs]) \n\t"
+ "lh %[tmp3], -2(%[p_data_in]) \n\t"
+ "lh %[tmp4], 2(%[p_coefs]) \n\t"
+ "mul %[tmp1], %[tmp1], %[tmp2] \n\t"
+ "addiu %[p_coefs], %[p_coefs], 4 \n\t"
+ "mul %[tmp3], %[tmp3], %[tmp4] \n\t"
+ "addiu %[j], %[j], -1 \n\t"
+ "addiu %[p_data_in], %[p_data_in], -4 \n\t"
+ "addu %[tmp1], %[tmp1], %[tmp3] \n\t"
+ "bgtz %[j], 2b \n\t"
+ " addu %[out_s32], %[out_s32], %[tmp1] \n\t"
+ "3: \n\t"
+ "beq %[k], $zero, 4f \n\t"
+ " nop \n\t"
+ "lh %[tmp1], 0(%[p_data_in]) \n\t"
+ "lh %[tmp2], 0(%[p_coefs]) \n\t"
+ "mul %[tmp1], %[tmp1], %[tmp2] \n\t"
+ "addu %[out_s32], %[out_s32], %[tmp1] \n\t"
+ "4: \n\t"
+ "sra %[out_s32], %[out_s32], 12 \n\t"
+ "addu %[p_data_in_0], %[p_data_in_0], %[factor_2] \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "shll_s.w %[out_s32], %[out_s32], 16 \n\t"
+ "sra %[out_s32], %[out_s32], 16 \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "slt %[tmp1], %[max_16], %[out_s32] \n\t"
+ "movn %[out_s32], %[max_16], %[tmp1] \n\t"
+ "slt %[tmp1], %[out_s32], %[min_16] \n\t"
+ "movn %[out_s32], %[min_16], %[tmp1] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "subu %[i], %[i], %[factor] \n\t"
+ "sh %[out_s32], 0(%[data_out]) \n\t"
+ "bgtz %[i], 1b \n\t"
+ " addiu %[data_out], %[data_out], 2 \n\t"
+ ".set pop \n\t"
+ : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
+ [tmp4] "=&r" (tmp4), [p_data_in] "=&r" (p_data_in), [k] "=&r" (k),
+ [p_data_in_0] "+r" (p_data_in_0), [p_coefs] "=&r" (p_coefficients),
+ [j] "=&r" (j), [out_s32] "=&r" (out_s32), [factor_2] "=&r" (factor_2),
+ [i] "=&r" (i)
+ : [coef_length] "r" (coefficients_length), [data_out] "r" (data_out),
+ [p_coefs_0] "r" (p_coefficients_0), [endpos] "r" (endpos),
+#if !defined(MIPS_DSP_R1_LE)
+ [max_16] "r" (max_16), [min_16] "r" (min_16),
+#endif // #if !defined(MIPS_DSP_R1_LE)
+ [delay] "r" (delay), [factor] "r" (factor)
+ : "memory", "hi", "lo"
+ );
+#endif // #if defined(MIPS_DSP_R2_LE)
+ return 0;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_neon.c b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_neon.c
new file mode 100644
index 0000000000..36fc0c8aee
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/downsample_fast_neon.c
@@ -0,0 +1,217 @@
+/*
+ * Copyright (c) 2014 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 <arm_neon.h>
+
+// NEON intrinsics version of WebRtcSpl_DownsampleFast()
+// for ARM 32-bit/64-bit platforms.
+int WebRtcSpl_DownsampleFastNeon(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) {
+ size_t i = 0;
+ size_t j = 0;
+ int32_t out_s32 = 0;
+ size_t endpos = delay + factor * (data_out_length - 1) + 1;
+ size_t res = data_out_length & 0x7;
+ size_t endpos1 = endpos - factor * res;
+
+ // 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;
+ }
+
+ // First part, unroll the loop 8 times, with 3 subcases
+ // (factor == 2, 4, others).
+ switch (factor) {
+ case 2: {
+ for (i = delay; i < endpos1; i += 16) {
+ // Round value, 0.5 in Q12.
+ int32x4_t out32x4_0 = vdupq_n_s32(2048);
+ int32x4_t out32x4_1 = vdupq_n_s32(2048);
+
+#if defined(WEBRTC_ARCH_ARM64)
+ // Unroll the loop 2 times.
+ for (j = 0; j < coefficients_length - 1; j += 2) {
+ int32x2_t coeff32 = vld1_dup_s32((int32_t*)&coefficients[j]);
+ int16x4_t coeff16x4 = vreinterpret_s16_s32(coeff32);
+ int16x8x2_t in16x8x2 = vld2q_s16(&data_in[i - j - 1]);
+
+ // Mul and accumulate low 64-bit data.
+ int16x4_t in16x4_0 = vget_low_s16(in16x8x2.val[0]);
+ int16x4_t in16x4_1 = vget_low_s16(in16x8x2.val[1]);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 1);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_1, coeff16x4, 0);
+
+ // Mul and accumulate high 64-bit data.
+ // TODO: vget_high_s16 need extra cost on ARM64. This could be
+ // replaced by vmlal_high_lane_s16. But for the interface of
+ // vmlal_high_lane_s16, there is a bug in gcc 4.9.
+ // This issue need to be tracked in the future.
+ int16x4_t in16x4_2 = vget_high_s16(in16x8x2.val[0]);
+ int16x4_t in16x4_3 = vget_high_s16(in16x8x2.val[1]);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_2, coeff16x4, 1);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_3, coeff16x4, 0);
+ }
+
+ for (; j < coefficients_length; j++) {
+ int16x4_t coeff16x4 = vld1_dup_s16(&coefficients[j]);
+ int16x8x2_t in16x8x2 = vld2q_s16(&data_in[i - j]);
+
+ // Mul and accumulate low 64-bit data.
+ int16x4_t in16x4_0 = vget_low_s16(in16x8x2.val[0]);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 0);
+
+ // Mul and accumulate high 64-bit data.
+ // TODO: vget_high_s16 need extra cost on ARM64. This could be
+ // replaced by vmlal_high_lane_s16. But for the interface of
+ // vmlal_high_lane_s16, there is a bug in gcc 4.9.
+ // This issue need to be tracked in the future.
+ int16x4_t in16x4_1 = vget_high_s16(in16x8x2.val[0]);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_1, coeff16x4, 0);
+ }
+#else
+ // On ARMv7, the loop unrolling 2 times results in performance
+ // regression.
+ for (j = 0; j < coefficients_length; j++) {
+ int16x4_t coeff16x4 = vld1_dup_s16(&coefficients[j]);
+ int16x8x2_t in16x8x2 = vld2q_s16(&data_in[i - j]);
+
+ // Mul and accumulate.
+ int16x4_t in16x4_0 = vget_low_s16(in16x8x2.val[0]);
+ int16x4_t in16x4_1 = vget_high_s16(in16x8x2.val[0]);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 0);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_1, coeff16x4, 0);
+ }
+#endif
+
+ // Saturate and store the output.
+ int16x4_t out16x4_0 = vqshrn_n_s32(out32x4_0, 12);
+ int16x4_t out16x4_1 = vqshrn_n_s32(out32x4_1, 12);
+ vst1q_s16(data_out, vcombine_s16(out16x4_0, out16x4_1));
+ data_out += 8;
+ }
+ break;
+ }
+ case 4: {
+ for (i = delay; i < endpos1; i += 32) {
+ // Round value, 0.5 in Q12.
+ int32x4_t out32x4_0 = vdupq_n_s32(2048);
+ int32x4_t out32x4_1 = vdupq_n_s32(2048);
+
+ // Unroll the loop 4 times.
+ for (j = 0; j < coefficients_length - 3; j += 4) {
+ int16x4_t coeff16x4 = vld1_s16(&coefficients[j]);
+ int16x8x4_t in16x8x4 = vld4q_s16(&data_in[i - j - 3]);
+
+ // Mul and accumulate low 64-bit data.
+ int16x4_t in16x4_0 = vget_low_s16(in16x8x4.val[0]);
+ int16x4_t in16x4_2 = vget_low_s16(in16x8x4.val[1]);
+ int16x4_t in16x4_4 = vget_low_s16(in16x8x4.val[2]);
+ int16x4_t in16x4_6 = vget_low_s16(in16x8x4.val[3]);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 3);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_2, coeff16x4, 2);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_4, coeff16x4, 1);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_6, coeff16x4, 0);
+
+ // Mul and accumulate high 64-bit data.
+ // TODO: vget_high_s16 need extra cost on ARM64. This could be
+ // replaced by vmlal_high_lane_s16. But for the interface of
+ // vmlal_high_lane_s16, there is a bug in gcc 4.9.
+ // This issue need to be tracked in the future.
+ int16x4_t in16x4_1 = vget_high_s16(in16x8x4.val[0]);
+ int16x4_t in16x4_3 = vget_high_s16(in16x8x4.val[1]);
+ int16x4_t in16x4_5 = vget_high_s16(in16x8x4.val[2]);
+ int16x4_t in16x4_7 = vget_high_s16(in16x8x4.val[3]);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_1, coeff16x4, 3);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_3, coeff16x4, 2);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_5, coeff16x4, 1);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_7, coeff16x4, 0);
+ }
+
+ for (; j < coefficients_length; j++) {
+ int16x4_t coeff16x4 = vld1_dup_s16(&coefficients[j]);
+ int16x8x4_t in16x8x4 = vld4q_s16(&data_in[i - j]);
+
+ // Mul and accumulate low 64-bit data.
+ int16x4_t in16x4_0 = vget_low_s16(in16x8x4.val[0]);
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 0);
+
+ // Mul and accumulate high 64-bit data.
+ // TODO: vget_high_s16 need extra cost on ARM64. This could be
+ // replaced by vmlal_high_lane_s16. But for the interface of
+ // vmlal_high_lane_s16, there is a bug in gcc 4.9.
+ // This issue need to be tracked in the future.
+ int16x4_t in16x4_1 = vget_high_s16(in16x8x4.val[0]);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_1, coeff16x4, 0);
+ }
+
+ // Saturate and store the output.
+ int16x4_t out16x4_0 = vqshrn_n_s32(out32x4_0, 12);
+ int16x4_t out16x4_1 = vqshrn_n_s32(out32x4_1, 12);
+ vst1q_s16(data_out, vcombine_s16(out16x4_0, out16x4_1));
+ data_out += 8;
+ }
+ break;
+ }
+ default: {
+ for (i = delay; i < endpos1; i += factor * 8) {
+ // Round value, 0.5 in Q12.
+ int32x4_t out32x4_0 = vdupq_n_s32(2048);
+ int32x4_t out32x4_1 = vdupq_n_s32(2048);
+
+ for (j = 0; j < coefficients_length; j++) {
+ int16x4_t coeff16x4 = vld1_dup_s16(&coefficients[j]);
+ int16x4_t in16x4_0 = vld1_dup_s16(&data_in[i - j]);
+ in16x4_0 = vld1_lane_s16(&data_in[i + factor - j], in16x4_0, 1);
+ in16x4_0 = vld1_lane_s16(&data_in[i + factor * 2 - j], in16x4_0, 2);
+ in16x4_0 = vld1_lane_s16(&data_in[i + factor * 3 - j], in16x4_0, 3);
+ int16x4_t in16x4_1 = vld1_dup_s16(&data_in[i + factor * 4 - j]);
+ in16x4_1 = vld1_lane_s16(&data_in[i + factor * 5 - j], in16x4_1, 1);
+ in16x4_1 = vld1_lane_s16(&data_in[i + factor * 6 - j], in16x4_1, 2);
+ in16x4_1 = vld1_lane_s16(&data_in[i + factor * 7 - j], in16x4_1, 3);
+
+ // Mul and accumulate.
+ out32x4_0 = vmlal_lane_s16(out32x4_0, in16x4_0, coeff16x4, 0);
+ out32x4_1 = vmlal_lane_s16(out32x4_1, in16x4_1, coeff16x4, 0);
+ }
+
+ // Saturate and store the output.
+ int16x4_t out16x4_0 = vqshrn_n_s32(out32x4_0, 12);
+ int16x4_t out16x4_1 = vqshrn_n_s32(out32x4_1, 12);
+ vst1q_s16(data_out, vcombine_s16(out16x4_0, out16x4_1));
+ data_out += 8;
+ }
+ break;
+ }
+ }
+
+ // Second part, do the rest iterations (if any).
+ for (; i < endpos; i += factor) {
+ out_s32 = 2048; // Round value, 0.5 in Q12.
+
+ for (j = 0; j < coefficients_length; j++) {
+ out_s32 = WebRtc_MulAccumW16(coefficients[j], data_in[i - j], out_s32);
+ }
+
+ // Saturate and store the output.
+ out_s32 >>= 12;
+ *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
+ }
+
+ return 0;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/energy.c b/third_party/libwebrtc/common_audio/signal_processing/energy.c
new file mode 100644
index 0000000000..5cce6b8777
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/energy.c
@@ -0,0 +1,39 @@
+/*
+ * 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_Energy().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+int32_t WebRtcSpl_Energy(int16_t* vector,
+ size_t vector_length,
+ int* scale_factor)
+{
+ int32_t en = 0;
+ size_t i;
+ int scaling =
+ WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
+ size_t looptimes = vector_length;
+ int16_t *vectorptr = vector;
+
+ for (i = 0; i < looptimes; i++)
+ {
+ en += (*vectorptr * *vectorptr) >> scaling;
+ vectorptr++;
+ }
+ *scale_factor = scaling;
+
+ return en;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ar.c b/third_party/libwebrtc/common_audio/signal_processing/filter_ar.c
new file mode 100644
index 0000000000..b1f666d723
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ar.c
@@ -0,0 +1,95 @@
+/*
+ * 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_FilterAR().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+#include "rtc_base/checks.h"
+
+size_t WebRtcSpl_FilterAR(const int16_t* a,
+ size_t a_length,
+ const int16_t* x,
+ size_t x_length,
+ int16_t* state,
+ size_t state_length,
+ int16_t* state_low,
+ size_t state_low_length,
+ int16_t* filtered,
+ int16_t* filtered_low,
+ size_t filtered_low_length)
+{
+ int64_t o;
+ int32_t oLOW;
+ size_t i, j, stop;
+ const int16_t* x_ptr = &x[0];
+ int16_t* filteredFINAL_ptr = filtered;
+ int16_t* filteredFINAL_LOW_ptr = filtered_low;
+
+ for (i = 0; i < x_length; i++)
+ {
+ // Calculate filtered[i] and filtered_low[i]
+ const int16_t* a_ptr = &a[1];
+ // The index can become negative, but the arrays will never be indexed
+ // with it when negative. Nevertheless, the index cannot be a size_t
+ // because of this.
+ int filtered_ix = (int)i - 1;
+ int16_t* state_ptr = &state[state_length - 1];
+ int16_t* state_low_ptr = &state_low[state_length - 1];
+
+ o = (int32_t)(*x_ptr++) * (1 << 12);
+ oLOW = (int32_t)0;
+
+ stop = (i < a_length) ? i + 1 : a_length;
+ for (j = 1; j < stop; j++)
+ {
+ RTC_DCHECK_GE(filtered_ix, 0);
+ o -= *a_ptr * filtered[filtered_ix];
+ oLOW -= *a_ptr++ * filtered_low[filtered_ix];
+ --filtered_ix;
+ }
+ for (j = i + 1; j < a_length; j++)
+ {
+ o -= *a_ptr * *state_ptr--;
+ oLOW -= *a_ptr++ * *state_low_ptr--;
+ }
+
+ o += (oLOW >> 12);
+ *filteredFINAL_ptr = (int16_t)((o + (int32_t)2048) >> 12);
+ *filteredFINAL_LOW_ptr++ =
+ (int16_t)(o - ((int32_t)(*filteredFINAL_ptr++) * (1 << 12)));
+ }
+
+ // Save the filter state
+ if (x_length >= state_length)
+ {
+ WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
+ WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
+ } else
+ {
+ for (i = 0; i < state_length - x_length; i++)
+ {
+ state[i] = state[i + x_length];
+ state_low[i] = state_low[i + x_length];
+ }
+ for (i = 0; i < x_length; i++)
+ {
+ state[state_length - x_length + i] = filtered[i];
+ state_low[state_length - x_length + i] = filtered_low[i];
+ }
+ }
+
+ return x_length;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c
new file mode 100644
index 0000000000..9010f1ce82
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12.c
@@ -0,0 +1,47 @@
+/*
+ * 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 <stddef.h>
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// TODO(bjornv): Change the return type to report errors.
+
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+ int16_t* data_out,
+ const int16_t* __restrict coefficients,
+ size_t coefficients_length,
+ size_t data_length) {
+ size_t i = 0;
+ size_t j = 0;
+
+ RTC_DCHECK_GT(data_length, 0);
+ RTC_DCHECK_GT(coefficients_length, 1);
+
+ for (i = 0; i < data_length; i++) {
+ int64_t output = 0;
+ int64_t sum = 0;
+
+ for (j = coefficients_length - 1; j > 0; 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.
+ sum += coefficients[j] * data_out[(ptrdiff_t) i - (ptrdiff_t) j];
+ }
+
+ output = coefficients[0] * data_in[i];
+ output -= sum;
+
+ // Saturate and store the output.
+ output = WEBRTC_SPL_SAT(134215679, output, -134217728);
+ data_out[i] = (int16_t)((output + 2048) >> 12);
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
new file mode 100644
index 0000000000..60319d29ff
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
@@ -0,0 +1,218 @@
+@
+@ 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 function WebRtcSpl_FilterARFastQ12(), optimized for
+@ ARMv7 platform. The description header can be found in
+@ signal_processing_library.h
+@
+@ Output is bit-exact with the generic C code as in filter_ar_fast_q12.c, and
+@ the reference C code at end of this file.
+
+@ Assumptions:
+@ (1) data_length > 0
+@ (2) coefficients_length > 1
+
+@ Register usage:
+@
+@ r0: &data_in[i]
+@ r1: &data_out[i], for result ouput
+@ r2: &coefficients[0]
+@ r3: coefficients_length
+@ r4: Iteration counter for the outer loop.
+@ r5: data_out[j] as multiplication inputs
+@ r6: Calculated value for output data_out[]; interation counter for inner loop
+@ r7: Partial sum of a filtering multiplication results
+@ r8: Partial sum of a filtering multiplication results
+@ r9: &data_out[], for filtering input; data_in[i]
+@ r10: coefficients[j]
+@ r11: Scratch
+@ r12: &coefficients[j]
+
+#include "rtc_base/system/asm_defines.h"
+
+GLOBAL_FUNCTION WebRtcSpl_FilterARFastQ12
+.align 2
+DEFINE_FUNCTION WebRtcSpl_FilterARFastQ12
+ push {r4-r11}
+
+ ldrsh r12, [sp, #32] @ data_length
+ subs r4, r12, #1
+ beq ODD_LENGTH @ jump if data_length == 1
+
+LOOP_LENGTH:
+ add r12, r2, r3, lsl #1
+ sub r12, #4 @ &coefficients[coefficients_length - 2]
+ sub r9, r1, r3, lsl #1
+ add r9, #2 @ &data_out[i - coefficients_length + 1]
+ ldr r5, [r9], #4 @ data_out[i - coefficients_length + {1,2}]
+
+ mov r7, #0 @ sum1
+ mov r8, #0 @ sum2
+ subs r6, r3, #3 @ Iteration counter for inner loop.
+ beq ODD_A_LENGTH @ branch if coefficients_length == 3
+ blt POST_LOOP_A_LENGTH @ branch if coefficients_length == 2
+
+LOOP_A_LENGTH:
+ ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
+ subs r6, #2
+ smlatt r8, r10, r5, r8 @ sum2 += coefficients[j] * data_out[i - j + 1];
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
+ smlabt r7, r10, r5, r7 @ coefficients[j - 1] * data_out[i - j + 1];
+ ldr r5, [r9], #4 @ data_out[i - j + 2], data_out[i - j + 3]
+ smlabb r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 2];
+ bgt LOOP_A_LENGTH
+ blt POST_LOOP_A_LENGTH
+
+ODD_A_LENGTH:
+ ldrsh r10, [r12, #2] @ Filter coefficients coefficients[2]
+ sub r12, #2 @ &coefficients[0]
+ smlabb r7, r10, r5, r7 @ sum1 += coefficients[2] * data_out[i - 2];
+ smlabt r8, r10, r5, r8 @ sum2 += coefficients[2] * data_out[i - 1];
+ ldr r5, [r9, #-2] @ data_out[i - 1], data_out[i]
+
+POST_LOOP_A_LENGTH:
+ ldr r10, [r12] @ coefficients[0], coefficients[1]
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
+
+ ldr r9, [r0], #4 @ data_in[i], data_in[i + 1]
+ smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
+ sub r6, r7 @ output1 -= sum1;
+
+ sbfx r11, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r11
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1], #2 @ Store data_out[i]
+
+ smlatb r8, r10, r6, r8 @ sum2 += coefficients[1] * data_out[i];
+ smulbt r6, r10, r9 @ output2 = coefficients[0] * data_in[i + 1];
+ sub r6, r8 @ output1 -= sum1;
+
+ sbfx r11, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r11
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1], #2 @ Store data_out[i + 1]
+
+ subs r4, #2
+ bgt LOOP_LENGTH
+ blt END @ For even data_length, it's done. Jump to END.
+
+@ Process i = data_length -1, for the case of an odd length.
+ODD_LENGTH:
+ add r12, r2, r3, lsl #1
+ sub r12, #4 @ &coefficients[coefficients_length - 2]
+ sub r9, r1, r3, lsl #1
+ add r9, #2 @ &data_out[i - coefficients_length + 1]
+ mov r7, #0 @ sum1
+ mov r8, #0 @ sum1
+ subs r6, r3, #2 @ inner loop counter
+ beq EVEN_A_LENGTH @ branch if coefficients_length == 2
+
+LOOP2_A_LENGTH:
+ ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
+ ldr r5, [r9], #4 @ data_out[i - j], data_out[i - j + 1]
+ subs r6, #2
+ smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
+ smlabt r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 1];
+ bgt LOOP2_A_LENGTH
+ addlt r12, #2
+ blt POST_LOOP2_A_LENGTH
+
+EVEN_A_LENGTH:
+ ldrsh r10, [r12, #2] @ Filter coefficients coefficients[1]
+ ldrsh r5, [r9] @ data_out[i - 1]
+ smlabb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
+
+POST_LOOP2_A_LENGTH:
+ ldrsh r10, [r12] @ Filter coefficients coefficients[0]
+ ldrsh r9, [r0] @ data_in[i]
+ smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
+ sub r6, r7 @ output1 -= sum1;
+ sub r6, r8 @ output1 -= sum1;
+ sbfx r8, r6, #12, #16
+ ssat r7, #16, r6, asr #12
+ cmp r7, r8
+ addeq r6, r6, #2048
+ ssat r6, #16, r6, asr #12
+ strh r6, [r1] @ Store the data_out[i]
+
+END:
+ pop {r4-r11}
+ bx lr
+
+@Reference C code:
+@
+@void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
+@ int16_t* data_out,
+@ int16_t* __restrict coefficients,
+@ size_t coefficients_length,
+@ size_t data_length) {
+@ size_t i = 0;
+@ size_t j = 0;
+@
+@ assert(data_length > 0);
+@ assert(coefficients_length > 1);
+@
+@ for (i = 0; i < data_length - 1; i += 2) {
+@ int32_t output1 = 0;
+@ int32_t sum1 = 0;
+@ int32_t output2 = 0;
+@ int32_t sum2 = 0;
+@
+@ for (j = coefficients_length - 1; j > 2; j -= 2) {
+@ sum1 += coefficients[j] * data_out[i - j];
+@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
+@ sum2 += coefficients[j] * data_out[i - j + 1];
+@ sum2 += coefficients[j - 1] * data_out[i - j + 2];
+@ }
+@
+@ if (j == 2) {
+@ sum1 += coefficients[2] * data_out[i - 2];
+@ sum2 += coefficients[2] * data_out[i - 1];
+@ }
+@
+@ sum1 += coefficients[1] * data_out[i - 1];
+@ output1 = coefficients[0] * data_in[i];
+@ output1 -= sum1;
+@ // Saturate and store the output.
+@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@
+@ sum2 += coefficients[1] * data_out[i];
+@ output2 = coefficients[0] * data_in[i + 1];
+@ output2 -= sum2;
+@ // Saturate and store the output.
+@ output2 = WEBRTC_SPL_SAT(134215679, output2, -134217728);
+@ data_out[i + 1] = (int16_t)((output2 + 2048) >> 12);
+@ }
+@
+@ if (i == data_length - 1) {
+@ int32_t output1 = 0;
+@ int32_t sum1 = 0;
+@
+@ for (j = coefficients_length - 1; j > 1; j -= 2) {
+@ sum1 += coefficients[j] * data_out[i - j];
+@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
+@ }
+@
+@ if (j == 1) {
+@ sum1 += coefficients[1] * data_out[i - 1];
+@ }
+@
+@ output1 = coefficients[0] * data_in[i];
+@ output1 -= sum1;
+@ // Saturate and store the output.
+@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
+@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
+@ }
+@}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
new file mode 100644
index 0000000000..b9ad30f006
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ar_fast_q12_mips.c
@@ -0,0 +1,140 @@
+/*
+ * 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 "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+ int16_t* data_out,
+ const int16_t* __restrict coefficients,
+ size_t coefficients_length,
+ size_t data_length) {
+ int r0, r1, r2, r3;
+ int coef0, offset;
+ int i, j, k;
+ int coefptr, outptr, tmpout, inptr;
+#if !defined(MIPS_DSP_R1_LE)
+ int max16 = 0x7FFF;
+ int min16 = 0xFFFF8000;
+#endif // #if !defined(MIPS_DSP_R1_LE)
+
+ RTC_DCHECK_GT(data_length, 0);
+ RTC_DCHECK_GT(coefficients_length, 1);
+
+ __asm __volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "addiu %[i], %[data_length], 0 \n\t"
+ "lh %[coef0], 0(%[coefficients]) \n\t"
+ "addiu %[j], %[coefficients_length], -1 \n\t"
+ "andi %[k], %[j], 1 \n\t"
+ "sll %[offset], %[j], 1 \n\t"
+ "subu %[outptr], %[data_out], %[offset] \n\t"
+ "addiu %[inptr], %[data_in], 0 \n\t"
+ "bgtz %[k], 3f \n\t"
+ " addu %[coefptr], %[coefficients], %[offset] \n\t"
+ "1: \n\t"
+ "lh %[r0], 0(%[inptr]) \n\t"
+ "addiu %[i], %[i], -1 \n\t"
+ "addiu %[tmpout], %[outptr], 0 \n\t"
+ "mult %[r0], %[coef0] \n\t"
+ "2: \n\t"
+ "lh %[r0], 0(%[tmpout]) \n\t"
+ "lh %[r1], 0(%[coefptr]) \n\t"
+ "lh %[r2], 2(%[tmpout]) \n\t"
+ "lh %[r3], -2(%[coefptr]) \n\t"
+ "addiu %[tmpout], %[tmpout], 4 \n\t"
+ "msub %[r0], %[r1] \n\t"
+ "msub %[r2], %[r3] \n\t"
+ "addiu %[j], %[j], -2 \n\t"
+ "bgtz %[j], 2b \n\t"
+ " addiu %[coefptr], %[coefptr], -4 \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "extr_r.w %[r0], $ac0, 12 \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "mflo %[r0] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "addu %[coefptr], %[coefficients], %[offset] \n\t"
+ "addiu %[inptr], %[inptr], 2 \n\t"
+ "addiu %[j], %[coefficients_length], -1 \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "shll_s.w %[r0], %[r0], 16 \n\t"
+ "sra %[r0], %[r0], 16 \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "addiu %[r0], %[r0], 2048 \n\t"
+ "sra %[r0], %[r0], 12 \n\t"
+ "slt %[r1], %[max16], %[r0] \n\t"
+ "movn %[r0], %[max16], %[r1] \n\t"
+ "slt %[r1], %[r0], %[min16] \n\t"
+ "movn %[r0], %[min16], %[r1] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "sh %[r0], 0(%[tmpout]) \n\t"
+ "bgtz %[i], 1b \n\t"
+ " addiu %[outptr], %[outptr], 2 \n\t"
+ "b 5f \n\t"
+ " nop \n\t"
+ "3: \n\t"
+ "lh %[r0], 0(%[inptr]) \n\t"
+ "addiu %[i], %[i], -1 \n\t"
+ "addiu %[tmpout], %[outptr], 0 \n\t"
+ "mult %[r0], %[coef0] \n\t"
+ "4: \n\t"
+ "lh %[r0], 0(%[tmpout]) \n\t"
+ "lh %[r1], 0(%[coefptr]) \n\t"
+ "lh %[r2], 2(%[tmpout]) \n\t"
+ "lh %[r3], -2(%[coefptr]) \n\t"
+ "addiu %[tmpout], %[tmpout], 4 \n\t"
+ "msub %[r0], %[r1] \n\t"
+ "msub %[r2], %[r3] \n\t"
+ "addiu %[j], %[j], -2 \n\t"
+ "bgtz %[j], 4b \n\t"
+ " addiu %[coefptr], %[coefptr], -4 \n\t"
+ "lh %[r0], 0(%[tmpout]) \n\t"
+ "lh %[r1], 0(%[coefptr]) \n\t"
+ "msub %[r0], %[r1] \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "extr_r.w %[r0], $ac0, 12 \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "mflo %[r0] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "addu %[coefptr], %[coefficients], %[offset] \n\t"
+ "addiu %[inptr], %[inptr], 2 \n\t"
+ "addiu %[j], %[coefficients_length], -1 \n\t"
+#if defined(MIPS_DSP_R1_LE)
+ "shll_s.w %[r0], %[r0], 16 \n\t"
+ "sra %[r0], %[r0], 16 \n\t"
+#else // #if defined(MIPS_DSP_R1_LE)
+ "addiu %[r0], %[r0], 2048 \n\t"
+ "sra %[r0], %[r0], 12 \n\t"
+ "slt %[r1], %[max16], %[r0] \n\t"
+ "movn %[r0], %[max16], %[r1] \n\t"
+ "slt %[r1], %[r0], %[min16] \n\t"
+ "movn %[r0], %[min16], %[r1] \n\t"
+#endif // #if defined(MIPS_DSP_R1_LE)
+ "sh %[r0], 2(%[tmpout]) \n\t"
+ "bgtz %[i], 3b \n\t"
+ " addiu %[outptr], %[outptr], 2 \n\t"
+ "5: \n\t"
+ ".set pop \n\t"
+ : [i] "=&r" (i), [j] "=&r" (j), [k] "=&r" (k), [r0] "=&r" (r0),
+ [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3),
+ [coef0] "=&r" (coef0), [offset] "=&r" (offset),
+ [outptr] "=&r" (outptr), [inptr] "=&r" (inptr),
+ [coefptr] "=&r" (coefptr), [tmpout] "=&r" (tmpout)
+ : [coefficients] "r" (coefficients), [data_length] "r" (data_length),
+ [coefficients_length] "r" (coefficients_length),
+#if !defined(MIPS_DSP_R1_LE)
+ [max16] "r" (max16), [min16] "r" (min16),
+#endif
+ [data_out] "r" (data_out), [data_in] "r" (data_in)
+ : "hi", "lo", "memory"
+ );
+}
+
diff --git a/third_party/libwebrtc/common_audio/signal_processing/filter_ma_fast_q12.c b/third_party/libwebrtc/common_audio/signal_processing/filter_ma_fast_q12.c
new file mode 100644
index 0000000000..329d47e14f
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/filter_ma_fast_q12.c
@@ -0,0 +1,55 @@
+/*
+ * 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_FilterMAFastQ12().
+ * 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"
+
+void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
+ int16_t* out_ptr,
+ const int16_t* B,
+ size_t B_length,
+ size_t length)
+{
+ size_t i, j;
+
+ rtc_MsanCheckInitialized(B, sizeof(B[0]), B_length);
+ rtc_MsanCheckInitialized(in_ptr - B_length + 1, sizeof(in_ptr[0]),
+ B_length + length - 1);
+
+ for (i = 0; i < length; i++)
+ {
+ int32_t o = 0;
+
+ for (j = 0; j < B_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.
+ o += B[j] * in_ptr[(ptrdiff_t) i - (ptrdiff_t) j];
+ }
+
+ // If output is higher than 32768, saturate it. Same with negative side
+ // 2^27 = 134217728, which corresponds to 32768 in Q12
+
+ // Saturate the output
+ o = WEBRTC_SPL_SAT((int32_t)134215679, o, (int32_t)-134217728);
+
+ *out_ptr++ = (int16_t)((o + (int32_t)2048) >> 12);
+ }
+ return;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/get_hanning_window.c b/third_party/libwebrtc/common_audio/signal_processing/get_hanning_window.c
new file mode 100644
index 0000000000..8f29da8d9b
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/get_hanning_window.c
@@ -0,0 +1,77 @@
+/*
+ * 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_GetHanningWindow().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Hanning table with 256 entries
+static const int16_t kHanningTable[] = {
+ 1, 2, 6, 10, 15, 22, 30, 39,
+ 50, 62, 75, 89, 104, 121, 138, 157,
+ 178, 199, 222, 246, 271, 297, 324, 353,
+ 383, 413, 446, 479, 513, 549, 586, 624,
+ 663, 703, 744, 787, 830, 875, 920, 967,
+ 1015, 1064, 1114, 1165, 1218, 1271, 1325, 1381,
+ 1437, 1494, 1553, 1612, 1673, 1734, 1796, 1859,
+ 1924, 1989, 2055, 2122, 2190, 2259, 2329, 2399,
+ 2471, 2543, 2617, 2691, 2765, 2841, 2918, 2995,
+ 3073, 3152, 3232, 3312, 3393, 3475, 3558, 3641,
+ 3725, 3809, 3895, 3980, 4067, 4154, 4242, 4330,
+ 4419, 4509, 4599, 4689, 4781, 4872, 4964, 5057,
+ 5150, 5244, 5338, 5432, 5527, 5622, 5718, 5814,
+ 5910, 6007, 6104, 6202, 6299, 6397, 6495, 6594,
+ 6693, 6791, 6891, 6990, 7090, 7189, 7289, 7389,
+ 7489, 7589, 7690, 7790, 7890, 7991, 8091, 8192,
+ 8293, 8393, 8494, 8594, 8694, 8795, 8895, 8995,
+ 9095, 9195, 9294, 9394, 9493, 9593, 9691, 9790,
+ 9889, 9987, 10085, 10182, 10280, 10377, 10474, 10570,
+10666, 10762, 10857, 10952, 11046, 11140, 11234, 11327,
+11420, 11512, 11603, 11695, 11785, 11875, 11965, 12054,
+12142, 12230, 12317, 12404, 12489, 12575, 12659, 12743,
+12826, 12909, 12991, 13072, 13152, 13232, 13311, 13389,
+13466, 13543, 13619, 13693, 13767, 13841, 13913, 13985,
+14055, 14125, 14194, 14262, 14329, 14395, 14460, 14525,
+14588, 14650, 14711, 14772, 14831, 14890, 14947, 15003,
+15059, 15113, 15166, 15219, 15270, 15320, 15369, 15417,
+15464, 15509, 15554, 15597, 15640, 15681, 15721, 15760,
+15798, 15835, 15871, 15905, 15938, 15971, 16001, 16031,
+16060, 16087, 16113, 16138, 16162, 16185, 16206, 16227,
+16246, 16263, 16280, 16295, 16309, 16322, 16334, 16345,
+16354, 16362, 16369, 16374, 16378, 16382, 16383, 16384
+};
+
+void WebRtcSpl_GetHanningWindow(int16_t *v, size_t size)
+{
+ size_t jj;
+ int16_t *vptr1;
+
+ int32_t index;
+ int32_t factor = ((int32_t)0x40000000);
+
+ factor = WebRtcSpl_DivW32W16(factor, (int16_t)size);
+ if (size < 513)
+ index = (int32_t)-0x200000;
+ else
+ index = (int32_t)-0x100000;
+ vptr1 = v;
+
+ for (jj = 0; jj < size; jj++)
+ {
+ index += factor;
+ (*vptr1++) = kHanningTable[index >> 22];
+ }
+
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/get_scaling_square.c b/third_party/libwebrtc/common_audio/signal_processing/get_scaling_square.c
new file mode 100644
index 0000000000..4eb126941e
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/get_scaling_square.c
@@ -0,0 +1,46 @@
+/*
+ * 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_GetScalingSquare().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
+ size_t in_vector_length,
+ size_t times)
+{
+ int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
+ size_t i;
+ int16_t smax = -1;
+ int16_t sabs;
+ int16_t *sptr = in_vector;
+ int16_t t;
+ size_t looptimes = in_vector_length;
+
+ for (i = looptimes; i > 0; i--)
+ {
+ sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
+ smax = (sabs > smax ? sabs : smax);
+ }
+ t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
+
+ if (smax == 0)
+ {
+ return 0; // Since norm(0) returns 0
+ } else
+ {
+ return (t > nbits) ? 0 : nbits - t;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/ilbc_specific_functions.c b/third_party/libwebrtc/common_audio/signal_processing/ilbc_specific_functions.c
new file mode 100644
index 0000000000..cbdd3dcbcd
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/ilbc_specific_functions.c
@@ -0,0 +1,90 @@
+/*
+ * 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 iLBC specific functions
+ * WebRtcSpl_ReverseOrderMultArrayElements()
+ * WebRtcSpl_ElementwiseVectorMult()
+ * WebRtcSpl_AddVectorsAndShift()
+ * WebRtcSpl_AddAffineVectorToVector()
+ * WebRtcSpl_AffineTransformVector()
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_ReverseOrderMultArrayElements(int16_t *out, const int16_t *in,
+ const int16_t *win,
+ size_t vector_length,
+ int16_t right_shifts)
+{
+ size_t i;
+ int16_t *outptr = out;
+ const int16_t *inptr = in;
+ const int16_t *winptr = win;
+ for (i = 0; i < vector_length; i++)
+ {
+ *outptr++ = (int16_t)((*inptr++ * *winptr--) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_ElementwiseVectorMult(int16_t *out, const int16_t *in,
+ const int16_t *win, size_t vector_length,
+ int16_t right_shifts)
+{
+ size_t i;
+ int16_t *outptr = out;
+ const int16_t *inptr = in;
+ const int16_t *winptr = win;
+ for (i = 0; i < vector_length; i++)
+ {
+ *outptr++ = (int16_t)((*inptr++ * *winptr++) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_AddVectorsAndShift(int16_t *out, const int16_t *in1,
+ const int16_t *in2, size_t vector_length,
+ int16_t right_shifts)
+{
+ size_t i;
+ int16_t *outptr = out;
+ const int16_t *in1ptr = in1;
+ const int16_t *in2ptr = in2;
+ for (i = vector_length; i > 0; i--)
+ {
+ (*outptr++) = (int16_t)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_AddAffineVectorToVector(int16_t *out, const int16_t *in,
+ int16_t gain, int32_t add_constant,
+ int16_t right_shifts,
+ size_t vector_length)
+{
+ size_t i;
+
+ for (i = 0; i < vector_length; i++)
+ {
+ out[i] += (int16_t)((in[i] * gain + add_constant) >> right_shifts);
+ }
+}
+
+void WebRtcSpl_AffineTransformVector(int16_t *out, const int16_t *in,
+ int16_t gain, int32_t add_constant,
+ int16_t right_shifts, size_t vector_length)
+{
+ size_t i;
+
+ for (i = 0; i < vector_length; i++)
+ {
+ out[i] = (int16_t)((in[i] * gain + add_constant) >> right_shifts);
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/include/real_fft.h b/third_party/libwebrtc/common_audio/signal_processing/include/real_fft.h
new file mode 100644
index 0000000000..a0da5096c1
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/include/real_fft.h
@@ -0,0 +1,96 @@
+/*
+ * 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_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/include/signal_processing_library.h b/third_party/libwebrtc/common_audio/signal_processing/include/signal_processing_library.h
new file mode 100644
index 0000000000..48c9b309b4
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -0,0 +1,1635 @@
+/*
+ * 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 header file includes all of the fix point signal processing library
+ * (SPL) function descriptions and declarations. For specific function calls,
+ * see bottom of file.
+ */
+
+#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
+
+#include <string.h>
+
+#include "common_audio/signal_processing/dot_product_with_scale.h"
+
+// Macros specific for the fixed point implementation
+#define WEBRTC_SPL_WORD16_MAX 32767
+#define WEBRTC_SPL_WORD16_MIN -32768
+#define WEBRTC_SPL_WORD32_MAX (int32_t)0x7fffffff
+#define WEBRTC_SPL_WORD32_MIN (int32_t)0x80000000
+#define WEBRTC_SPL_MAX_LPC_ORDER 14
+#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
+#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
+// TODO(kma/bjorn): For the next two macros, investigate how to correct the code
+// for inputs of a = WEBRTC_SPL_WORD16_MIN or WEBRTC_SPL_WORD32_MIN.
+#define WEBRTC_SPL_ABS_W16(a) (((int16_t)a >= 0) ? ((int16_t)a) : -((int16_t)a))
+#define WEBRTC_SPL_ABS_W32(a) (((int32_t)a >= 0) ? ((int32_t)a) : -((int32_t)a))
+
+#define WEBRTC_SPL_MUL(a, b) ((int32_t)((int32_t)(a) * (int32_t)(b)))
+#define WEBRTC_SPL_UMUL(a, b) ((uint32_t)((uint32_t)(a) * (uint32_t)(b)))
+#define WEBRTC_SPL_UMUL_32_16(a, b) ((uint32_t)((uint32_t)(a) * (uint16_t)(b)))
+#define WEBRTC_SPL_MUL_16_U16(a, b) ((int32_t)(int16_t)(a) * (uint16_t)(b))
+
+// clang-format off
+// clang-format would choose some identation
+// leading to presubmit error (cpplint.py)
+#ifndef WEBRTC_ARCH_ARM_V7
+// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
+#ifndef MIPS32_LE
+// For MIPS platforms, these are inline functions in spl_inl_mips.h
+#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t)(((int16_t)(a)) * ((int16_t)(b))))
+#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
+ (WEBRTC_SPL_MUL_16_16(a, b >> 16) + \
+ ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
+#endif
+#endif
+
+#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
+ (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 5) + \
+ (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
+#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
+ (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 2) + \
+ (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
+#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
+ ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 1)) + \
+ (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x2000) >> 14))
+// clang-format on
+
+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
+
+#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
+ ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t)(((int32_t)1) << ((c)-1)))) >> (c))
+
+// C + the 32 most significant bits of A * B
+#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
+ (C + (B >> 16) * A + (((uint32_t)(B & 0x0000FFFF) * A) >> 16))
+
+#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
+
+// Shifting with negative numbers allowed
+// Positive means left shift
+#define WEBRTC_SPL_SHIFT_W32(x, c) ((c) >= 0 ? (x) * (1 << (c)) : (x) >> -(c))
+
+// Shifting with negative numbers not allowed
+// We cannot do casting here due to signed/unsigned problem
+#define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
+
+#define WEBRTC_SPL_RSHIFT_U32(x, c) ((uint32_t)(x) >> (c))
+
+#define WEBRTC_SPL_RAND(a) ((int16_t)((((int16_t)a * 18816) >> 7) & 0x00007fff))
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
+ memcpy(v1, v2, (length) * sizeof(int16_t))
+
+// inline functions:
+#include "common_audio/signal_processing/include/spl_inl.h"
+
+// third party math functions
+#include "common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h"
+
+int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
+ size_t in_vector_length,
+ size_t times);
+
+// Copy and set operations. Implementation in copy_set_operations.c.
+// Descriptions at bottom of file.
+void WebRtcSpl_MemSetW16(int16_t* vector,
+ int16_t set_value,
+ size_t vector_length);
+void WebRtcSpl_MemSetW32(int32_t* vector,
+ int32_t set_value,
+ size_t vector_length);
+void WebRtcSpl_MemCpyReversedOrder(int16_t* out_vector,
+ int16_t* in_vector,
+ size_t vector_length);
+void WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
+ size_t in_vector_length,
+ size_t samples,
+ int16_t* out_vector);
+void WebRtcSpl_ZerosArrayW16(int16_t* vector, size_t vector_length);
+void WebRtcSpl_ZerosArrayW32(int32_t* vector, size_t vector_length);
+// End: Copy and set operations.
+
+// Minimum and maximum operation functions and their pointers.
+// Implementation in min_max_operations.c.
+
+// Returns the largest absolute value in a signed 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Maximum absolute value in vector.
+typedef int16_t (*MaxAbsValueW16)(const int16_t* vector, size_t length);
+extern const MaxAbsValueW16 WebRtcSpl_MaxAbsValueW16;
+int16_t WebRtcSpl_MaxAbsValueW16C(const int16_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length);
+#endif
+
+// Returns the largest absolute value in a signed 32-bit vector.
+//
+// Input:
+// - vector : 32-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Maximum absolute value in vector.
+typedef int32_t (*MaxAbsValueW32)(const int32_t* vector, size_t length);
+extern const MaxAbsValueW32 WebRtcSpl_MaxAbsValueW32;
+int32_t WebRtcSpl_MaxAbsValueW32C(const int32_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MaxAbsValueW32Neon(const int32_t* vector, size_t length);
+#endif
+#if defined(MIPS_DSP_R1_LE)
+int32_t WebRtcSpl_MaxAbsValueW32_mips(const int32_t* vector, size_t length);
+#endif
+
+// Returns the maximum value of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Maximum sample value in `vector`.
+typedef int16_t (*MaxValueW16)(const int16_t* vector, size_t length);
+extern const MaxValueW16 WebRtcSpl_MaxValueW16;
+int16_t WebRtcSpl_MaxValueW16C(const int16_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length);
+#endif
+
+// Returns the maximum value of a 32-bit vector.
+//
+// Input:
+// - vector : 32-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Maximum sample value in `vector`.
+typedef int32_t (*MaxValueW32)(const int32_t* vector, size_t length);
+extern const MaxValueW32 WebRtcSpl_MaxValueW32;
+int32_t WebRtcSpl_MaxValueW32C(const int32_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length);
+#endif
+#if defined(MIPS32_LE)
+int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length);
+#endif
+
+// Returns the minimum value of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Minimum sample value in `vector`.
+typedef int16_t (*MinValueW16)(const int16_t* vector, size_t length);
+extern const MinValueW16 WebRtcSpl_MinValueW16;
+int16_t WebRtcSpl_MinValueW16C(const int16_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length);
+#endif
+#if defined(MIPS32_LE)
+int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length);
+#endif
+
+// Returns the minimum value of a 32-bit vector.
+//
+// Input:
+// - vector : 32-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Minimum sample value in `vector`.
+typedef int32_t (*MinValueW32)(const int32_t* vector, size_t length);
+extern const MinValueW32 WebRtcSpl_MinValueW32;
+int32_t WebRtcSpl_MinValueW32C(const int32_t* vector, size_t length);
+#if defined(WEBRTC_HAS_NEON)
+int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length);
+#endif
+#if defined(MIPS32_LE)
+int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length);
+#endif
+
+// Returns both the minimum and maximum values of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+// Ouput:
+// - max_val : Maximum sample value in `vector`.
+// - min_val : Minimum sample value in `vector`.
+void WebRtcSpl_MinMaxW16(const int16_t* vector,
+ size_t length,
+ int16_t* min_val,
+ int16_t* max_val);
+#if defined(WEBRTC_HAS_NEON)
+void WebRtcSpl_MinMaxW16Neon(const int16_t* vector,
+ size_t length,
+ int16_t* min_val,
+ int16_t* max_val);
+#endif
+
+// Returns the vector index to the largest absolute value of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Index to the maximum absolute value in vector.
+// If there are multiple equal maxima, return the index of the
+// first. -32768 will always have precedence over 32767 (despite
+// -32768 presenting an int16 absolute value of 32767).
+size_t WebRtcSpl_MaxAbsIndexW16(const int16_t* vector, size_t length);
+
+// Returns the element with the largest absolute value of a 16-bit vector. Note
+// that this function can return a negative value.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : The element with the largest absolute value. Note that this
+// may be a negative value.
+int16_t WebRtcSpl_MaxAbsElementW16(const int16_t* vector, size_t length);
+
+// Returns the vector index to the maximum sample value of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Index to the maximum value in vector (if multiple
+// indexes have the maximum, return the first).
+size_t WebRtcSpl_MaxIndexW16(const int16_t* vector, size_t length);
+
+// Returns the vector index to the maximum sample value of a 32-bit vector.
+//
+// Input:
+// - vector : 32-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Index to the maximum value in vector (if multiple
+// indexes have the maximum, return the first).
+size_t WebRtcSpl_MaxIndexW32(const int32_t* vector, size_t length);
+
+// Returns the vector index to the minimum sample value of a 16-bit vector.
+//
+// Input:
+// - vector : 16-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Index to the mimimum value in vector (if multiple
+// indexes have the minimum, return the first).
+size_t WebRtcSpl_MinIndexW16(const int16_t* vector, size_t length);
+
+// Returns the vector index to the minimum sample value of a 32-bit vector.
+//
+// Input:
+// - vector : 32-bit input vector.
+// - length : Number of samples in vector.
+//
+// Return value : Index to the mimimum value in vector (if multiple
+// indexes have the minimum, return the first).
+size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length);
+
+// End: Minimum and maximum operations.
+
+// Vector scaling operations. Implementation in vector_scaling_operations.c.
+// Description at bottom of file.
+void WebRtcSpl_VectorBitShiftW16(int16_t* out_vector,
+ size_t vector_length,
+ const int16_t* in_vector,
+ int16_t right_shifts);
+void WebRtcSpl_VectorBitShiftW32(int32_t* out_vector,
+ size_t vector_length,
+ const int32_t* in_vector,
+ int16_t right_shifts);
+void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out_vector,
+ size_t vector_length,
+ const int32_t* in_vector,
+ int right_shifts);
+void WebRtcSpl_ScaleVector(const int16_t* in_vector,
+ int16_t* out_vector,
+ int16_t gain,
+ size_t vector_length,
+ int16_t right_shifts);
+void WebRtcSpl_ScaleVectorWithSat(const int16_t* in_vector,
+ int16_t* out_vector,
+ int16_t gain,
+ size_t vector_length,
+ int16_t right_shifts);
+void WebRtcSpl_ScaleAndAddVectors(const int16_t* in_vector1,
+ int16_t gain1,
+ int right_shifts1,
+ const int16_t* in_vector2,
+ int16_t gain2,
+ int right_shifts2,
+ int16_t* out_vector,
+ size_t vector_length);
+
+// The functions (with related pointer) perform the vector operation:
+// out_vector[k] = ((scale1 * in_vector1[k]) + (scale2 * in_vector2[k])
+// + round_value) >> right_shifts,
+// where round_value = (1 << right_shifts) >> 1.
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - in_vector1_scale : Gain to be used for vector 1
+// - in_vector2 : Input vector 2
+// - in_vector2_scale : Gain to be used for vector 2
+// - right_shifts : Number of right bit shifts to be applied
+// - length : Number of elements in the input vectors
+//
+// Output:
+// - out_vector : Output vector
+// Return value : 0 if OK, -1 if (in_vector1 == null
+// || in_vector2 == null || out_vector == null
+// || length <= 0 || right_shift < 0).
+typedef int (*ScaleAndAddVectorsWithRound)(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);
+extern const ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
+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);
+#if defined(MIPS_DSP_R1_LE)
+int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(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);
+#endif
+// End: Vector scaling operations.
+
+// iLBC specific functions. Implementations in ilbc_specific_functions.c.
+// Description at bottom of file.
+void WebRtcSpl_ReverseOrderMultArrayElements(int16_t* out_vector,
+ const int16_t* in_vector,
+ const int16_t* window,
+ size_t vector_length,
+ int16_t right_shifts);
+void WebRtcSpl_ElementwiseVectorMult(int16_t* out_vector,
+ const int16_t* in_vector,
+ const int16_t* window,
+ size_t vector_length,
+ int16_t right_shifts);
+void WebRtcSpl_AddVectorsAndShift(int16_t* out_vector,
+ const int16_t* in_vector1,
+ const int16_t* in_vector2,
+ size_t vector_length,
+ int16_t right_shifts);
+void WebRtcSpl_AddAffineVectorToVector(int16_t* out_vector,
+ const int16_t* in_vector,
+ int16_t gain,
+ int32_t add_constant,
+ int16_t right_shifts,
+ size_t vector_length);
+void WebRtcSpl_AffineTransformVector(int16_t* out_vector,
+ const int16_t* in_vector,
+ int16_t gain,
+ int32_t add_constant,
+ int16_t right_shifts,
+ size_t vector_length);
+// End: iLBC specific functions.
+
+// Signal processing operations.
+
+// A 32-bit fix-point implementation of auto-correlation computation
+//
+// Input:
+// - in_vector : Vector to calculate autocorrelation upon
+// - in_vector_length : Length (in samples) of `vector`
+// - order : The order up to which the autocorrelation should be
+// calculated
+//
+// Output:
+// - result : auto-correlation values (values should be seen
+// relative to each other since the absolute values
+// might have been down shifted to avoid overflow)
+//
+// - scale : The number of left shifts required to obtain the
+// auto-correlation in Q0
+//
+// Return value : Number of samples in `result`, i.e. (order+1)
+size_t WebRtcSpl_AutoCorrelation(const int16_t* in_vector,
+ size_t in_vector_length,
+ size_t order,
+ int32_t* result,
+ int* scale);
+
+// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
+// does NOT use the 64 bit class
+//
+// Input:
+// - auto_corr : Vector with autocorrelation values of length >= `order`+1
+// - order : The LPC filter order (support up to order 20)
+//
+// Output:
+// - lpc_coef : lpc_coef[0..order] LPC coefficients in Q12
+// - refl_coef : refl_coef[0...order-1]| Reflection coefficients in Q15
+//
+// Return value : 1 for stable 0 for unstable
+int16_t WebRtcSpl_LevinsonDurbin(const int32_t* auto_corr,
+ int16_t* lpc_coef,
+ int16_t* refl_coef,
+ size_t order);
+
+// Converts reflection coefficients `refl_coef` to LPC coefficients `lpc_coef`.
+// This version is a 16 bit operation.
+//
+// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
+// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
+// "rare" cases even if the reflection coefficients are stable.
+//
+// Input:
+// - refl_coef : Reflection coefficients in Q15 that should be converted
+// to LPC coefficients
+// - use_order : Number of coefficients in `refl_coef`
+//
+// Output:
+// - lpc_coef : LPC coefficients in Q12
+void WebRtcSpl_ReflCoefToLpc(const int16_t* refl_coef,
+ int use_order,
+ int16_t* lpc_coef);
+
+// Converts LPC coefficients `lpc_coef` to reflection coefficients `refl_coef`.
+// This version is a 16 bit operation.
+// The conversion is implemented by the step-down algorithm.
+//
+// Input:
+// - lpc_coef : LPC coefficients in Q12, that should be converted to
+// reflection coefficients
+// - use_order : Number of coefficients in `lpc_coef`
+//
+// Output:
+// - refl_coef : Reflection coefficients in Q15.
+void WebRtcSpl_LpcToReflCoef(int16_t* lpc_coef,
+ int use_order,
+ int16_t* refl_coef);
+
+// Calculates reflection coefficients (16 bit) from auto-correlation values
+//
+// Input:
+// - auto_corr : Auto-correlation values
+// - use_order : Number of coefficients wanted be calculated
+//
+// Output:
+// - refl_coef : Reflection coefficients in Q15.
+void WebRtcSpl_AutoCorrToReflCoef(const int32_t* auto_corr,
+ int use_order,
+ int16_t* refl_coef);
+
+// The functions (with related pointer) calculate the cross-correlation between
+// two sequences `seq1` and `seq2`.
+// `seq1` is fixed and `seq2` slides as the pointer is increased with the
+// amount `step_seq2`. Note the arguments should obey the relationship:
+// `dim_seq` - 1 + `step_seq2` * (`dim_cross_correlation` - 1) <
+// buffer size of `seq2`
+//
+// Input:
+// - seq1 : First sequence (fixed throughout the correlation)
+// - seq2 : Second sequence (slides `step_vector2` for each
+// new correlation)
+// - dim_seq : Number of samples to use in the cross-correlation
+// - dim_cross_correlation : Number of cross-correlations to calculate (the
+// start position for `vector2` is updated for each
+// new one)
+// - right_shifts : Number of right bit shifts to use. This will
+// become the output Q-domain.
+// - step_seq2 : How many (positive or negative) steps the
+// `vector2` pointer should be updated for each new
+// cross-correlation value.
+//
+// Output:
+// - cross_correlation : The cross-correlation in Q(-right_shifts)
+typedef void (*CrossCorrelation)(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);
+extern const CrossCorrelation WebRtcSpl_CrossCorrelation;
+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);
+#if defined(WEBRTC_HAS_NEON)
+void WebRtcSpl_CrossCorrelationNeon(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);
+#endif
+#if defined(MIPS32_LE)
+void WebRtcSpl_CrossCorrelation_mips(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);
+#endif
+
+// Creates (the first half of) a Hanning window. Size must be at least 1 and
+// at most 512.
+//
+// Input:
+// - size : Length of the requested Hanning window (1 to 512)
+//
+// Output:
+// - window : Hanning vector in Q14.
+void WebRtcSpl_GetHanningWindow(int16_t* window, size_t size);
+
+// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
+// `in_vector`. Input and output values are in Q15.
+//
+// Inputs:
+// - in_vector : Values to calculate sqrt(1 - x^2) of
+// - vector_length : Length of vector `in_vector`
+//
+// Output:
+// - out_vector : Output values in Q15
+void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t* in_vector,
+ size_t vector_length,
+ int16_t* out_vector);
+// End: Signal processing operations.
+
+// Randomization functions. Implementations collected in
+// randomization_functions.c and descriptions at bottom of this file.
+int16_t WebRtcSpl_RandU(uint32_t* seed);
+int16_t WebRtcSpl_RandN(uint32_t* seed);
+int16_t WebRtcSpl_RandUArray(int16_t* vector,
+ int16_t vector_length,
+ uint32_t* seed);
+// End: Randomization functions.
+
+// Math functions
+int32_t WebRtcSpl_Sqrt(int32_t value);
+
+// Divisions. Implementations collected in division_operations.c and
+// descriptions at bottom of this file.
+uint32_t WebRtcSpl_DivU32U16(uint32_t num, uint16_t den);
+int32_t WebRtcSpl_DivW32W16(int32_t num, int16_t den);
+int16_t WebRtcSpl_DivW32W16ResW16(int32_t num, int16_t den);
+int32_t WebRtcSpl_DivResultInQ31(int32_t num, int32_t den);
+int32_t WebRtcSpl_DivW32HiLow(int32_t num, int16_t den_hi, int16_t den_low);
+// End: Divisions.
+
+int32_t WebRtcSpl_Energy(int16_t* vector,
+ size_t vector_length,
+ int* scale_factor);
+
+// Filter operations.
+size_t WebRtcSpl_FilterAR(const int16_t* ar_coef,
+ size_t ar_coef_length,
+ const int16_t* in_vector,
+ size_t in_vector_length,
+ int16_t* filter_state,
+ size_t filter_state_length,
+ int16_t* filter_state_low,
+ size_t filter_state_low_length,
+ int16_t* out_vector,
+ int16_t* out_vector_low,
+ size_t out_vector_low_length);
+
+// WebRtcSpl_FilterMAFastQ12(...)
+//
+// Performs a MA filtering on a vector in Q12
+//
+// Input:
+// - in_vector : Input samples (state in positions
+// in_vector[-order] .. in_vector[-1])
+// - ma_coef : Filter coefficients (in Q12)
+// - ma_coef_length : Number of B coefficients (order+1)
+// - vector_length : Number of samples to be filtered
+//
+// Output:
+// - out_vector : Filtered samples
+//
+void WebRtcSpl_FilterMAFastQ12(const int16_t* in_vector,
+ int16_t* out_vector,
+ const int16_t* ma_coef,
+ size_t ma_coef_length,
+ size_t vector_length);
+
+// Performs a AR filtering on a vector in Q12
+// Input:
+// - data_in : Input samples
+// - data_out : State information in positions
+// data_out[-order] .. data_out[-1]
+// - coefficients : Filter coefficients (in Q12)
+// - coefficients_length: Number of coefficients (order+1)
+// - data_length : Number of samples to be filtered
+// Output:
+// - data_out : Filtered samples
+void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
+ int16_t* data_out,
+ const int16_t* __restrict coefficients,
+ size_t coefficients_length,
+ size_t data_length);
+
+// The functions (with related pointer) perform a MA down sampling filter
+// on a vector.
+// Input:
+// - data_in : Input samples (state in positions
+// data_in[-order] .. data_in[-1])
+// - data_in_length : Number of samples in `data_in` to be filtered.
+// This must be at least
+// `delay` + `factor`*(`out_vector_length`-1) + 1)
+// - data_out_length : Number of down sampled samples desired
+// - coefficients : Filter coefficients (in Q12)
+// - coefficients_length: Number of coefficients (order+1)
+// - factor : Decimation factor
+// - delay : Delay of filter (compensated for in out_vector)
+// Output:
+// - data_out : Filtered samples
+// Return value : 0 if OK, -1 if `in_vector` is too short
+typedef int (*DownsampleFast)(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);
+extern const DownsampleFast WebRtcSpl_DownsampleFast;
+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);
+#if defined(WEBRTC_HAS_NEON)
+int WebRtcSpl_DownsampleFastNeon(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);
+#endif
+#if defined(MIPS32_LE)
+int WebRtcSpl_DownsampleFast_mips(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);
+#endif
+
+// End: Filter operations.
+
+// FFT operations
+
+int WebRtcSpl_ComplexFFT(int16_t vector[], int stages, int mode);
+int WebRtcSpl_ComplexIFFT(int16_t vector[], int stages, int mode);
+
+// Treat a 16-bit complex data buffer `complex_data` as an array of 32-bit
+// values, and swap elements whose indexes are bit-reverses of each other.
+//
+// Input:
+// - complex_data : Complex data buffer containing 2^`stages` real
+// elements interleaved with 2^`stages` imaginary
+// elements: [Re Im Re Im Re Im....]
+// - stages : Number of FFT stages. Must be at least 3 and at most
+// 10, since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// Output:
+// - complex_data : The complex data buffer.
+
+void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
+
+// End: FFT operations
+
+/************************************************************
+ *
+ * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
+ *
+ ************************************************************/
+
+/*******************************************************************
+ * resample.c
+ *
+ * Includes the following resampling combinations
+ * 22 kHz -> 16 kHz
+ * 16 kHz -> 22 kHz
+ * 22 kHz -> 8 kHz
+ * 8 kHz -> 22 kHz
+ *
+ ******************************************************************/
+
+// state structure for 22 -> 16 resampler
+typedef struct {
+ int32_t S_22_44[8];
+ int32_t S_44_32[8];
+ int32_t S_32_16[8];
+} WebRtcSpl_State22khzTo16khz;
+
+void WebRtcSpl_Resample22khzTo16khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State22khzTo16khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
+
+// state structure for 16 -> 22 resampler
+typedef struct {
+ int32_t S_16_32[8];
+ int32_t S_32_22[8];
+} WebRtcSpl_State16khzTo22khz;
+
+void WebRtcSpl_Resample16khzTo22khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State16khzTo22khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
+
+// state structure for 22 -> 8 resampler
+typedef struct {
+ int32_t S_22_22[16];
+ int32_t S_22_16[8];
+ int32_t S_16_8[8];
+} WebRtcSpl_State22khzTo8khz;
+
+void WebRtcSpl_Resample22khzTo8khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State22khzTo8khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
+
+// state structure for 8 -> 22 resampler
+typedef struct {
+ int32_t S_8_16[8];
+ int32_t S_16_11[8];
+ int32_t S_11_22[8];
+} WebRtcSpl_State8khzTo22khz;
+
+void WebRtcSpl_Resample8khzTo22khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State8khzTo22khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
+
+/*******************************************************************
+ * resample_fractional.c
+ * Functions for internal use in the other resample functions
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 32 kHz
+ * 32 kHz -> 24 kHz
+ * 44 kHz -> 32 kHz
+ *
+ ******************************************************************/
+
+void WebRtcSpl_Resample48khzTo32khz(const int32_t* In, int32_t* Out, size_t K);
+
+void WebRtcSpl_Resample32khzTo24khz(const int32_t* In, int32_t* Out, size_t K);
+
+void WebRtcSpl_Resample44khzTo32khz(const int32_t* In, int32_t* Out, size_t K);
+
+/*******************************************************************
+ * resample_48khz.c
+ *
+ * Includes the following resampling combinations
+ * 48 kHz -> 16 kHz
+ * 16 kHz -> 48 kHz
+ * 48 kHz -> 8 kHz
+ * 8 kHz -> 48 kHz
+ *
+ ******************************************************************/
+
+typedef struct {
+ int32_t S_48_48[16];
+ int32_t S_48_32[8];
+ int32_t S_32_16[8];
+} WebRtcSpl_State48khzTo16khz;
+
+void WebRtcSpl_Resample48khzTo16khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State48khzTo16khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
+
+typedef struct {
+ int32_t S_16_32[8];
+ int32_t S_32_24[8];
+ int32_t S_24_48[8];
+} WebRtcSpl_State16khzTo48khz;
+
+void WebRtcSpl_Resample16khzTo48khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State16khzTo48khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
+
+typedef struct {
+ int32_t S_48_24[8];
+ int32_t S_24_24[16];
+ int32_t S_24_16[8];
+ int32_t S_16_8[8];
+} WebRtcSpl_State48khzTo8khz;
+
+void WebRtcSpl_Resample48khzTo8khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State48khzTo8khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
+
+typedef struct {
+ int32_t S_8_16[8];
+ int32_t S_16_12[8];
+ int32_t S_12_24[8];
+ int32_t S_24_48[8];
+} WebRtcSpl_State8khzTo48khz;
+
+void WebRtcSpl_Resample8khzTo48khz(const int16_t* in,
+ int16_t* out,
+ WebRtcSpl_State8khzTo48khz* state,
+ int32_t* tmpmem);
+
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
+
+/*******************************************************************
+ * resample_by_2.c
+ *
+ * Includes down and up sampling by a factor of two.
+ *
+ ******************************************************************/
+
+void WebRtcSpl_DownsampleBy2(const int16_t* in,
+ size_t len,
+ int16_t* out,
+ int32_t* filtState);
+
+void WebRtcSpl_UpsampleBy2(const int16_t* in,
+ size_t len,
+ int16_t* out,
+ int32_t* filtState);
+
+/************************************************************
+ * END OF RESAMPLING FUNCTIONS
+ ************************************************************/
+void WebRtcSpl_AnalysisQMF(const int16_t* in_data,
+ size_t in_data_length,
+ int16_t* low_band,
+ int16_t* high_band,
+ int32_t* filter_state1,
+ int32_t* filter_state2);
+void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
+ const int16_t* high_band,
+ size_t band_length,
+ int16_t* out_data,
+ int32_t* filter_state1,
+ int32_t* filter_state2);
+
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
+
+//
+// WebRtcSpl_AddSatW16(...)
+// WebRtcSpl_AddSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
+// the numbers specified by the `var1` and `var2` parameters.
+//
+// Input:
+// - var1 : Input variable 1
+// - var2 : Input variable 2
+//
+// Return value : Added and saturated value
+//
+
+//
+// WebRtcSpl_SubSatW16(...)
+// WebRtcSpl_SubSatW32(...)
+//
+// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
+// of the numbers specified by the `var1` and `var2` parameters.
+//
+// Input:
+// - var1 : Input variable 1
+// - var2 : Input variable 2
+//
+// Returned value : Subtracted and saturated value
+//
+
+//
+// WebRtcSpl_GetSizeInBits(...)
+//
+// Returns the # of bits that are needed at the most to represent the number
+// specified by the `value` parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bits needed to represent `value`
+//
+
+//
+// WebRtcSpl_NormW32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
+// signed number specified by the `value` parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize `value`
+//
+
+//
+// WebRtcSpl_NormW16(...)
+//
+// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
+// signed number specified by the `value` parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize `value`
+//
+
+//
+// WebRtcSpl_NormU32(...)
+//
+// Norm returns the # of left shifts required to 32-bit normalize the unsigned
+// 32-bit number specified by the `value` parameter.
+//
+// Input:
+// - value : Input value
+//
+// Return value : Number of bit shifts needed to 32-bit normalize `value`
+//
+
+//
+// WebRtcSpl_GetScalingSquare(...)
+//
+// Returns the # of bits required to scale the samples specified in the
+// `in_vector` parameter so that, if the squares of the samples are added the
+// # of times specified by the `times` parameter, the 32-bit addition will not
+// overflow (result in int32_t).
+//
+// Input:
+// - in_vector : Input vector to check scaling on
+// - in_vector_length : Samples in `in_vector`
+// - times : Number of additions to be performed
+//
+// Return value : Number of right bit shifts needed to avoid
+// overflow in the addition calculation
+//
+
+//
+// WebRtcSpl_MemSetW16(...)
+//
+// Sets all the values in the int16_t vector `vector` of length
+// `vector_length` to the specified value `set_value`
+//
+// Input:
+// - vector : Pointer to the int16_t vector
+// - set_value : Value specified
+// - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemSetW32(...)
+//
+// Sets all the values in the int32_t vector `vector` of length
+// `vector_length` to the specified value `set_value`
+//
+// Input:
+// - vector : Pointer to the int16_t vector
+// - set_value : Value specified
+// - vector_length : Length of vector
+//
+
+//
+// WebRtcSpl_MemCpyReversedOrder(...)
+//
+// Copies all the values from the source int16_t vector `in_vector` to a
+// destination int16_t vector `out_vector`. It is done in reversed order,
+// meaning that the first sample of `in_vector` is copied to the last sample of
+// the `out_vector`. The procedure continues until the last sample of
+// `in_vector` has been copied to the first sample of `out_vector`. This
+// creates a reversed vector. Used in e.g. prediction in iLBC.
+//
+// Input:
+// - in_vector : Pointer to the first sample in a int16_t vector
+// of length `length`
+// - vector_length : Number of elements to copy
+//
+// Output:
+// - out_vector : Pointer to the last sample in a int16_t vector
+// of length `length`
+//
+
+//
+// WebRtcSpl_CopyFromEndW16(...)
+//
+// Copies the rightmost `samples` of `in_vector` (of length `in_vector_length`)
+// to the vector `out_vector`.
+//
+// Input:
+// - in_vector : Input vector
+// - in_vector_length : Number of samples in `in_vector`
+// - samples : Number of samples to extract (from right side)
+// from `in_vector`
+//
+// Output:
+// - out_vector : Vector with the requested samples
+//
+
+//
+// WebRtcSpl_ZerosArrayW16(...)
+// WebRtcSpl_ZerosArrayW32(...)
+//
+// Inserts the value "zero" in all positions of a w16 and a w32 vector
+// respectively.
+//
+// Input:
+// - vector_length : Number of samples in vector
+//
+// Output:
+// - vector : Vector containing all zeros
+//
+
+//
+// WebRtcSpl_VectorBitShiftW16(...)
+// WebRtcSpl_VectorBitShiftW32(...)
+//
+// Bit shifts all the values in a vector up or downwards. Different calls for
+// int16_t and int32_t vectors respectively.
+//
+// Input:
+// - vector_length : Length of vector
+// - in_vector : Pointer to the vector that should be bit shifted
+// - right_shifts : Number of right bit shifts (negative value gives left
+// shifts)
+//
+// Output:
+// - out_vector : Pointer to the result vector (can be the same as
+// `in_vector`)
+//
+
+//
+// WebRtcSpl_VectorBitShiftW32ToW16(...)
+//
+// Bit shifts all the values in a int32_t vector up or downwards and
+// stores the result as an int16_t vector. The function will saturate the
+// signal if needed, before storing in the output vector.
+//
+// Input:
+// - vector_length : Length of vector
+// - in_vector : Pointer to the vector that should be bit shifted
+// - right_shifts : Number of right bit shifts (negative value gives left
+// shifts)
+//
+// Output:
+// - out_vector : Pointer to the result vector (can be the same as
+// `in_vector`)
+//
+
+//
+// WebRtcSpl_ScaleVector(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (gain*in_vector[k])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Scaling gain
+// - vector_length : Elements in the `in_vector`
+// - right_shifts : Number of right bit shifts applied
+//
+// Output:
+// - out_vector : Output vector (can be the same as `in_vector`)
+//
+
+//
+// WebRtcSpl_ScaleVectorWithSat(...)
+//
+// Performs the vector operation:
+// out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Scaling gain
+// - vector_length : Elements in the `in_vector`
+// - right_shifts : Number of right bit shifts applied
+//
+// Output:
+// - out_vector : Output vector (can be the same as `in_vector`)
+//
+
+//
+// WebRtcSpl_ScaleAndAddVectors(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
+// + (gain2*in_vector2[k])>>right_shifts2
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - gain1 : Gain to be used for vector 1
+// - right_shifts1 : Right bit shift to be used for vector 1
+// - in_vector2 : Input vector 2
+// - gain2 : Gain to be used for vector 2
+// - right_shifts2 : Right bit shift to be used for vector 2
+// - vector_length : Elements in the input vectors
+//
+// Output:
+// - out_vector : Output vector
+//
+
+//
+// WebRtcSpl_ReverseOrderMultArrayElements(...)
+//
+// Performs the vector operation:
+// out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - window : Window vector (should be reversed). The pointer
+// should be set to the last value in the vector
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in `in_vector`
+//
+// Output:
+// - out_vector : Output vector (can be same as `in_vector`)
+//
+
+//
+// WebRtcSpl_ElementwiseVectorMult(...)
+//
+// Performs the vector operation:
+// out_vector[n] = (in_vector[n]*window[n])>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - window : Window vector.
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in `in_vector`
+//
+// Output:
+// - out_vector : Output vector (can be same as `in_vector`)
+//
+
+//
+// WebRtcSpl_AddVectorsAndShift(...)
+//
+// Performs the vector operation:
+// out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
+//
+// Input:
+// - in_vector1 : Input vector 1
+// - in_vector2 : Input vector 2
+// - right_shifts : Number of right bit shift to be applied after the
+// multiplication
+// - vector_length : Number of elements in `in_vector1` and `in_vector2`
+//
+// Output:
+// - out_vector : Output vector (can be same as `in_vector1`)
+//
+
+//
+// WebRtcSpl_AddAffineVectorToVector(...)
+//
+// Adds an affine transformed vector to another vector `out_vector`, i.e,
+// performs
+// out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Gain value, used to multiply the in vector with
+// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
+// but others can be used as well
+// - right_shifts : Number of right bit shifts (0-16)
+// - vector_length : Number of samples in `in_vector` and `out_vector`
+//
+// Output:
+// - out_vector : Vector with the output
+//
+
+//
+// WebRtcSpl_AffineTransformVector(...)
+//
+// Affine transforms a vector, i.e, performs
+// out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
+//
+// Input:
+// - in_vector : Input vector
+// - gain : Gain value, used to multiply the in vector with
+// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
+// but others can be used as well
+// - right_shifts : Number of right bit shifts (0-16)
+// - vector_length : Number of samples in `in_vector` and `out_vector`
+//
+// Output:
+// - out_vector : Vector with the output
+//
+
+//
+// WebRtcSpl_IncreaseSeed(...)
+//
+// Increases the seed (and returns the new value)
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : The new seed value
+//
+
+//
+// WebRtcSpl_RandU(...)
+//
+// Produces a uniformly distributed value in the int16_t range
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : Uniformly distributed value in the range
+// [Word16_MIN...Word16_MAX]
+//
+
+//
+// WebRtcSpl_RandN(...)
+//
+// Produces a normal distributed value in the int16_t range
+//
+// Input:
+// - seed : Seed for random calculation
+//
+// Output:
+// - seed : Updated seed value
+//
+// Return value : N(0,1) value in the Q13 domain
+//
+
+//
+// WebRtcSpl_RandUArray(...)
+//
+// Produces a uniformly distributed vector with elements in the int16_t
+// range
+//
+// Input:
+// - vector_length : Samples wanted in the vector
+// - seed : Seed for random calculation
+//
+// Output:
+// - vector : Vector with the uniform values
+// - seed : Updated seed value
+//
+// Return value : Number of samples in vector, i.e., `vector_length`
+//
+
+//
+// WebRtcSpl_Sqrt(...)
+//
+// Returns the square root of the input value `value`. The precision of this
+// function is integer precision, i.e., sqrt(8) gives 2 as answer.
+// If `value` is a negative number then 0 is returned.
+//
+// Algorithm:
+//
+// A sixth order Taylor Series expansion is used here to compute the square
+// root of a number y^0.5 = (1+x)^0.5
+// where
+// x = y-1
+// = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+// 0.5 <= x < 1
+//
+// Input:
+// - value : Value to calculate sqrt of
+//
+// Return value : Result of the sqrt calculation
+//
+
+//
+// WebRtcSpl_DivU32U16(...)
+//
+// Divides a uint32_t `num` by a uint16_t `den`.
+//
+// If `den`==0, (uint32_t)0xFFFFFFFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a uint32_t), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16(...)
+//
+// Divides a int32_t `num` by a int16_t `den`.
+//
+// If `den`==0, (int32_t)0x7FFFFFFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a int32_t), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivW32W16ResW16(...)
+//
+// Divides a int32_t `num` by a int16_t `den`, assuming that the
+// result is less than 32768, otherwise an unpredictable result will occur.
+//
+// If `den`==0, (int16_t)0x7FFF is returned.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division (as a int16_t), i.e., the
+// integer part of num/den.
+//
+
+//
+// WebRtcSpl_DivResultInQ31(...)
+//
+// Divides a int32_t `num` by a int16_t `den`, assuming that the
+// absolute value of the denominator is larger than the numerator, otherwise
+// an unpredictable result will occur.
+//
+// Input:
+// - num : Numerator
+// - den : Denominator
+//
+// Return value : Result of the division in Q31.
+//
+
+//
+// WebRtcSpl_DivW32HiLow(...)
+//
+// Divides a int32_t `num` by a denominator in hi, low format. The
+// absolute value of the denominator has to be larger (or equal to) the
+// numerator.
+//
+// Input:
+// - num : Numerator
+// - den_hi : High part of denominator
+// - den_low : Low part of denominator
+//
+// Return value : Divided value in Q31
+//
+
+//
+// WebRtcSpl_Energy(...)
+//
+// Calculates the energy of a vector
+//
+// Input:
+// - vector : Vector which the energy should be calculated on
+// - vector_length : Number of samples in vector
+//
+// Output:
+// - scale_factor : Number of left bit shifts needed to get the physical
+// energy value, i.e, to get the Q0 value
+//
+// Return value : Energy value in Q(-`scale_factor`)
+//
+
+//
+// WebRtcSpl_FilterAR(...)
+//
+// Performs a 32-bit AR filtering on a vector in Q12
+//
+// Input:
+// - ar_coef : AR-coefficient vector (values in Q12),
+// ar_coef[0] must be 4096.
+// - ar_coef_length : Number of coefficients in `ar_coef`.
+// - in_vector : Vector to be filtered.
+// - in_vector_length : Number of samples in `in_vector`.
+// - filter_state : Current state (higher part) of the filter.
+// - filter_state_length : Length (in samples) of `filter_state`.
+// - filter_state_low : Current state (lower part) of the filter.
+// - filter_state_low_length : Length (in samples) of `filter_state_low`.
+// - out_vector_low_length : Maximum length (in samples) of
+// `out_vector_low`.
+//
+// Output:
+// - filter_state : Updated state (upper part) vector.
+// - filter_state_low : Updated state (lower part) vector.
+// - out_vector : Vector containing the upper part of the
+// filtered values.
+// - out_vector_low : Vector containing the lower part of the
+// filtered values.
+//
+// Return value : Number of samples in the `out_vector`.
+//
+
+//
+// WebRtcSpl_ComplexIFFT(...)
+//
+// Complex Inverse FFT
+//
+// Computes an inverse complex 2^`stages`-point FFT on the input vector, which
+// is in bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With X as the input complex vector, y as the output complex
+// vector and with M = 2^`stages`, the following is computed:
+//
+// M-1
+// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+// i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// Input:
+// - vector : In pointer to complex vector containing 2^`stages`
+// real elements interleaved with 2^`stages` imaginary
+// elements.
+// [ReImReImReIm....]
+// The elements are in Q(-scale) domain, see more on Return
+// Value below.
+//
+// - stages : Number of FFT stages. Must be at least 3 and at most 10,
+// since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// - mode : This parameter gives the user to choose how the FFT
+// should work.
+// mode==0: Low-complexity and Low-accuracy mode
+// mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+// - vector : Out pointer to the FFT vector (the same as input).
+//
+// Return Value : The scale value that tells the number of left bit shifts
+// that the elements in the `vector` should be shifted with
+// in order to get Q0 values, i.e. the physically correct
+// values. The scale parameter is always 0 or positive,
+// except if N>1024 (`stages`>10), which returns a scale
+// value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_ComplexFFT(...)
+//
+// Complex FFT
+//
+// Computes a complex 2^`stages`-point FFT on the input vector, which is in
+// bit-reversed order. The original content of the vector is destroyed in
+// the process, since the input is overwritten by the output, normal-ordered,
+// FFT vector. With x as the input complex vector, Y as the output complex
+// vector and with M = 2^`stages`, the following is computed:
+//
+// M-1
+// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
+// i=0
+//
+// The implementations are optimized for speed, not for code size. It uses the
+// decimation-in-time algorithm with radix-2 butterfly technique.
+//
+// This routine prevents overflow by scaling by 2 before each FFT stage. This is
+// a fixed scaling, for proper normalization - there will be log2(n) passes, so
+// this results in an overall factor of 1/n, distributed to maximize arithmetic
+// accuracy.
+//
+// Input:
+// - vector : In pointer to complex vector containing 2^`stages` real
+// elements interleaved with 2^`stages` imaginary elements.
+// [ReImReImReIm....]
+// The output is in the Q0 domain.
+//
+// - stages : Number of FFT stages. Must be at least 3 and at most 10,
+// since the table WebRtcSpl_kSinTable1024[] is 1024
+// elements long.
+//
+// - mode : This parameter gives the user to choose how the FFT
+// should work.
+// mode==0: Low-complexity and Low-accuracy mode
+// mode==1: High-complexity and High-accuracy mode
+//
+// Output:
+// - vector : The output FFT vector is in the Q0 domain.
+//
+// Return value : The scale parameter is always 0, except if N>1024,
+// which returns a scale value of -1, indicating error.
+//
+
+//
+// WebRtcSpl_AnalysisQMF(...)
+//
+// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
+// current version has F = 8000, therefore, a super-wideband audio signal is
+// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
+//
+// Input:
+// - in_data : Wide band speech signal, 320 samples (10 ms)
+//
+// Input & Output:
+// - filter_state1 : Filter state for first All-pass filter
+// - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+// - low_band : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
+// - high_band : Upper-band signal 8-16 kHz band (flipped in frequency
+// domain), 160 samples (10 ms)
+//
+
+//
+// WebRtcSpl_SynthesisQMF(...)
+//
+// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
+// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
+// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
+// kHz audio.
+//
+// Input:
+// - low_band : The signal with the 0-8 kHz band, 160 samples (10 ms)
+// - high_band : The signal with the 8-16 kHz band, 160 samples (10 ms)
+//
+// Input & Output:
+// - filter_state1 : Filter state for first All-pass filter
+// - filter_state2 : Filter state for second All-pass filter
+//
+// Output:
+// - out_data : Super-wideband speech signal, 0-16 kHz
+//
+
+// int16_t WebRtcSpl_SatW32ToW16(...)
+//
+// This function saturates a 32-bit word into a 16-bit word.
+//
+// Input:
+// - value32 : The value of a 32-bit word.
+//
+// Output:
+// - out16 : the saturated 16-bit word.
+//
+
+// int32_t WebRtc_MulAccumW16(...)
+//
+// This function multiply a 16-bit word by a 16-bit word, and accumulate this
+// value to a 32-bit integer.
+//
+// Input:
+// - a : The value of the first 16-bit word.
+// - b : The value of the second 16-bit word.
+// - c : The value of an 32-bit integer.
+//
+// Return Value: The value of a * b + c.
+//
diff --git a/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl.h b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl.h
new file mode 100644
index 0000000000..2b0995886a
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl.h
@@ -0,0 +1,155 @@
+/*
+ * 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 header file includes the inline functions in
+// the fix point signal processing library.
+
+#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
+
+#include <stdint.h>
+
+#include "rtc_base/compile_assert_c.h"
+
+extern const int8_t kWebRtcSpl_CountLeadingZeros32_Table[64];
+
+// Don't call this directly except in tests!
+static __inline int WebRtcSpl_CountLeadingZeros32_NotBuiltin(uint32_t n) {
+ // Normalize n by rounding up to the nearest number that is a sequence of 0
+ // bits followed by a sequence of 1 bits. This number has the same number of
+ // leading zeros as the original n. There are exactly 33 such values.
+ n |= n >> 1;
+ n |= n >> 2;
+ n |= n >> 4;
+ n |= n >> 8;
+ n |= n >> 16;
+
+ // Multiply the modified n with a constant selected (by exhaustive search)
+ // such that each of the 33 possible values of n give a product whose 6 most
+ // significant bits are unique. Then look up the answer in the table.
+ return kWebRtcSpl_CountLeadingZeros32_Table[(n * 0x8c0b2891) >> 26];
+}
+
+// Don't call this directly except in tests!
+static __inline int WebRtcSpl_CountLeadingZeros64_NotBuiltin(uint64_t n) {
+ const int leading_zeros = n >> 32 == 0 ? 32 : 0;
+ return leading_zeros + WebRtcSpl_CountLeadingZeros32_NotBuiltin(
+ (uint32_t)(n >> (32 - leading_zeros)));
+}
+
+// Returns the number of leading zero bits in the argument.
+static __inline int WebRtcSpl_CountLeadingZeros32(uint32_t n) {
+#ifdef __GNUC__
+ RTC_COMPILE_ASSERT(sizeof(unsigned int) == sizeof(uint32_t));
+ return n == 0 ? 32 : __builtin_clz(n);
+#else
+ return WebRtcSpl_CountLeadingZeros32_NotBuiltin(n);
+#endif
+}
+
+// Returns the number of leading zero bits in the argument.
+static __inline int WebRtcSpl_CountLeadingZeros64(uint64_t n) {
+#ifdef __GNUC__
+ RTC_COMPILE_ASSERT(sizeof(unsigned long long) == sizeof(uint64_t)); // NOLINT
+ return n == 0 ? 64 : __builtin_clzll(n);
+#else
+ return WebRtcSpl_CountLeadingZeros64_NotBuiltin(n);
+#endif
+}
+
+#ifdef WEBRTC_ARCH_ARM_V7
+#include "common_audio/signal_processing/include/spl_inl_armv7.h"
+#else
+
+#if defined(MIPS32_LE)
+#include "common_audio/signal_processing/include/spl_inl_mips.h"
+#endif
+
+#if !defined(MIPS_DSP_R1_LE)
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+ int16_t out16 = (int16_t)value32;
+
+ if (value32 > 32767)
+ out16 = 32767;
+ else if (value32 < -32768)
+ out16 = -32768;
+
+ return out16;
+}
+
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t a, int32_t b) {
+ // Do the addition in unsigned numbers, since signed overflow is undefined
+ // behavior.
+ const int32_t sum = (int32_t)((uint32_t)a + (uint32_t)b);
+
+ // a + b can't overflow if a and b have different signs. If they have the
+ // same sign, a + b also has the same sign iff it didn't overflow.
+ if ((a < 0) == (b < 0) && (a < 0) != (sum < 0)) {
+ // The direction of the overflow is obvious from the sign of a + b.
+ return sum < 0 ? INT32_MAX : INT32_MIN;
+ }
+ return sum;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t a, int32_t b) {
+ // Do the subtraction in unsigned numbers, since signed overflow is undefined
+ // behavior.
+ const int32_t diff = (int32_t)((uint32_t)a - (uint32_t)b);
+
+ // a - b can't overflow if a and b have the same sign. If they have different
+ // signs, a - b has the same sign as a iff it didn't overflow.
+ if ((a < 0) != (b < 0) && (a < 0) != (diff < 0)) {
+ // The direction of the overflow is obvious from the sign of a - b.
+ return diff < 0 ? INT32_MAX : INT32_MIN;
+ }
+ return diff;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+ return WebRtcSpl_SatW32ToW16((int32_t)a + (int32_t)b);
+}
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+ return WebRtcSpl_SatW32ToW16((int32_t)var1 - (int32_t)var2);
+}
+#endif // #if !defined(MIPS_DSP_R1_LE)
+
+#if !defined(MIPS32_LE)
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
+ return 32 - WebRtcSpl_CountLeadingZeros32(n);
+}
+
+// Return the number of steps a can be left-shifted without overflow,
+// or 0 if a == 0.
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
+ return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a < 0 ? ~a : a) - 1;
+}
+
+// Return the number of steps a can be left-shifted without overflow,
+// or 0 if a == 0.
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
+ return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a);
+}
+
+// Return the number of steps a can be left-shifted without overflow,
+// or 0 if a == 0.
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
+ const int32_t a32 = a;
+ return a == 0 ? 0 : WebRtcSpl_CountLeadingZeros32(a < 0 ? ~a32 : a32) - 17;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
+ return (a * b + c);
+}
+#endif // #if !defined(MIPS32_LE)
+
+#endif // WEBRTC_ARCH_ARM_V7
+
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_armv7.h b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_armv7.h
new file mode 100644
index 0000000000..6fc3e7c1b8
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_armv7.h
@@ -0,0 +1,138 @@
+/*
+ * 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 header file includes the inline functions for ARM processors in
+ * the fix point signal processing library.
+ */
+
+#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
+
+#include <stdint.h>
+
+/* TODO(kma): Replace some assembly code with GCC intrinsics
+ * (e.g. __builtin_clz).
+ */
+
+/* This function produces result that is not bit exact with that by the generic
+ * C version in some cases, although the former is at least as accurate as the
+ * later.
+ */
+static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a, int32_t b) {
+ int32_t tmp = 0;
+ __asm __volatile("smulwb %0, %1, %2" : "=r"(tmp) : "r"(b), "r"(a));
+ return tmp;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_16_16(int16_t a, int16_t b) {
+ int32_t tmp = 0;
+ __asm __volatile("smulbb %0, %1, %2" : "=r"(tmp) : "r"(a), "r"(b));
+ return tmp;
+}
+
+// TODO(kma): add unit test.
+static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
+ int32_t tmp = 0;
+ __asm __volatile("smlabb %0, %1, %2, %3"
+ : "=r"(tmp)
+ : "r"(a), "r"(b), "r"(c));
+ return tmp;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+ int32_t s_sum = 0;
+
+ __asm __volatile("qadd16 %0, %1, %2" : "=r"(s_sum) : "r"(a), "r"(b));
+
+ return (int16_t)s_sum;
+}
+
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
+ int32_t l_sum = 0;
+
+ __asm __volatile("qadd %0, %1, %2" : "=r"(l_sum) : "r"(l_var1), "r"(l_var2));
+
+ return l_sum;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
+ int32_t l_sub = 0;
+
+ __asm __volatile("qsub %0, %1, %2" : "=r"(l_sub) : "r"(l_var1), "r"(l_var2));
+
+ return l_sub;
+}
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+ int32_t s_sub = 0;
+
+ __asm __volatile("qsub16 %0, %1, %2" : "=r"(s_sub) : "r"(var1), "r"(var2));
+
+ return (int16_t)s_sub;
+}
+
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
+ int32_t tmp = 0;
+
+ __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(n));
+
+ return (int16_t)(32 - tmp);
+}
+
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
+ int32_t tmp = 0;
+
+ if (a == 0) {
+ return 0;
+ } else if (a < 0) {
+ a ^= 0xFFFFFFFF;
+ }
+
+ __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a));
+
+ return (int16_t)(tmp - 1);
+}
+
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
+ int tmp = 0;
+
+ if (a == 0)
+ return 0;
+
+ __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a));
+
+ return (int16_t)tmp;
+}
+
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
+ int32_t tmp = 0;
+ int32_t a_32 = a;
+
+ if (a_32 == 0) {
+ return 0;
+ } else if (a_32 < 0) {
+ a_32 ^= 0xFFFFFFFF;
+ }
+
+ __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a_32));
+
+ return (int16_t)(tmp - 17);
+}
+
+// TODO(kma): add unit test.
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+ int32_t out = 0;
+
+ __asm __volatile("ssat %0, #16, %1" : "=r"(out) : "r"(value32));
+
+ return (int16_t)out;
+}
+
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_mips.h b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_mips.h
new file mode 100644
index 0000000000..1db95e8254
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/include/spl_inl_mips.h
@@ -0,0 +1,204 @@
+/*
+ * 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.
+ */
+
+// This header file includes the inline functions in
+// the fix point signal processing library.
+
+#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
+
+static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a, int32_t b) {
+ int32_t value32 = 0;
+ int32_t a1 = 0, b1 = 0;
+
+ __asm __volatile(
+#if defined(MIPS32_R2_LE)
+ "seh %[a1], %[a] \n\t"
+ "seh %[b1], %[b] \n\t"
+#else
+ "sll %[a1], %[a], 16 \n\t"
+ "sll %[b1], %[b], 16 \n\t"
+ "sra %[a1], %[a1], 16 \n\t"
+ "sra %[b1], %[b1], 16 \n\t"
+#endif
+ "mul %[value32], %[a1], %[b1] \n\t"
+ : [value32] "=r"(value32), [a1] "=&r"(a1), [b1] "=&r"(b1)
+ : [a] "r"(a), [b] "r"(b)
+ : "hi", "lo");
+ return value32;
+}
+
+static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a, int32_t b) {
+ int32_t value32 = 0, b1 = 0, b2 = 0;
+ int32_t a1 = 0;
+
+ __asm __volatile(
+#if defined(MIPS32_R2_LE)
+ "seh %[a1], %[a] \n\t"
+#else
+ "sll %[a1], %[a], 16 \n\t"
+ "sra %[a1], %[a1], 16 \n\t"
+#endif
+ "andi %[b2], %[b], 0xFFFF \n\t"
+ "sra %[b1], %[b], 16 \n\t"
+ "sra %[b2], %[b2], 1 \n\t"
+ "mul %[value32], %[a1], %[b1] \n\t"
+ "mul %[b2], %[a1], %[b2] \n\t"
+ "addiu %[b2], %[b2], 0x4000 \n\t"
+ "sra %[b2], %[b2], 15 \n\t"
+ "addu %[value32], %[value32], %[b2] \n\t"
+ : [value32] "=&r"(value32), [b1] "=&r"(b1), [b2] "=&r"(b2), [a1] "=&r"(a1)
+ : [a] "r"(a), [b] "r"(b)
+ : "hi", "lo");
+ return value32;
+}
+
+#if defined(MIPS_DSP_R1_LE)
+static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
+ __asm __volatile(
+ "shll_s.w %[value32], %[value32], 16 \n\t"
+ "sra %[value32], %[value32], 16 \n\t"
+ : [value32] "+r"(value32)
+ :);
+ int16_t out16 = (int16_t)value32;
+ return out16;
+}
+
+static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
+ int32_t value32 = 0;
+
+ __asm __volatile("addq_s.ph %[value32], %[a], %[b] \n\t"
+ : [value32] "=r"(value32)
+ : [a] "r"(a), [b] "r"(b));
+ return (int16_t)value32;
+}
+
+static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
+ int32_t l_sum;
+
+ __asm __volatile(
+ "addq_s.w %[l_sum], %[l_var1], %[l_var2] \n\t"
+ : [l_sum] "=r"(l_sum)
+ : [l_var1] "r"(l_var1), [l_var2] "r"(l_var2));
+
+ return l_sum;
+}
+
+static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
+ int32_t value32;
+
+ __asm __volatile("subq_s.ph %[value32], %[var1], %[var2] \n\t"
+ : [value32] "=r"(value32)
+ : [var1] "r"(var1), [var2] "r"(var2));
+
+ return (int16_t)value32;
+}
+
+static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
+ int32_t l_diff;
+
+ __asm __volatile(
+ "subq_s.w %[l_diff], %[l_var1], %[l_var2] \n\t"
+ : [l_diff] "=r"(l_diff)
+ : [l_var1] "r"(l_var1), [l_var2] "r"(l_var2));
+
+ return l_diff;
+}
+#endif
+
+static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
+ int bits = 0;
+ int i32 = 32;
+
+ __asm __volatile(
+ "clz %[bits], %[n] \n\t"
+ "subu %[bits], %[i32], %[bits] \n\t"
+ : [bits] "=&r"(bits)
+ : [n] "r"(n), [i32] "r"(i32));
+
+ return (int16_t)bits;
+}
+
+static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
+ int zeros = 0;
+
+ __asm __volatile(
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "bnez %[a], 1f \n\t"
+ " sra %[zeros], %[a], 31 \n\t"
+ "b 2f \n\t"
+ " move %[zeros], $zero \n\t"
+ "1: \n\t"
+ "xor %[zeros], %[a], %[zeros] \n\t"
+ "clz %[zeros], %[zeros] \n\t"
+ "addiu %[zeros], %[zeros], -1 \n\t"
+ "2: \n\t"
+ ".set pop \n\t"
+ : [zeros] "=&r"(zeros)
+ : [a] "r"(a));
+
+ return (int16_t)zeros;
+}
+
+static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
+ int zeros = 0;
+
+ __asm __volatile("clz %[zeros], %[a] \n\t"
+ : [zeros] "=r"(zeros)
+ : [a] "r"(a));
+
+ return (int16_t)(zeros & 0x1f);
+}
+
+static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
+ int zeros = 0;
+ int a0 = a << 16;
+
+ __asm __volatile(
+ ".set push \n\t"
+ ".set noreorder \n\t"
+ "bnez %[a0], 1f \n\t"
+ " sra %[zeros], %[a0], 31 \n\t"
+ "b 2f \n\t"
+ " move %[zeros], $zero \n\t"
+ "1: \n\t"
+ "xor %[zeros], %[a0], %[zeros] \n\t"
+ "clz %[zeros], %[zeros] \n\t"
+ "addiu %[zeros], %[zeros], -1 \n\t"
+ "2: \n\t"
+ ".set pop \n\t"
+ : [zeros] "=&r"(zeros)
+ : [a0] "r"(a0));
+
+ return (int16_t)zeros;
+}
+
+static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
+ int32_t res = 0, c1 = 0;
+ __asm __volatile(
+#if defined(MIPS32_R2_LE)
+ "seh %[a], %[a] \n\t"
+ "seh %[b], %[b] \n\t"
+#else
+ "sll %[a], %[a], 16 \n\t"
+ "sll %[b], %[b], 16 \n\t"
+ "sra %[a], %[a], 16 \n\t"
+ "sra %[b], %[b], 16 \n\t"
+#endif
+ "mul %[res], %[a], %[b] \n\t"
+ "addu %[c1], %[c], %[res] \n\t"
+ : [c1] "=r"(c1), [res] "=&r"(res)
+ : [a] "r"(a), [b] "r"(b), [c] "r"(c)
+ : "hi", "lo");
+ return (c1);
+}
+
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/levinson_durbin.c b/third_party/libwebrtc/common_audio/signal_processing/levinson_durbin.c
new file mode 100644
index 0000000000..2c5cbaeeaa
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/levinson_durbin.c
@@ -0,0 +1,249 @@
+/*
+ * 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_LevinsonDurbin().
+ * 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"
+
+#define SPL_LEVINSON_MAXORDER 20
+
+int16_t RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
+WebRtcSpl_LevinsonDurbin(const int32_t* R, int16_t* A, int16_t* K,
+ size_t order)
+{
+ size_t i, j;
+ // Auto-correlation coefficients in high precision
+ int16_t R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
+ // LPC coefficients in high precision
+ int16_t A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
+ // LPC coefficients for next iteration
+ int16_t A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
+ // Reflection coefficient in high precision
+ int16_t K_hi, K_low;
+ // Prediction gain Alpha in high precision and with scale factor
+ int16_t Alpha_hi, Alpha_low, Alpha_exp;
+ int16_t tmp_hi, tmp_low;
+ int32_t temp1W32, temp2W32, temp3W32;
+ int16_t norm;
+
+ // Normalize the autocorrelation R[0]...R[order+1]
+
+ norm = WebRtcSpl_NormW32(R[0]);
+
+ for (i = 0; i <= order; ++i)
+ {
+ temp1W32 = R[i] * (1 << norm);
+ // UBSan: 12 * 268435456 cannot be represented in type 'int'
+
+ // Put R in hi and low format
+ R_hi[i] = (int16_t)(temp1W32 >> 16);
+ R_low[i] = (int16_t)((temp1W32 - ((int32_t)R_hi[i] * 65536)) >> 1);
+ }
+
+ // K = A[1] = -R[1] / R[0]
+
+ temp2W32 = R[1] * (1 << norm); // R[1] in Q31
+ temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
+ temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
+ // Put back the sign on R[1]
+ if (temp2W32 > 0)
+ {
+ temp1W32 = -temp1W32;
+ }
+
+ // Put K in hi and low format
+ K_hi = (int16_t)(temp1W32 >> 16);
+ K_low = (int16_t)((temp1W32 - ((int32_t)K_hi * 65536)) >> 1);
+
+ // Store first reflection coefficient
+ K[0] = K_hi;
+
+ temp1W32 >>= 4; // A[1] in Q27.
+
+ // Put A[1] in hi and low format
+ A_hi[1] = (int16_t)(temp1W32 >> 16);
+ A_low[1] = (int16_t)((temp1W32 - ((int32_t)A_hi[1] * 65536)) >> 1);
+
+ // Alpha = R[0] * (1-K^2)
+
+ temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) * 2; // = k^2 in Q31
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+ temp1W32 = (int32_t)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
+
+ // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
+ tmp_hi = (int16_t)(temp1W32 >> 16);
+ tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
+
+ // Calculate Alpha in Q31
+ temp1W32 = (R_hi[0] * tmp_hi + (R_hi[0] * tmp_low >> 15) +
+ (R_low[0] * tmp_hi >> 15)) << 1;
+
+ // Normalize Alpha and put it in hi and low format
+
+ Alpha_exp = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
+ Alpha_hi = (int16_t)(temp1W32 >> 16);
+ Alpha_low = (int16_t)((temp1W32 - ((int32_t)Alpha_hi << 16)) >> 1);
+
+ // Perform the iterative calculations in the Levinson-Durbin algorithm
+
+ for (i = 2; i <= order; i++)
+ {
+ /* ----
+ temp1W32 = R[i] + > R[j]*A[i-j]
+ /
+ ----
+ j=1..i-1
+ */
+
+ temp1W32 = 0;
+
+ for (j = 1; j < i; j++)
+ {
+ // temp1W32 is in Q31
+ temp1W32 += (R_hi[j] * A_hi[i - j] * 2) +
+ (((R_hi[j] * A_low[i - j] >> 15) +
+ (R_low[j] * A_hi[i - j] >> 15)) * 2);
+ }
+
+ temp1W32 = temp1W32 * 16;
+ temp1W32 += ((int32_t)R_hi[i] * 65536)
+ + WEBRTC_SPL_LSHIFT_W32((int32_t)R_low[i], 1);
+
+ // K = -temp1W32 / Alpha
+ temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
+ temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
+
+ // Put the sign of temp1W32 back again
+ if (temp1W32 > 0)
+ {
+ temp3W32 = -temp3W32;
+ }
+
+ // Use the Alpha shifts from earlier to de-normalize
+ norm = WebRtcSpl_NormW32(temp3W32);
+ if ((Alpha_exp <= norm) || (temp3W32 == 0))
+ {
+ temp3W32 = temp3W32 * (1 << Alpha_exp);
+ } else
+ {
+ if (temp3W32 > 0)
+ {
+ temp3W32 = (int32_t)0x7fffffffL;
+ } else
+ {
+ temp3W32 = (int32_t)0x80000000L;
+ }
+ }
+
+ // Put K on hi and low format
+ K_hi = (int16_t)(temp3W32 >> 16);
+ K_low = (int16_t)((temp3W32 - ((int32_t)K_hi * 65536)) >> 1);
+
+ // Store Reflection coefficient in Q15
+ K[i - 1] = K_hi;
+
+ // Test for unstable filter.
+ // If unstable return 0 and let the user decide what to do in that case
+
+ if ((int32_t)WEBRTC_SPL_ABS_W16(K_hi) > (int32_t)32750)
+ {
+ return 0; // Unstable filter
+ }
+
+ /*
+ Compute updated LPC coefficient: Anew[i]
+ Anew[j]= A[j] + K*A[i-j] for j=1..i-1
+ Anew[i]= K
+ */
+
+ for (j = 1; j < i; j++)
+ {
+ // temp1W32 = A[j] in Q27
+ temp1W32 = (int32_t)A_hi[j] * 65536
+ + WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[j],1);
+
+ // temp1W32 += K*A[i-j] in Q27
+ temp1W32 += (K_hi * A_hi[i - j] + (K_hi * A_low[i - j] >> 15) +
+ (K_low * A_hi[i - j] >> 15)) * 2;
+
+ // Put Anew in hi and low format
+ A_upd_hi[j] = (int16_t)(temp1W32 >> 16);
+ A_upd_low[j] = (int16_t)(
+ (temp1W32 - ((int32_t)A_upd_hi[j] * 65536)) >> 1);
+ }
+
+ // temp3W32 = K in Q27 (Convert from Q31 to Q27)
+ temp3W32 >>= 4;
+
+ // Store Anew in hi and low format
+ A_upd_hi[i] = (int16_t)(temp3W32 >> 16);
+ A_upd_low[i] = (int16_t)(
+ (temp3W32 - ((int32_t)A_upd_hi[i] * 65536)) >> 1);
+
+ // Alpha = Alpha * (1-K^2)
+
+ temp1W32 = ((K_hi * K_low >> 14) + K_hi * K_hi) * 2; // K*K in Q31
+
+ temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
+ temp1W32 = (int32_t)0x7fffffffL - temp1W32; // 1 - K*K in Q31
+
+ // Convert 1- K^2 in hi and low format
+ tmp_hi = (int16_t)(temp1W32 >> 16);
+ tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1);
+
+ // Calculate Alpha = Alpha * (1-K^2) in Q31
+ temp1W32 = (Alpha_hi * tmp_hi + (Alpha_hi * tmp_low >> 15) +
+ (Alpha_low * tmp_hi >> 15)) << 1;
+
+ // Normalize Alpha and store it on hi and low format
+
+ norm = WebRtcSpl_NormW32(temp1W32);
+ temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
+
+ Alpha_hi = (int16_t)(temp1W32 >> 16);
+ Alpha_low = (int16_t)((temp1W32 - ((int32_t)Alpha_hi << 16)) >> 1);
+
+ // Update the total normalization of Alpha
+ Alpha_exp = Alpha_exp + norm;
+
+ // Update A[]
+
+ for (j = 1; j <= i; j++)
+ {
+ A_hi[j] = A_upd_hi[j];
+ A_low[j] = A_upd_low[j];
+ }
+ }
+
+ /*
+ Set A[0] to 1.0 and store the A[i] i=1...order in Q12
+ (Convert from Q27 and use rounding)
+ */
+
+ A[0] = 4096;
+
+ for (i = 1; i <= order; i++)
+ {
+ // temp1W32 in Q27
+ temp1W32 = (int32_t)A_hi[i] * 65536
+ + WEBRTC_SPL_LSHIFT_W32((int32_t)A_low[i], 1);
+ // Round and store upper word
+ A[i] = (int16_t)(((temp1W32 * 2) + 32768) >> 16);
+ }
+ return 1; // Stable filters
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/lpc_to_refl_coef.c b/third_party/libwebrtc/common_audio/signal_processing/lpc_to_refl_coef.c
new file mode 100644
index 0000000000..7a5e25191b
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/lpc_to_refl_coef.c
@@ -0,0 +1,56 @@
+/*
+ * 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_LpcToReflCoef().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+#define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50
+
+void WebRtcSpl_LpcToReflCoef(int16_t* a16, int use_order, int16_t* k16)
+{
+ int m, k;
+ int32_t tmp32[SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER];
+ int32_t tmp_inv_denom32;
+ int16_t tmp_inv_denom16;
+
+ k16[use_order - 1] = a16[use_order] << 3; // Q12<<3 => Q15
+ for (m = use_order - 1; m > 0; m--)
+ {
+ // (1 - k^2) in Q30
+ tmp_inv_denom32 = 1073741823 - k16[m] * k16[m];
+ // (1 - k^2) in Q15
+ tmp_inv_denom16 = (int16_t)(tmp_inv_denom32 >> 15);
+
+ for (k = 1; k <= m; k++)
+ {
+ // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
+
+ // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
+ tmp32[k] = (a16[k] << 16) - (k16[m] * a16[m - k + 1] << 1);
+
+ tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
+ }
+
+ for (k = 1; k < m; k++)
+ {
+ a16[k] = (int16_t)(tmp32[k] >> 1); // Q13>>1 => Q12
+ }
+
+ tmp32[m] = WEBRTC_SPL_SAT(8191, tmp32[m], -8191);
+ k16[m - 1] = (int16_t)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
+ }
+ return;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/min_max_operations.c b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations.c
new file mode 100644
index 0000000000..1b9542e7ef
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations.c
@@ -0,0 +1,256 @@
+/*
+ * 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 "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 = abs((int)vector[i]);
+ 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
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_mips.c b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_mips.c
new file mode 100644
index 0000000000..8a7fc65c42
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_mips.c
@@ -0,0 +1,375 @@
+/*
+ * 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.
+ */
+
+/*
+ * This file contains the implementation of function
+ * WebRtcSpl_MaxAbsValueW16()
+ *
+ * The description header can be found in signal_processing_library.h.
+ *
+ */
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Maximum absolute value of word16 vector.
+int16_t WebRtcSpl_MaxAbsValueW16_mips(const int16_t* vector, size_t length) {
+ int32_t totMax = 0;
+ int32_t tmp32_0, tmp32_1, tmp32_2, tmp32_3;
+ size_t i, loop_size;
+
+ RTC_DCHECK_GT(length, 0);
+
+#if defined(MIPS_DSP_R1)
+ const int32_t* tmpvec32 = (int32_t*)vector;
+ loop_size = length >> 4;
+
+ for (i = 0; i < loop_size; i++) {
+ __asm__ volatile (
+ "lw %[tmp32_0], 0(%[tmpvec32]) \n\t"
+ "lw %[tmp32_1], 4(%[tmpvec32]) \n\t"
+ "lw %[tmp32_2], 8(%[tmpvec32]) \n\t"
+ "lw %[tmp32_3], 12(%[tmpvec32]) \n\t"
+
+ "absq_s.ph %[tmp32_0], %[tmp32_0] \n\t"
+ "absq_s.ph %[tmp32_1], %[tmp32_1] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_0] \n\t"
+ "pick.ph %[totMax], %[tmp32_0], %[totMax] \n\t"
+
+ "lw %[tmp32_0], 16(%[tmpvec32]) \n\t"
+ "absq_s.ph %[tmp32_2], %[tmp32_2] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_1] \n\t"
+ "pick.ph %[totMax], %[tmp32_1], %[totMax] \n\t"
+
+ "lw %[tmp32_1], 20(%[tmpvec32]) \n\t"
+ "absq_s.ph %[tmp32_3], %[tmp32_3] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_2] \n\t"
+ "pick.ph %[totMax], %[tmp32_2], %[totMax] \n\t"
+
+ "lw %[tmp32_2], 24(%[tmpvec32]) \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_3] \n\t"
+ "pick.ph %[totMax], %[tmp32_3], %[totMax] \n\t"
+
+ "lw %[tmp32_3], 28(%[tmpvec32]) \n\t"
+ "absq_s.ph %[tmp32_0], %[tmp32_0] \n\t"
+ "absq_s.ph %[tmp32_1], %[tmp32_1] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_0] \n\t"
+ "pick.ph %[totMax], %[tmp32_0], %[totMax] \n\t"
+
+ "absq_s.ph %[tmp32_2], %[tmp32_2] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_1] \n\t"
+ "pick.ph %[totMax], %[tmp32_1], %[totMax] \n\t"
+ "absq_s.ph %[tmp32_3], %[tmp32_3] \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_2] \n\t"
+ "pick.ph %[totMax], %[tmp32_2], %[totMax] \n\t"
+
+ "cmp.lt.ph %[totMax], %[tmp32_3] \n\t"
+ "pick.ph %[totMax], %[tmp32_3], %[totMax] \n\t"
+
+ "addiu %[tmpvec32], %[tmpvec32], 32 \n\t"
+ : [tmp32_0] "=&r" (tmp32_0), [tmp32_1] "=&r" (tmp32_1),
+ [tmp32_2] "=&r" (tmp32_2), [tmp32_3] "=&r" (tmp32_3),
+ [totMax] "+r" (totMax), [tmpvec32] "+r" (tmpvec32)
+ :
+ : "memory"
+ );
+ }
+ __asm__ volatile (
+ "rotr %[tmp32_0], %[totMax], 16 \n\t"
+ "cmp.lt.ph %[totMax], %[tmp32_0] \n\t"
+ "pick.ph %[totMax], %[tmp32_0], %[totMax] \n\t"
+ "packrl.ph %[totMax], $0, %[totMax] \n\t"
+ : [tmp32_0] "=&r" (tmp32_0), [totMax] "+r" (totMax)
+ :
+ );
+ loop_size = length & 0xf;
+ for (i = 0; i < loop_size; i++) {
+ __asm__ volatile (
+ "lh %[tmp32_0], 0(%[tmpvec32]) \n\t"
+ "addiu %[tmpvec32], %[tmpvec32], 2 \n\t"
+ "absq_s.w %[tmp32_0], %[tmp32_0] \n\t"
+ "slt %[tmp32_1], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[tmp32_1] \n\t"
+ : [tmp32_0] "=&r" (tmp32_0), [tmp32_1] "=&r" (tmp32_1),
+ [tmpvec32] "+r" (tmpvec32), [totMax] "+r" (totMax)
+ :
+ : "memory"
+ );
+ }
+#else // #if defined(MIPS_DSP_R1)
+ int32_t v16MaxMax = WEBRTC_SPL_WORD16_MAX;
+ int32_t r, r1, r2, r3;
+ const int16_t* tmpvector = vector;
+ loop_size = length >> 4;
+ for (i = 0; i < loop_size; i++) {
+ __asm__ volatile (
+ "lh %[tmp32_0], 0(%[tmpvector]) \n\t"
+ "lh %[tmp32_1], 2(%[tmpvector]) \n\t"
+ "lh %[tmp32_2], 4(%[tmpvector]) \n\t"
+ "lh %[tmp32_3], 6(%[tmpvector]) \n\t"
+
+ "abs %[tmp32_0], %[tmp32_0] \n\t"
+ "abs %[tmp32_1], %[tmp32_1] \n\t"
+ "abs %[tmp32_2], %[tmp32_2] \n\t"
+ "abs %[tmp32_3], %[tmp32_3] \n\t"
+
+ "slt %[r], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[r] \n\t"
+ "slt %[r1], %[totMax], %[tmp32_1] \n\t"
+ "movn %[totMax], %[tmp32_1], %[r1] \n\t"
+ "slt %[r2], %[totMax], %[tmp32_2] \n\t"
+ "movn %[totMax], %[tmp32_2], %[r2] \n\t"
+ "slt %[r3], %[totMax], %[tmp32_3] \n\t"
+ "movn %[totMax], %[tmp32_3], %[r3] \n\t"
+
+ "lh %[tmp32_0], 8(%[tmpvector]) \n\t"
+ "lh %[tmp32_1], 10(%[tmpvector]) \n\t"
+ "lh %[tmp32_2], 12(%[tmpvector]) \n\t"
+ "lh %[tmp32_3], 14(%[tmpvector]) \n\t"
+
+ "abs %[tmp32_0], %[tmp32_0] \n\t"
+ "abs %[tmp32_1], %[tmp32_1] \n\t"
+ "abs %[tmp32_2], %[tmp32_2] \n\t"
+ "abs %[tmp32_3], %[tmp32_3] \n\t"
+
+ "slt %[r], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[r] \n\t"
+ "slt %[r1], %[totMax], %[tmp32_1] \n\t"
+ "movn %[totMax], %[tmp32_1], %[r1] \n\t"
+ "slt %[r2], %[totMax], %[tmp32_2] \n\t"
+ "movn %[totMax], %[tmp32_2], %[r2] \n\t"
+ "slt %[r3], %[totMax], %[tmp32_3] \n\t"
+ "movn %[totMax], %[tmp32_3], %[r3] \n\t"
+
+ "lh %[tmp32_0], 16(%[tmpvector]) \n\t"
+ "lh %[tmp32_1], 18(%[tmpvector]) \n\t"
+ "lh %[tmp32_2], 20(%[tmpvector]) \n\t"
+ "lh %[tmp32_3], 22(%[tmpvector]) \n\t"
+
+ "abs %[tmp32_0], %[tmp32_0] \n\t"
+ "abs %[tmp32_1], %[tmp32_1] \n\t"
+ "abs %[tmp32_2], %[tmp32_2] \n\t"
+ "abs %[tmp32_3], %[tmp32_3] \n\t"
+
+ "slt %[r], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[r] \n\t"
+ "slt %[r1], %[totMax], %[tmp32_1] \n\t"
+ "movn %[totMax], %[tmp32_1], %[r1] \n\t"
+ "slt %[r2], %[totMax], %[tmp32_2] \n\t"
+ "movn %[totMax], %[tmp32_2], %[r2] \n\t"
+ "slt %[r3], %[totMax], %[tmp32_3] \n\t"
+ "movn %[totMax], %[tmp32_3], %[r3] \n\t"
+
+ "lh %[tmp32_0], 24(%[tmpvector]) \n\t"
+ "lh %[tmp32_1], 26(%[tmpvector]) \n\t"
+ "lh %[tmp32_2], 28(%[tmpvector]) \n\t"
+ "lh %[tmp32_3], 30(%[tmpvector]) \n\t"
+
+ "abs %[tmp32_0], %[tmp32_0] \n\t"
+ "abs %[tmp32_1], %[tmp32_1] \n\t"
+ "abs %[tmp32_2], %[tmp32_2] \n\t"
+ "abs %[tmp32_3], %[tmp32_3] \n\t"
+
+ "slt %[r], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[r] \n\t"
+ "slt %[r1], %[totMax], %[tmp32_1] \n\t"
+ "movn %[totMax], %[tmp32_1], %[r1] \n\t"
+ "slt %[r2], %[totMax], %[tmp32_2] \n\t"
+ "movn %[totMax], %[tmp32_2], %[r2] \n\t"
+ "slt %[r3], %[totMax], %[tmp32_3] \n\t"
+ "movn %[totMax], %[tmp32_3], %[r3] \n\t"
+
+ "addiu %[tmpvector], %[tmpvector], 32 \n\t"
+ : [tmp32_0] "=&r" (tmp32_0), [tmp32_1] "=&r" (tmp32_1),
+ [tmp32_2] "=&r" (tmp32_2), [tmp32_3] "=&r" (tmp32_3),
+ [totMax] "+r" (totMax), [r] "=&r" (r), [tmpvector] "+r" (tmpvector),
+ [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3)
+ :
+ : "memory"
+ );
+ }
+ loop_size = length & 0xf;
+ for (i = 0; i < loop_size; i++) {
+ __asm__ volatile (
+ "lh %[tmp32_0], 0(%[tmpvector]) \n\t"
+ "addiu %[tmpvector], %[tmpvector], 2 \n\t"
+ "abs %[tmp32_0], %[tmp32_0] \n\t"
+ "slt %[tmp32_1], %[totMax], %[tmp32_0] \n\t"
+ "movn %[totMax], %[tmp32_0], %[tmp32_1] \n\t"
+ : [tmp32_0] "=&r" (tmp32_0), [tmp32_1] "=&r" (tmp32_1),
+ [tmpvector] "+r" (tmpvector), [totMax] "+r" (totMax)
+ :
+ : "memory"
+ );
+ }
+
+ __asm__ volatile (
+ "slt %[r], %[v16MaxMax], %[totMax] \n\t"
+ "movn %[totMax], %[v16MaxMax], %[r] \n\t"
+ : [totMax] "+r" (totMax), [r] "=&r" (r)
+ : [v16MaxMax] "r" (v16MaxMax)
+ );
+#endif // #if defined(MIPS_DSP_R1)
+ return (int16_t)totMax;
+}
+
+#if defined(MIPS_DSP_R1_LE)
+// Maximum absolute value of word32 vector. Version for MIPS platform.
+int32_t WebRtcSpl_MaxAbsValueW32_mips(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;
+ int tmp1 = 0, max_value = 0x7fffffff;
+
+ RTC_DCHECK_GT(length, 0);
+
+ __asm__ volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "1: \n\t"
+ "lw %[absolute], 0(%[vector]) \n\t"
+ "absq_s.w %[absolute], %[absolute] \n\t"
+ "addiu %[length], %[length], -1 \n\t"
+ "slt %[tmp1], %[maximum], %[absolute] \n\t"
+ "movn %[maximum], %[absolute], %[tmp1] \n\t"
+ "bgtz %[length], 1b \n\t"
+ " addiu %[vector], %[vector], 4 \n\t"
+ "slt %[tmp1], %[max_value], %[maximum] \n\t"
+ "movn %[maximum], %[max_value], %[tmp1] \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [maximum] "+r" (maximum), [absolute] "+r" (absolute)
+ : [vector] "r" (vector), [length] "r" (length), [max_value] "r" (max_value)
+ : "memory"
+ );
+
+ return (int32_t)maximum;
+}
+#endif // #if defined(MIPS_DSP_R1_LE)
+
+// Maximum value of word16 vector. Version for MIPS platform.
+int16_t WebRtcSpl_MaxValueW16_mips(const int16_t* vector, size_t length) {
+ int16_t maximum = WEBRTC_SPL_WORD16_MIN;
+ int tmp1;
+ int16_t value;
+
+ RTC_DCHECK_GT(length, 0);
+
+ __asm__ volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "1: \n\t"
+ "lh %[value], 0(%[vector]) \n\t"
+ "addiu %[length], %[length], -1 \n\t"
+ "slt %[tmp1], %[maximum], %[value] \n\t"
+ "movn %[maximum], %[value], %[tmp1] \n\t"
+ "bgtz %[length], 1b \n\t"
+ " addiu %[vector], %[vector], 2 \n\t"
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [maximum] "+r" (maximum), [value] "=&r" (value)
+ : [vector] "r" (vector), [length] "r" (length)
+ : "memory"
+ );
+
+ return maximum;
+}
+
+// Maximum value of word32 vector. Version for MIPS platform.
+int32_t WebRtcSpl_MaxValueW32_mips(const int32_t* vector, size_t length) {
+ int32_t maximum = WEBRTC_SPL_WORD32_MIN;
+ int tmp1, value;
+
+ RTC_DCHECK_GT(length, 0);
+
+ __asm__ volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "1: \n\t"
+ "lw %[value], 0(%[vector]) \n\t"
+ "addiu %[length], %[length], -1 \n\t"
+ "slt %[tmp1], %[maximum], %[value] \n\t"
+ "movn %[maximum], %[value], %[tmp1] \n\t"
+ "bgtz %[length], 1b \n\t"
+ " addiu %[vector], %[vector], 4 \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [maximum] "+r" (maximum), [value] "=&r" (value)
+ : [vector] "r" (vector), [length] "r" (length)
+ : "memory"
+ );
+
+ return maximum;
+}
+
+// Minimum value of word16 vector. Version for MIPS platform.
+int16_t WebRtcSpl_MinValueW16_mips(const int16_t* vector, size_t length) {
+ int16_t minimum = WEBRTC_SPL_WORD16_MAX;
+ int tmp1;
+ int16_t value;
+
+ RTC_DCHECK_GT(length, 0);
+
+ __asm__ volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "1: \n\t"
+ "lh %[value], 0(%[vector]) \n\t"
+ "addiu %[length], %[length], -1 \n\t"
+ "slt %[tmp1], %[value], %[minimum] \n\t"
+ "movn %[minimum], %[value], %[tmp1] \n\t"
+ "bgtz %[length], 1b \n\t"
+ " addiu %[vector], %[vector], 2 \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [minimum] "+r" (minimum), [value] "=&r" (value)
+ : [vector] "r" (vector), [length] "r" (length)
+ : "memory"
+ );
+
+ return minimum;
+}
+
+// Minimum value of word32 vector. Version for MIPS platform.
+int32_t WebRtcSpl_MinValueW32_mips(const int32_t* vector, size_t length) {
+ int32_t minimum = WEBRTC_SPL_WORD32_MAX;
+ int tmp1, value;
+
+ RTC_DCHECK_GT(length, 0);
+
+ __asm__ volatile (
+ ".set push \n\t"
+ ".set noreorder \n\t"
+
+ "1: \n\t"
+ "lw %[value], 0(%[vector]) \n\t"
+ "addiu %[length], %[length], -1 \n\t"
+ "slt %[tmp1], %[value], %[minimum] \n\t"
+ "movn %[minimum], %[value], %[tmp1] \n\t"
+ "bgtz %[length], 1b \n\t"
+ " addiu %[vector], %[vector], 4 \n\t"
+
+ ".set pop \n\t"
+
+ : [tmp1] "=&r" (tmp1), [minimum] "+r" (minimum), [value] "=&r" (value)
+ : [vector] "r" (vector), [length] "r" (length)
+ : "memory"
+ );
+
+ return minimum;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_neon.c b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_neon.c
new file mode 100644
index 0000000000..e5b4b7c71b
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/min_max_operations_neon.c
@@ -0,0 +1,333 @@
+/*
+ * Copyright (c) 2014 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 <stdlib.h>
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Maximum absolute value of word16 vector. C version for generic platforms.
+int16_t WebRtcSpl_MaxAbsValueW16Neon(const int16_t* vector, size_t length) {
+ int absolute = 0, maximum = 0;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int16_t* p_start = vector;
+ size_t rest = length & 7;
+ const int16_t* p_end = vector + length - rest;
+
+ int16x8_t v;
+ uint16x8_t max_qv;
+ max_qv = vdupq_n_u16(0);
+
+ while (p_start < p_end) {
+ v = vld1q_s16(p_start);
+ // Note vabs doesn't change the value of -32768.
+ v = vabsq_s16(v);
+ // Use u16 so we don't lose the value -32768.
+ max_qv = vmaxq_u16(max_qv, vreinterpretq_u16_s16(v));
+ p_start += 8;
+ }
+
+#ifdef WEBRTC_ARCH_ARM64
+ maximum = (int)vmaxvq_u16(max_qv);
+#else
+ uint16x4_t max_dv;
+ max_dv = vmax_u16(vget_low_u16(max_qv), vget_high_u16(max_qv));
+ max_dv = vpmax_u16(max_dv, max_dv);
+ max_dv = vpmax_u16(max_dv, max_dv);
+
+ maximum = (int)vget_lane_u16(max_dv, 0);
+#endif
+
+ p_end = vector + length;
+ while (p_start < p_end) {
+ absolute = abs((int)(*p_start));
+
+ if (absolute > maximum) {
+ maximum = absolute;
+ }
+ p_start++;
+ }
+
+ // 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. NEON intrinsics version for
+// ARM 32-bit/64-bit platforms.
+int32_t WebRtcSpl_MaxAbsValueW32Neon(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;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int32_t* p_start = vector;
+ uint32x4_t max32x4_0 = vdupq_n_u32(0);
+ uint32x4_t max32x4_1 = vdupq_n_u32(0);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int32x4_t in32x4_0 = vld1q_s32(p_start);
+ p_start += 4;
+ int32x4_t in32x4_1 = vld1q_s32(p_start);
+ p_start += 4;
+ in32x4_0 = vabsq_s32(in32x4_0);
+ in32x4_1 = vabsq_s32(in32x4_1);
+ // vabs doesn't change the value of 0x80000000.
+ // Use u32 so we don't lose the value 0x80000000.
+ max32x4_0 = vmaxq_u32(max32x4_0, vreinterpretq_u32_s32(in32x4_0));
+ max32x4_1 = vmaxq_u32(max32x4_1, vreinterpretq_u32_s32(in32x4_1));
+ }
+
+ uint32x4_t max32x4 = vmaxq_u32(max32x4_0, max32x4_1);
+#if defined(WEBRTC_ARCH_ARM64)
+ maximum = vmaxvq_u32(max32x4);
+#else
+ uint32x2_t max32x2 = vmax_u32(vget_low_u32(max32x4), vget_high_u32(max32x4));
+ max32x2 = vpmax_u32(max32x2, max32x2);
+
+ maximum = vget_lane_u32(max32x2, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ absolute = abs((int)(*p_start));
+ if (absolute > maximum) {
+ maximum = absolute;
+ }
+ p_start++;
+ }
+
+ // Guard against the case for 0x80000000.
+ maximum = WEBRTC_SPL_MIN(maximum, WEBRTC_SPL_WORD32_MAX);
+
+ return (int32_t)maximum;
+}
+
+// Maximum value of word16 vector. NEON intrinsics version for
+// ARM 32-bit/64-bit platforms.
+int16_t WebRtcSpl_MaxValueW16Neon(const int16_t* vector, size_t length) {
+ int16_t maximum = WEBRTC_SPL_WORD16_MIN;
+ size_t i = 0;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int16_t* p_start = vector;
+ int16x8_t max16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MIN);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int16x8_t in16x8 = vld1q_s16(p_start);
+ max16x8 = vmaxq_s16(max16x8, in16x8);
+ p_start += 8;
+ }
+
+#if defined(WEBRTC_ARCH_ARM64)
+ maximum = vmaxvq_s16(max16x8);
+#else
+ int16x4_t max16x4 = vmax_s16(vget_low_s16(max16x8), vget_high_s16(max16x8));
+ max16x4 = vpmax_s16(max16x4, max16x4);
+ max16x4 = vpmax_s16(max16x4, max16x4);
+
+ maximum = vget_lane_s16(max16x4, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ if (*p_start > maximum)
+ maximum = *p_start;
+ p_start++;
+ }
+ return maximum;
+}
+
+// Maximum value of word32 vector. NEON intrinsics version for
+// ARM 32-bit/64-bit platforms.
+int32_t WebRtcSpl_MaxValueW32Neon(const int32_t* vector, size_t length) {
+ int32_t maximum = WEBRTC_SPL_WORD32_MIN;
+ size_t i = 0;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int32_t* p_start = vector;
+ int32x4_t max32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MIN);
+ int32x4_t max32x4_1 = vdupq_n_s32(WEBRTC_SPL_WORD32_MIN);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int32x4_t in32x4_0 = vld1q_s32(p_start);
+ p_start += 4;
+ int32x4_t in32x4_1 = vld1q_s32(p_start);
+ p_start += 4;
+ max32x4_0 = vmaxq_s32(max32x4_0, in32x4_0);
+ max32x4_1 = vmaxq_s32(max32x4_1, in32x4_1);
+ }
+
+ int32x4_t max32x4 = vmaxq_s32(max32x4_0, max32x4_1);
+#if defined(WEBRTC_ARCH_ARM64)
+ maximum = vmaxvq_s32(max32x4);
+#else
+ int32x2_t max32x2 = vmax_s32(vget_low_s32(max32x4), vget_high_s32(max32x4));
+ max32x2 = vpmax_s32(max32x2, max32x2);
+
+ maximum = vget_lane_s32(max32x2, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ if (*p_start > maximum)
+ maximum = *p_start;
+ p_start++;
+ }
+ return maximum;
+}
+
+// Minimum value of word16 vector. NEON intrinsics version for
+// ARM 32-bit/64-bit platforms.
+int16_t WebRtcSpl_MinValueW16Neon(const int16_t* vector, size_t length) {
+ int16_t minimum = WEBRTC_SPL_WORD16_MAX;
+ size_t i = 0;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int16_t* p_start = vector;
+ int16x8_t min16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MAX);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int16x8_t in16x8 = vld1q_s16(p_start);
+ min16x8 = vminq_s16(min16x8, in16x8);
+ p_start += 8;
+ }
+
+#if defined(WEBRTC_ARCH_ARM64)
+ minimum = vminvq_s16(min16x8);
+#else
+ int16x4_t min16x4 = vmin_s16(vget_low_s16(min16x8), vget_high_s16(min16x8));
+ min16x4 = vpmin_s16(min16x4, min16x4);
+ min16x4 = vpmin_s16(min16x4, min16x4);
+
+ minimum = vget_lane_s16(min16x4, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ if (*p_start < minimum)
+ minimum = *p_start;
+ p_start++;
+ }
+ return minimum;
+}
+
+// Minimum value of word32 vector. NEON intrinsics version for
+// ARM 32-bit/64-bit platforms.
+int32_t WebRtcSpl_MinValueW32Neon(const int32_t* vector, size_t length) {
+ int32_t minimum = WEBRTC_SPL_WORD32_MAX;
+ size_t i = 0;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int32_t* p_start = vector;
+ int32x4_t min32x4_0 = vdupq_n_s32(WEBRTC_SPL_WORD32_MAX);
+ int32x4_t min32x4_1 = vdupq_n_s32(WEBRTC_SPL_WORD32_MAX);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int32x4_t in32x4_0 = vld1q_s32(p_start);
+ p_start += 4;
+ int32x4_t in32x4_1 = vld1q_s32(p_start);
+ p_start += 4;
+ min32x4_0 = vminq_s32(min32x4_0, in32x4_0);
+ min32x4_1 = vminq_s32(min32x4_1, in32x4_1);
+ }
+
+ int32x4_t min32x4 = vminq_s32(min32x4_0, min32x4_1);
+#if defined(WEBRTC_ARCH_ARM64)
+ minimum = vminvq_s32(min32x4);
+#else
+ int32x2_t min32x2 = vmin_s32(vget_low_s32(min32x4), vget_high_s32(min32x4));
+ min32x2 = vpmin_s32(min32x2, min32x2);
+
+ minimum = vget_lane_s32(min32x2, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ if (*p_start < minimum)
+ minimum = *p_start;
+ p_start++;
+ }
+ return minimum;
+}
+
+// Finds both the minimum and maximum elements in an array of 16-bit integers.
+void WebRtcSpl_MinMaxW16Neon(const int16_t* vector, size_t length,
+ int16_t* min_val, int16_t* max_val) {
+ int16_t minimum = WEBRTC_SPL_WORD16_MAX;
+ int16_t maximum = WEBRTC_SPL_WORD16_MIN;
+ size_t i = 0;
+ size_t residual = length & 0x7;
+
+ RTC_DCHECK_GT(length, 0);
+
+ const int16_t* p_start = vector;
+ int16x8_t min16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MAX);
+ int16x8_t max16x8 = vdupq_n_s16(WEBRTC_SPL_WORD16_MIN);
+
+ // First part, unroll the loop 8 times.
+ for (i = 0; i < length - residual; i += 8) {
+ int16x8_t in16x8 = vld1q_s16(p_start);
+ min16x8 = vminq_s16(min16x8, in16x8);
+ max16x8 = vmaxq_s16(max16x8, in16x8);
+ p_start += 8;
+ }
+
+#if defined(WEBRTC_ARCH_ARM64)
+ minimum = vminvq_s16(min16x8);
+ maximum = vmaxvq_s16(max16x8);
+#else
+ int16x4_t min16x4 = vmin_s16(vget_low_s16(min16x8), vget_high_s16(min16x8));
+ min16x4 = vpmin_s16(min16x4, min16x4);
+ min16x4 = vpmin_s16(min16x4, min16x4);
+
+ minimum = vget_lane_s16(min16x4, 0);
+
+ int16x4_t max16x4 = vmax_s16(vget_low_s16(max16x8), vget_high_s16(max16x8));
+ max16x4 = vpmax_s16(max16x4, max16x4);
+ max16x4 = vpmax_s16(max16x4, max16x4);
+
+ maximum = vget_lane_s16(max16x4, 0);
+#endif
+
+ // Second part, do the remaining iterations (if any).
+ for (i = residual; i > 0; i--) {
+ if (*p_start < minimum)
+ minimum = *p_start;
+ if (*p_start > maximum)
+ maximum = *p_start;
+ p_start++;
+ }
+ *min_val = minimum;
+ *max_val = maximum;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/randomization_functions.c b/third_party/libwebrtc/common_audio/signal_processing/randomization_functions.c
new file mode 100644
index 0000000000..a445c572c7
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/randomization_functions.c
@@ -0,0 +1,115 @@
+/*
+ * 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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/real_fft.c b/third_party/libwebrtc/common_audio/signal_processing/real_fft.c
new file mode 100644
index 0000000000..780e517a15
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/real_fft.c
@@ -0,0 +1,102 @@
+/*
+ * 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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/real_fft_unittest.cc b/third_party/libwebrtc/common_audio/signal_processing/real_fft_unittest.cc
new file mode 100644
index 0000000000..7cabe7d9fe
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/real_fft_unittest.cc
@@ -0,0 +1,98 @@
+/*
+ * 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 "common_audio/signal_processing/include/signal_processing_library.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+// FFT order.
+const int kOrder = 5;
+// Lengths for real FFT's time and frequency bufffers.
+// For N-point FFT, the length requirements from API are N and N+2 respectively.
+const int kTimeDataLength = 1 << kOrder;
+const int kFreqDataLength = (1 << kOrder) + 2;
+// For complex FFT's time and freq buffer. The implementation requires
+// 2*N 16-bit words.
+const int kComplexFftDataLength = 2 << kOrder;
+// Reference data for time signal.
+const int16_t kRefData[kTimeDataLength] = {
+ 11739, 6848, -8688, 31980, -30295, 25242, 27085, 19410,
+ -26299, 15607, -10791, 11778, -23819, 14498, -25772, 10076,
+ 1173, 6848, -8688, 31980, -30295, 2522, 27085, 19410,
+ -2629, 5607, -3, 1178, -23819, 1498, -25772, 10076};
+
+TEST(RealFFTTest, CreateFailsOnBadInput) {
+ RealFFT* fft = WebRtcSpl_CreateRealFFT(11);
+ EXPECT_TRUE(fft == nullptr);
+ fft = WebRtcSpl_CreateRealFFT(-1);
+ EXPECT_TRUE(fft == nullptr);
+}
+
+TEST(RealFFTTest, RealAndComplexMatch) {
+ int i = 0;
+ int j = 0;
+ int16_t real_fft_time[kTimeDataLength] = {0};
+ int16_t real_fft_freq[kFreqDataLength] = {0};
+ // One common buffer for complex FFT's time and frequency data.
+ int16_t complex_fft_buff[kComplexFftDataLength] = {0};
+
+ // Prepare the inputs to forward FFT's.
+ memcpy(real_fft_time, kRefData, sizeof(kRefData));
+ for (i = 0, j = 0; i < kTimeDataLength; i += 1, j += 2) {
+ complex_fft_buff[j] = kRefData[i];
+ complex_fft_buff[j + 1] = 0; // Insert zero's to imaginary parts.
+ }
+
+ // Create and run real forward FFT.
+ RealFFT* fft = WebRtcSpl_CreateRealFFT(kOrder);
+ EXPECT_TRUE(fft != nullptr);
+ EXPECT_EQ(0, WebRtcSpl_RealForwardFFT(fft, real_fft_time, real_fft_freq));
+
+ // Run complex forward FFT.
+ WebRtcSpl_ComplexBitReverse(complex_fft_buff, kOrder);
+ EXPECT_EQ(0, WebRtcSpl_ComplexFFT(complex_fft_buff, kOrder, 1));
+
+ // Verify the results between complex and real forward FFT.
+ for (i = 0; i < kFreqDataLength; i++) {
+ EXPECT_EQ(real_fft_freq[i], complex_fft_buff[i]);
+ }
+
+ // Prepare the inputs to inverse real FFT.
+ // We use whatever data in complex_fft_buff[] since we don't care
+ // about data contents. Only kFreqDataLength 16-bit words are copied
+ // from complex_fft_buff to real_fft_freq since remaining words (2nd half)
+ // are conjugate-symmetric to the first half in theory.
+ memcpy(real_fft_freq, complex_fft_buff, sizeof(real_fft_freq));
+
+ // Run real inverse FFT.
+ int real_scale = WebRtcSpl_RealInverseFFT(fft, real_fft_freq, real_fft_time);
+ EXPECT_GE(real_scale, 0);
+
+ // Run complex inverse FFT.
+ WebRtcSpl_ComplexBitReverse(complex_fft_buff, kOrder);
+ int complex_scale = WebRtcSpl_ComplexIFFT(complex_fft_buff, kOrder, 1);
+
+ // Verify the results between complex and real inverse FFT.
+ // They are not bit-exact, since complex IFFT doesn't produce
+ // exactly conjugate-symmetric data (between first and second half).
+ EXPECT_EQ(real_scale, complex_scale);
+ for (i = 0, j = 0; i < kTimeDataLength; i += 1, j += 2) {
+ EXPECT_LE(abs(real_fft_time[i] - complex_fft_buff[j]), 1);
+ }
+
+ WebRtcSpl_FreeRealFFT(fft);
+}
+
+} // namespace
+} // namespace webrtc
diff --git a/third_party/libwebrtc/common_audio/signal_processing/refl_coef_to_lpc.c b/third_party/libwebrtc/common_audio/signal_processing/refl_coef_to_lpc.c
new file mode 100644
index 0000000000..b0858b2b0e
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/refl_coef_to_lpc.c
@@ -0,0 +1,59 @@
+/*
+ * 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_ReflCoefToLpc().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a)
+{
+ int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+ int16_t *aptr, *aptr2, *anyptr;
+ const int16_t *kptr;
+ int m, i;
+
+ kptr = k;
+ *a = 4096; // i.e., (Word16_MAX >> 3)+1.
+ *any = *a;
+ a[1] = *k >> 3;
+
+ for (m = 1; m < use_order; m++)
+ {
+ kptr++;
+ aptr = a;
+ aptr++;
+ aptr2 = &a[m];
+ anyptr = any;
+ anyptr++;
+
+ any[m + 1] = *kptr >> 3;
+ for (i = 0; i < m; i++)
+ {
+ *anyptr = *aptr + (int16_t)((*aptr2 * *kptr) >> 15);
+ anyptr++;
+ aptr++;
+ aptr2--;
+ }
+
+ aptr = a;
+ anyptr = any;
+ for (i = 0; i < (m + 2); i++)
+ {
+ *aptr = *anyptr;
+ aptr++;
+ anyptr++;
+ }
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample.c b/third_party/libwebrtc/common_audio/signal_processing/resample.c
new file mode 100644
index 0000000000..d4b2736476
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample.c
@@ -0,0 +1,505 @@
+/*
+ * 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 resampling functions for 22 kHz.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/resample_by_2_internal.h"
+
+// Declaration of internally used functions
+static void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In, int16_t *Out,
+ int32_t K);
+
+void WebRtcSpl_32khzTo22khzIntToInt(const int32_t *In, int32_t *Out,
+ int32_t K);
+
+// interpolation coefficients
+static const int16_t kCoefficients32To22[5][9] = {
+ {127, -712, 2359, -6333, 23456, 16775, -3695, 945, -154},
+ {-39, 230, -830, 2785, 32366, -2324, 760, -218, 38},
+ {117, -663, 2222, -6133, 26634, 13070, -3174, 831, -137},
+ {-77, 457, -1677, 5958, 31175, -4136, 1405, -408, 71},
+ { 98, -560, 1900, -5406, 29240, 9423, -2480, 663, -110}
+};
+
+//////////////////////
+// 22 kHz -> 16 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_22_16 5
+
+// 22 -> 16 resampler
+void WebRtcSpl_Resample22khzTo16khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State22khzTo16khz* state, int32_t* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_22_16; k++)
+ {
+ ///// 22 --> 44 /////
+ // int16_t in[220/SUB_BLOCKS_22_16]
+ // int32_t out[440/SUB_BLOCKS_22_16]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
+
+ ///// 44 --> 32 /////
+ // int32_t in[440/SUB_BLOCKS_22_16]
+ // int32_t out[320/SUB_BLOCKS_22_16]
+ /////
+ // copy state to and from input array
+ tmpmem[8] = state->S_44_32[0];
+ tmpmem[9] = state->S_44_32[1];
+ tmpmem[10] = state->S_44_32[2];
+ tmpmem[11] = state->S_44_32[3];
+ tmpmem[12] = state->S_44_32[4];
+ tmpmem[13] = state->S_44_32[5];
+ tmpmem[14] = state->S_44_32[6];
+ tmpmem[15] = state->S_44_32[7];
+ state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
+ state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
+ state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
+ state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
+ state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
+ state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
+ state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
+ state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
+
+ WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
+
+ ///// 32 --> 16 /////
+ // int32_t in[320/SUB_BLOCKS_22_16]
+ // int32_t out[160/SUB_BLOCKS_22_16]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
+
+ // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
+ in += 220 / SUB_BLOCKS_22_16;
+ out += 160 / SUB_BLOCKS_22_16;
+ }
+}
+
+// initialize state of 22 -> 16 resampler
+void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_22_44[k] = 0;
+ state->S_44_32[k] = 0;
+ state->S_32_16[k] = 0;
+ }
+}
+
+//////////////////////
+// 16 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 4, 5, 10
+#define SUB_BLOCKS_16_22 4
+
+// 16 -> 22 resampler
+void WebRtcSpl_Resample16khzTo22khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State16khzTo22khz* state, int32_t* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_16_22; k++)
+ {
+ ///// 16 --> 32 /////
+ // int16_t in[160/SUB_BLOCKS_16_22]
+ // int32_t out[320/SUB_BLOCKS_16_22]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
+
+ ///// 32 --> 22 /////
+ // int32_t in[320/SUB_BLOCKS_16_22]
+ // int32_t out[220/SUB_BLOCKS_16_22]
+ /////
+ // copy state to and from input array
+ tmpmem[0] = state->S_32_22[0];
+ tmpmem[1] = state->S_32_22[1];
+ tmpmem[2] = state->S_32_22[2];
+ tmpmem[3] = state->S_32_22[3];
+ tmpmem[4] = state->S_32_22[4];
+ tmpmem[5] = state->S_32_22[5];
+ tmpmem[6] = state->S_32_22[6];
+ tmpmem[7] = state->S_32_22[7];
+ state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
+ state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
+ state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
+ state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
+ state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
+ state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
+ state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
+ state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
+
+ WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
+
+ // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
+ in += 160 / SUB_BLOCKS_16_22;
+ out += 220 / SUB_BLOCKS_16_22;
+ }
+}
+
+// initialize state of 16 -> 22 resampler
+void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_16_32[k] = 0;
+ state->S_32_22[k] = 0;
+ }
+}
+
+//////////////////////
+// 22 kHz -> 8 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_22_8 2
+
+// 22 -> 8 resampler
+void WebRtcSpl_Resample22khzTo8khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State22khzTo8khz* state, int32_t* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_22_8; k++)
+ {
+ ///// 22 --> 22 lowpass /////
+ // int16_t in[220/SUB_BLOCKS_22_8]
+ // int32_t out[220/SUB_BLOCKS_22_8]
+ /////
+ WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
+
+ ///// 22 --> 16 /////
+ // int32_t in[220/SUB_BLOCKS_22_8]
+ // int32_t out[160/SUB_BLOCKS_22_8]
+ /////
+ // copy state to and from input array
+ tmpmem[8] = state->S_22_16[0];
+ tmpmem[9] = state->S_22_16[1];
+ tmpmem[10] = state->S_22_16[2];
+ tmpmem[11] = state->S_22_16[3];
+ tmpmem[12] = state->S_22_16[4];
+ tmpmem[13] = state->S_22_16[5];
+ tmpmem[14] = state->S_22_16[6];
+ tmpmem[15] = state->S_22_16[7];
+ state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
+ state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
+ state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
+ state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
+ state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
+ state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
+ state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
+ state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
+
+ WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
+
+ ///// 16 --> 8 /////
+ // int32_t in[160/SUB_BLOCKS_22_8]
+ // int32_t out[80/SUB_BLOCKS_22_8]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
+
+ // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
+ in += 220 / SUB_BLOCKS_22_8;
+ out += 80 / SUB_BLOCKS_22_8;
+ }
+}
+
+// initialize state of 22 -> 8 resampler
+void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_22_22[k] = 0;
+ state->S_22_22[k + 8] = 0;
+ state->S_22_16[k] = 0;
+ state->S_16_8[k] = 0;
+ }
+}
+
+//////////////////////
+// 8 kHz -> 22 kHz //
+//////////////////////
+
+// number of subblocks; options: 1, 2, 5, 10
+#define SUB_BLOCKS_8_22 2
+
+// 8 -> 22 resampler
+void WebRtcSpl_Resample8khzTo22khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State8khzTo22khz* state, int32_t* tmpmem)
+{
+ int k;
+
+ // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
+ for (k = 0; k < SUB_BLOCKS_8_22; k++)
+ {
+ ///// 8 --> 16 /////
+ // int16_t in[80/SUB_BLOCKS_8_22]
+ // int32_t out[160/SUB_BLOCKS_8_22]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
+
+ ///// 16 --> 11 /////
+ // int32_t in[160/SUB_BLOCKS_8_22]
+ // int32_t out[110/SUB_BLOCKS_8_22]
+ /////
+ // copy state to and from input array
+ tmpmem[10] = state->S_16_11[0];
+ tmpmem[11] = state->S_16_11[1];
+ tmpmem[12] = state->S_16_11[2];
+ tmpmem[13] = state->S_16_11[3];
+ tmpmem[14] = state->S_16_11[4];
+ tmpmem[15] = state->S_16_11[5];
+ tmpmem[16] = state->S_16_11[6];
+ tmpmem[17] = state->S_16_11[7];
+ state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
+ state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
+ state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
+ state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
+ state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
+ state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
+ state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
+ state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
+
+ WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
+
+ ///// 11 --> 22 /////
+ // int32_t in[110/SUB_BLOCKS_8_22]
+ // int16_t out[220/SUB_BLOCKS_8_22]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
+
+ // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
+ in += 80 / SUB_BLOCKS_8_22;
+ out += 220 / SUB_BLOCKS_8_22;
+ }
+}
+
+// initialize state of 8 -> 22 resampler
+void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
+{
+ int k;
+ for (k = 0; k < 8; k++)
+ {
+ state->S_8_16[k] = 0;
+ state->S_16_11[k] = 0;
+ state->S_11_22[k] = 0;
+ }
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToInt(const int32_t* in1, const int32_t* in2,
+ const int16_t* coef_ptr, int32_t* out1,
+ int32_t* out2)
+{
+ int32_t tmp1 = 16384;
+ int32_t tmp2 = 16384;
+ int16_t coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ *out1 = tmp1 + coef * in1[8];
+ *out2 = tmp2 + coef * in2[-8];
+}
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_DotProdIntToShort(const int32_t* in1, const int32_t* in2,
+ const int16_t* coef_ptr, int16_t* out1,
+ int16_t* out2)
+{
+ int32_t tmp1 = 16384;
+ int32_t tmp2 = 16384;
+ int16_t coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ tmp1 += coef * in1[8];
+ tmp2 += coef * in2[-8];
+
+ // scale down, round and saturate
+ tmp1 >>= 15;
+ if (tmp1 > (int32_t)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (int32_t)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ tmp2 >>= 15;
+ if (tmp2 > (int32_t)0x00007FFF)
+ tmp2 = 0x00007FFF;
+ if (tmp2 < (int32_t)0xFFFF8000)
+ tmp2 = 0xFFFF8000;
+ *out1 = (int16_t)tmp1;
+ *out2 = (int16_t)tmp2;
+}
+
+// Resampling ratio: 11/16
+// input: int32_t (normalized, not saturated) :: size 16 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 11 * K
+// K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToInt(const int32_t* In,
+ int32_t* Out,
+ int32_t K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (16 input samples -> 11 output samples);
+ // process in sub blocks of size 16 samples.
+ int32_t m;
+
+ for (m = 0; m < K; m++)
+ {
+ // first output sample
+ Out[0] = ((int32_t)In[3] << 15) + (1 << 14);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+ // update pointers
+ In += 16;
+ Out += 11;
+ }
+}
+
+// Resampling ratio: 11/16
+// input: int32_t (normalized, not saturated) :: size 16 * K
+// output: int16_t (saturated) :: size 11 * K
+// K: Number of blocks
+
+void WebRtcSpl_32khzTo22khzIntToShort(const int32_t *In,
+ int16_t *Out,
+ int32_t K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (16 input samples -> 11 output samples);
+ // process in sub blocks of size 16 samples.
+ int32_t tmp;
+ int32_t m;
+
+ for (m = 0; m < K; m++)
+ {
+ // first output sample
+ tmp = In[3];
+ if (tmp > (int32_t)0x00007FFF)
+ tmp = 0x00007FFF;
+ if (tmp < (int32_t)0xFFFF8000)
+ tmp = 0xFFFF8000;
+ Out[0] = (int16_t)tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
+
+ // update pointers
+ In += 16;
+ Out += 11;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_48khz.c b/third_party/libwebrtc/common_audio/signal_processing/resample_48khz.c
new file mode 100644
index 0000000000..8518e7b1ce
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_48khz.c
@@ -0,0 +1,186 @@
+/*
+ * 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 resampling functions between 48 kHz and nb/wb.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include <string.h>
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/resample_by_2_internal.h"
+
+////////////////////////////
+///// 48 kHz -> 16 kHz /////
+////////////////////////////
+
+// 48 -> 16 resampler
+void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State48khzTo16khz* state, int32_t* tmpmem)
+{
+ ///// 48 --> 48(LP) /////
+ // int16_t in[480]
+ // int32_t out[480]
+ /////
+ WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
+
+ ///// 48 --> 32 /////
+ // int32_t in[480]
+ // int32_t out[320]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(int32_t));
+ memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(int32_t));
+ WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
+
+ ///// 32 --> 16 /////
+ // int32_t in[320]
+ // int16_t out[160]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
+}
+
+// initialize state of 48 -> 16 resampler
+void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
+{
+ memset(state->S_48_48, 0, 16 * sizeof(int32_t));
+ memset(state->S_48_32, 0, 8 * sizeof(int32_t));
+ memset(state->S_32_16, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+///// 16 kHz -> 48 kHz /////
+////////////////////////////
+
+// 16 -> 48 resampler
+void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State16khzTo48khz* state, int32_t* tmpmem)
+{
+ ///// 16 --> 32 /////
+ // int16_t in[160]
+ // int32_t out[320]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
+
+ ///// 32 --> 24 /////
+ // int32_t in[320]
+ // int32_t out[240]
+ // copy state to and from input array
+ /////
+ memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(int32_t));
+ memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(int32_t));
+ WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
+
+ ///// 24 --> 48 /////
+ // int32_t in[240]
+ // int16_t out[480]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 16 -> 48 resampler
+void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
+{
+ memset(state->S_16_32, 0, 8 * sizeof(int32_t));
+ memset(state->S_32_24, 0, 8 * sizeof(int32_t));
+ memset(state->S_24_48, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+///// 48 kHz -> 8 kHz /////
+////////////////////////////
+
+// 48 -> 8 resampler
+void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State48khzTo8khz* state, int32_t* tmpmem)
+{
+ ///// 48 --> 24 /////
+ // int16_t in[480]
+ // int32_t out[240]
+ /////
+ WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
+
+ ///// 24 --> 24(LP) /////
+ // int32_t in[240]
+ // int32_t out[240]
+ /////
+ WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
+
+ ///// 24 --> 16 /////
+ // int32_t in[240]
+ // int32_t out[160]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(int32_t));
+ memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(int32_t));
+ WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
+
+ ///// 16 --> 8 /////
+ // int32_t in[160]
+ // int16_t out[80]
+ /////
+ WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
+}
+
+// initialize state of 48 -> 8 resampler
+void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
+{
+ memset(state->S_48_24, 0, 8 * sizeof(int32_t));
+ memset(state->S_24_24, 0, 16 * sizeof(int32_t));
+ memset(state->S_24_16, 0, 8 * sizeof(int32_t));
+ memset(state->S_16_8, 0, 8 * sizeof(int32_t));
+}
+
+////////////////////////////
+///// 8 kHz -> 48 kHz /////
+////////////////////////////
+
+// 8 -> 48 resampler
+void WebRtcSpl_Resample8khzTo48khz(const int16_t* in, int16_t* out,
+ WebRtcSpl_State8khzTo48khz* state, int32_t* tmpmem)
+{
+ ///// 8 --> 16 /////
+ // int16_t in[80]
+ // int32_t out[160]
+ /////
+ WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
+
+ ///// 16 --> 12 /////
+ // int32_t in[160]
+ // int32_t out[120]
+ /////
+ // copy state to and from input array
+ memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(int32_t));
+ memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(int32_t));
+ WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
+
+ ///// 12 --> 24 /////
+ // int32_t in[120]
+ // int16_t out[240]
+ /////
+ WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
+
+ ///// 24 --> 48 /////
+ // int32_t in[240]
+ // int16_t out[480]
+ /////
+ WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
+}
+
+// initialize state of 8 -> 48 resampler
+void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
+{
+ memset(state->S_8_16, 0, 8 * sizeof(int32_t));
+ memset(state->S_16_12, 0, 8 * sizeof(int32_t));
+ memset(state->S_12_24, 0, 8 * sizeof(int32_t));
+ memset(state->S_24_48, 0, 8 * sizeof(int32_t));
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_by_2.c b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2.c
new file mode 100644
index 0000000000..73e1950654
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2.c
@@ -0,0 +1,183 @@
+/*
+ * 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 resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+#ifdef WEBRTC_ARCH_ARM_V7
+
+// allpass filter coefficients.
+static const uint32_t kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
+static const uint32_t kResampleAllpass2[3] =
+ {12199, 37471 << 15, 60255 << 15};
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: state + ((diff * tbl_value) >> 16)
+
+static __inline int32_t MUL_ACCUM_1(int32_t tbl_value,
+ int32_t diff,
+ int32_t state) {
+ int32_t result;
+ __asm __volatile ("smlawb %0, %1, %2, %3": "=r"(result): "r"(diff),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+// Multiply two 32-bit values and accumulate to another input value.
+// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
+//
+// The reason to introduce this function is that, in case we can't use smlawb
+// instruction (in MUL_ACCUM_1) due to input value range, we can still use
+// smmla to save some cycles.
+
+static __inline int32_t MUL_ACCUM_2(int32_t tbl_value,
+ int32_t diff,
+ int32_t state) {
+ int32_t result;
+ __asm __volatile ("smmla %0, %1, %2, %3": "=r"(result): "r"(diff << 1),
+ "r"(tbl_value), "r"(state));
+ return result;
+}
+
+#else
+
+// allpass filter coefficients.
+static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528};
+static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+#endif // WEBRTC_ARCH_ARM_V7
+
+
+// decimator
+#if !defined(MIPS32_LE)
+void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
+ int16_t* out, int32_t* filtState) {
+ int32_t tmp1, tmp2, diff, in32, out32;
+ size_t i;
+
+ register int32_t state0 = filtState[0];
+ register int32_t state1 = filtState[1];
+ register int32_t state2 = filtState[2];
+ register int32_t state3 = filtState[3];
+ register int32_t state4 = filtState[4];
+ register int32_t state5 = filtState[5];
+ register int32_t state6 = filtState[6];
+ register int32_t state7 = filtState[7];
+
+ for (i = (len >> 1); i > 0; i--) {
+ // lower allpass filter
+ in32 = (int32_t)(*in++) * (1 << 10);
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (int32_t)(*in++) * (1 << 10);
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
+#endif // #if defined(MIPS32_LE)
+
+
+void WebRtcSpl_UpsampleBy2(const int16_t* in, size_t len,
+ int16_t* out, int32_t* filtState) {
+ int32_t tmp1, tmp2, diff, in32, out32;
+ size_t i;
+
+ register int32_t state0 = filtState[0];
+ register int32_t state1 = filtState[1];
+ register int32_t state2 = filtState[2];
+ register int32_t state3 = filtState[3];
+ register int32_t state4 = filtState[4];
+ register int32_t state5 = filtState[5];
+ register int32_t state6 = filtState[6];
+ register int32_t state7 = filtState[7];
+
+ for (i = len; i > 0; i--) {
+ // lower allpass filter
+ in32 = (int32_t)(*in++) * (1 << 10);
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
+ state2 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state3 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+
+ // upper allpass filter
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
+ state6 = tmp2;
+
+ // round; limit amplitude to prevent wrap-around; write to output array
+ out32 = (state7 + 512) >> 10;
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+
+ filtState[0] = state0;
+ filtState[1] = state1;
+ filtState[2] = state2;
+ filtState[3] = state3;
+ filtState[4] = state4;
+ filtState[5] = state5;
+ filtState[6] = state6;
+ filtState[7] = state7;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.c b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.c
new file mode 100644
index 0000000000..99592b20b5
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.c
@@ -0,0 +1,689 @@
+/*
+ * 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 header file contains some internal resampling functions.
+ *
+ */
+
+#include "common_audio/signal_processing/resample_by_2_internal.h"
+#include "rtc_base/sanitizer.h"
+
+// allpass filter coefficients.
+static const int16_t kResampleAllpass[2][3] = {
+ {821, 6110, 12382},
+ {3050, 9368, 15063}
+};
+
+//
+// decimator
+// input: int32_t (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
+// output: int16_t (saturated) (of length len/2)
+// state: filter state array; length = 8
+
+void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
+WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
+ int32_t *state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ len >>= 1;
+
+ // lower allpass filter (operates on even input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[1];
+ // UBSan: -1771017321 - 999586185 cannot be represented in type 'int'
+
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // divide by two and store temporarily
+ in[i << 1] = (state[3] >> 1);
+ }
+
+ in++;
+
+ // upper allpass filter (operates on odd input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // divide by two and store temporarily
+ in[i << 1] = (state[7] >> 1);
+ }
+
+ in--;
+
+ // combine allpass outputs
+ for (i = 0; i < len; i += 2)
+ {
+ // divide by two, add both allpass outputs and round
+ tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
+ tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
+ if (tmp0 > (int32_t)0x00007FFF)
+ tmp0 = 0x00007FFF;
+ if (tmp0 < (int32_t)0xFFFF8000)
+ tmp0 = 0xFFFF8000;
+ out[i] = (int16_t)tmp0;
+ if (tmp1 > (int32_t)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (int32_t)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i + 1] = (int16_t)tmp1;
+ }
+}
+
+//
+// decimator
+// input: int16_t
+// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len/2)
+// state: filter state array; length = 8
+
+void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
+WebRtcSpl_DownBy2ShortToInt(const int16_t *in,
+ int32_t len,
+ int32_t *out,
+ int32_t *state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ len >>= 1;
+
+ // lower allpass filter (operates on even input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // UBSan: -1379909682 - 834099714 cannot be represented in type 'int'
+
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // divide by two and store temporarily
+ out[i] = (state[3] >> 1);
+ }
+
+ in++;
+
+ // upper allpass filter (operates on odd input samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // divide by two and store temporarily
+ out[i] += (state[7] >> 1);
+ }
+
+ in--;
+}
+
+//
+// interpolator
+// input: int16_t
+// output: int32_t (normalized, not saturated) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len, int32_t *out,
+ int32_t *state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[7] >> 15;
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i] << 15) + (1 << 14);
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 15;
+ }
+}
+
+//
+// interpolator
+// input: int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int32_t (shifted 15 positions to the left, + offset 16384) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2IntToInt(const int32_t *in, int32_t len, int32_t *out,
+ int32_t *state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[7];
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3];
+ }
+}
+
+//
+// interpolator
+// input: int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int16_t (saturated) (of length len*2)
+// state: filter state array; length = 8
+void WebRtcSpl_UpBy2IntToShort(const int32_t *in, int32_t len, int16_t *out,
+ int32_t *state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ // upper allpass filter (generates odd output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // scale down, saturate and store
+ tmp1 = state[7] >> 15;
+ if (tmp1 > (int32_t)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (int32_t)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i << 1] = (int16_t)tmp1;
+ }
+
+ out++;
+
+ // lower allpass filter (generates even output samples)
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i];
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, saturate and store
+ tmp1 = state[3] >> 15;
+ if (tmp1 > (int32_t)0x00007FFF)
+ tmp1 = 0x00007FFF;
+ if (tmp1 < (int32_t)0xFFFF8000)
+ tmp1 = 0xFFFF8000;
+ out[i << 1] = (int16_t)tmp1;
+ }
+}
+
+// lowpass filter
+// input: int16_t
+// output: int32_t (normalized, not saturated)
+// state: filter state array; length = 8
+void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len, int32_t* out,
+ int32_t* state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ len >>= 1;
+
+ // lower allpass filter: odd input -> even output samples
+ in++;
+ // initial state of polyphase delay element
+ tmp0 = state[12];
+ for (i = 0; i < len; i++)
+ {
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 1;
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ }
+ in--;
+
+ // upper allpass filter: even input -> even output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[5];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+ }
+
+ // switch to odd output samples
+ out++;
+
+ // lower allpass filter: even input -> odd output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[9];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[8] + diff * kResampleAllpass[1][0];
+ state[8] = tmp0;
+ diff = tmp1 - state[10];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[9] + diff * kResampleAllpass[1][1];
+ state[9] = tmp1;
+ diff = tmp0 - state[11];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[11] = state[10] + diff * kResampleAllpass[1][2];
+ state[10] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[11] >> 1;
+ }
+
+ // upper allpass filter: odd input -> odd output samples
+ in++;
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = ((int32_t)in[i << 1] << 15) + (1 << 14);
+ diff = tmp0 - state[13];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[12] + diff * kResampleAllpass[0][0];
+ state[12] = tmp0;
+ diff = tmp1 - state[14];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[13] + diff * kResampleAllpass[0][1];
+ state[13] = tmp1;
+ diff = tmp0 - state[15];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[15] = state[14] + diff * kResampleAllpass[0][2];
+ state[14] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+ }
+}
+
+// lowpass filter
+// input: int32_t (shifted 15 positions to the left, + offset 16384)
+// output: int32_t (normalized, not saturated)
+// state: filter state array; length = 8
+void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5486
+WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
+ int32_t* state)
+{
+ int32_t tmp0, tmp1, diff;
+ int32_t i;
+
+ len >>= 1;
+
+ // lower allpass filter: odd input -> even output samples
+ in++;
+ // initial state of polyphase delay element
+ tmp0 = state[12];
+ for (i = 0; i < len; i++)
+ {
+ diff = tmp0 - state[1];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[0] + diff * kResampleAllpass[1][0];
+ state[0] = tmp0;
+ diff = tmp1 - state[2];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[1] + diff * kResampleAllpass[1][1];
+ state[1] = tmp1;
+ diff = tmp0 - state[3];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[3] = state[2] + diff * kResampleAllpass[1][2];
+ state[2] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[3] >> 1;
+ tmp0 = in[i << 1];
+ }
+ in--;
+
+ // upper allpass filter: even input -> even output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[5];
+ // UBSan: -794814117 - 1566149201 cannot be represented in type 'int'
+
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[4] + diff * kResampleAllpass[0][0];
+ state[4] = tmp0;
+ diff = tmp1 - state[6];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[5] + diff * kResampleAllpass[0][1];
+ state[5] = tmp1;
+ diff = tmp0 - state[7];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[7] = state[6] + diff * kResampleAllpass[0][2];
+ state[6] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
+ }
+
+ // switch to odd output samples
+ out++;
+
+ // lower allpass filter: even input -> odd output samples
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[9];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[8] + diff * kResampleAllpass[1][0];
+ state[8] = tmp0;
+ diff = tmp1 - state[10];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[9] + diff * kResampleAllpass[1][1];
+ state[9] = tmp1;
+ diff = tmp0 - state[11];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[11] = state[10] + diff * kResampleAllpass[1][2];
+ state[10] = tmp0;
+
+ // scale down, round and store
+ out[i << 1] = state[11] >> 1;
+ }
+
+ // upper allpass filter: odd input -> odd output samples
+ in++;
+ for (i = 0; i < len; i++)
+ {
+ tmp0 = in[i << 1];
+ diff = tmp0 - state[13];
+ // scale down and round
+ diff = (diff + (1 << 13)) >> 14;
+ tmp1 = state[12] + diff * kResampleAllpass[0][0];
+ state[12] = tmp0;
+ diff = tmp1 - state[14];
+ // scale down and round
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ tmp0 = state[13] + diff * kResampleAllpass[0][1];
+ state[13] = tmp1;
+ diff = tmp0 - state[15];
+ // scale down and truncate
+ diff = diff >> 14;
+ if (diff < 0)
+ diff += 1;
+ state[15] = state[14] + diff * kResampleAllpass[0][2];
+ state[14] = tmp0;
+
+ // average the two allpass outputs, scale down and store
+ out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.h b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.h
new file mode 100644
index 0000000000..145395a4cb
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_internal.h
@@ -0,0 +1,60 @@
+/*
+ * 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 header file contains some internal resampling functions.
+ *
+ */
+
+#ifndef COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
+#define COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
+
+#include <stdint.h>
+
+/*******************************************************************
+ * resample_by_2_fast.c
+ * Functions for internal use in the other resample functions
+ ******************************************************************/
+void WebRtcSpl_DownBy2IntToShort(int32_t* in,
+ int32_t len,
+ int16_t* out,
+ int32_t* state);
+
+void WebRtcSpl_DownBy2ShortToInt(const int16_t* in,
+ int32_t len,
+ int32_t* out,
+ int32_t* state);
+
+void WebRtcSpl_UpBy2ShortToInt(const int16_t* in,
+ int32_t len,
+ int32_t* out,
+ int32_t* state);
+
+void WebRtcSpl_UpBy2IntToInt(const int32_t* in,
+ int32_t len,
+ int32_t* out,
+ int32_t* state);
+
+void WebRtcSpl_UpBy2IntToShort(const int32_t* in,
+ int32_t len,
+ int16_t* out,
+ int32_t* state);
+
+void WebRtcSpl_LPBy2ShortToInt(const int16_t* in,
+ int32_t len,
+ int32_t* out,
+ int32_t* state);
+
+void WebRtcSpl_LPBy2IntToInt(const int32_t* in,
+ int32_t len,
+ int32_t* out,
+ int32_t* state);
+
+#endif // COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_mips.c b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_mips.c
new file mode 100644
index 0000000000..f41bab7519
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_by_2_mips.c
@@ -0,0 +1,292 @@
+/*
+ * 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.
+ */
+
+
+/*
+ * This file contains the resampling by two functions.
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#if defined(MIPS32_LE)
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+#if !defined(MIPS_DSP_R2_LE)
+// allpass filter coefficients.
+static const uint16_t kResampleAllpass1[3] = {3284, 24441, 49528};
+static const uint16_t kResampleAllpass2[3] = {12199, 37471, 60255};
+#endif
+
+// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
+#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
+
+// decimator
+void WebRtcSpl_DownsampleBy2(const int16_t* in,
+ size_t len,
+ int16_t* out,
+ int32_t* filtState) {
+ int32_t out32;
+ size_t i, len1;
+
+ register int32_t state0 = filtState[0];
+ register int32_t state1 = filtState[1];
+ register int32_t state2 = filtState[2];
+ register int32_t state3 = filtState[3];
+ register int32_t state4 = filtState[4];
+ register int32_t state5 = filtState[5];
+ register int32_t state6 = filtState[6];
+ register int32_t state7 = filtState[7];
+
+#if defined(MIPS_DSP_R2_LE)
+ int32_t k1Res0, k1Res1, k1Res2, k2Res0, k2Res1, k2Res2;
+
+ k1Res0= 3284;
+ k1Res1= 24441;
+ k1Res2= 49528;
+ k2Res0= 12199;
+ k2Res1= 37471;
+ k2Res2= 60255;
+ len1 = (len >> 1);
+
+ const int32_t* inw = (int32_t*)in;
+ int32_t tmp11, tmp12, tmp21, tmp22;
+ int32_t in322, in321;
+ int32_t diff1, diff2;
+ for (i = len1; i > 0; i--) {
+ __asm__ volatile (
+ "lh %[in321], 0(%[inw]) \n\t"
+ "lh %[in322], 2(%[inw]) \n\t"
+
+ "sll %[in321], %[in321], 10 \n\t"
+ "sll %[in322], %[in322], 10 \n\t"
+
+ "addiu %[inw], %[inw], 4 \n\t"
+
+ "subu %[diff1], %[in321], %[state1] \n\t"
+ "subu %[diff2], %[in322], %[state5] \n\t"
+
+ : [in322] "=&r" (in322), [in321] "=&r" (in321),
+ [diff1] "=&r" (diff1), [diff2] "=r" (diff2), [inw] "+r" (inw)
+ : [state1] "r" (state1), [state5] "r" (state5)
+ : "memory"
+ );
+
+ __asm__ volatile (
+ "mult $ac0, %[diff1], %[k2Res0] \n\t"
+ "mult $ac1, %[diff2], %[k1Res0] \n\t"
+
+ "extr.w %[tmp11], $ac0, 16 \n\t"
+ "extr.w %[tmp12], $ac1, 16 \n\t"
+
+ "addu %[tmp11], %[state0], %[tmp11] \n\t"
+ "addu %[tmp12], %[state4], %[tmp12] \n\t"
+
+ "addiu %[state0], %[in321], 0 \n\t"
+ "addiu %[state4], %[in322], 0 \n\t"
+
+ "subu %[diff1], %[tmp11], %[state2] \n\t"
+ "subu %[diff2], %[tmp12], %[state6] \n\t"
+
+ "mult $ac0, %[diff1], %[k2Res1] \n\t"
+ "mult $ac1, %[diff2], %[k1Res1] \n\t"
+
+ "extr.w %[tmp21], $ac0, 16 \n\t"
+ "extr.w %[tmp22], $ac1, 16 \n\t"
+
+ "addu %[tmp21], %[state1], %[tmp21] \n\t"
+ "addu %[tmp22], %[state5], %[tmp22] \n\t"
+
+ "addiu %[state1], %[tmp11], 0 \n\t"
+ "addiu %[state5], %[tmp12], 0 \n\t"
+ : [tmp22] "=r" (tmp22), [tmp21] "=&r" (tmp21),
+ [tmp11] "=&r" (tmp11), [state0] "+r" (state0),
+ [state1] "+r" (state1),
+ [state2] "+r" (state2),
+ [state4] "+r" (state4), [tmp12] "=&r" (tmp12),
+ [state6] "+r" (state6), [state5] "+r" (state5)
+ : [k1Res1] "r" (k1Res1), [k2Res1] "r" (k2Res1), [k2Res0] "r" (k2Res0),
+ [diff2] "r" (diff2), [diff1] "r" (diff1), [in322] "r" (in322),
+ [in321] "r" (in321), [k1Res0] "r" (k1Res0)
+ : "hi", "lo", "$ac1hi", "$ac1lo"
+ );
+
+ // upper allpass filter
+ __asm__ volatile (
+ "subu %[diff1], %[tmp21], %[state3] \n\t"
+ "subu %[diff2], %[tmp22], %[state7] \n\t"
+
+ "mult $ac0, %[diff1], %[k2Res2] \n\t"
+ "mult $ac1, %[diff2], %[k1Res2] \n\t"
+ "extr.w %[state3], $ac0, 16 \n\t"
+ "extr.w %[state7], $ac1, 16 \n\t"
+ "addu %[state3], %[state2], %[state3] \n\t"
+ "addu %[state7], %[state6], %[state7] \n\t"
+
+ "addiu %[state2], %[tmp21], 0 \n\t"
+ "addiu %[state6], %[tmp22], 0 \n\t"
+
+ // add two allpass outputs, divide by two and round
+ "addu %[out32], %[state3], %[state7] \n\t"
+ "addiu %[out32], %[out32], 1024 \n\t"
+ "sra %[out32], %[out32], 11 \n\t"
+ : [state3] "+r" (state3), [state6] "+r" (state6),
+ [state2] "+r" (state2), [diff2] "=&r" (diff2),
+ [out32] "=r" (out32), [diff1] "=&r" (diff1), [state7] "+r" (state7)
+ : [tmp22] "r" (tmp22), [tmp21] "r" (tmp21),
+ [k1Res2] "r" (k1Res2), [k2Res2] "r" (k2Res2)
+ : "hi", "lo", "$ac1hi", "$ac1lo"
+ );
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+#else // #if defined(MIPS_DSP_R2_LE)
+ int32_t tmp1, tmp2, diff;
+ int32_t in32;
+ len1 = (len >> 1)/4;
+ for (i = len1; i > 0; i--) {
+ // lower allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ // lower allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ // lower allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ // lower allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state1;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
+ state0 = in32;
+ diff = tmp1 - state2;
+ tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
+ state1 = tmp1;
+ diff = tmp2 - state3;
+ state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
+ state2 = tmp2;
+
+ // upper allpass filter
+ in32 = (int32_t)(*in++) << 10;
+ diff = in32 - state5;
+ tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
+ state4 = in32;
+ diff = tmp1 - state6;
+ tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
+ state5 = tmp1;
+ diff = tmp2 - state7;
+ state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
+ state6 = tmp2;
+
+ // add two allpass outputs, divide by two and round
+ out32 = (state3 + state7 + 1024) >> 11;
+
+ // limit amplitude to prevent wrap-around, and write to output array
+ *out++ = WebRtcSpl_SatW32ToW16(out32);
+ }
+#endif // #if defined(MIPS_DSP_R2_LE)
+ __asm__ volatile (
+ "sw %[state0], 0(%[filtState]) \n\t"
+ "sw %[state1], 4(%[filtState]) \n\t"
+ "sw %[state2], 8(%[filtState]) \n\t"
+ "sw %[state3], 12(%[filtState]) \n\t"
+ "sw %[state4], 16(%[filtState]) \n\t"
+ "sw %[state5], 20(%[filtState]) \n\t"
+ "sw %[state6], 24(%[filtState]) \n\t"
+ "sw %[state7], 28(%[filtState]) \n\t"
+ :
+ : [state0] "r" (state0), [state1] "r" (state1), [state2] "r" (state2),
+ [state3] "r" (state3), [state4] "r" (state4), [state5] "r" (state5),
+ [state6] "r" (state6), [state7] "r" (state7), [filtState] "r" (filtState)
+ : "memory"
+ );
+}
+
+#endif // #if defined(MIPS32_LE)
diff --git a/third_party/libwebrtc/common_audio/signal_processing/resample_fractional.c b/third_party/libwebrtc/common_audio/signal_processing/resample_fractional.c
new file mode 100644
index 0000000000..9ffe0aca60
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/resample_fractional.c
@@ -0,0 +1,239 @@
+/*
+ * 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 resampling functions between 48, 44, 32 and 24 kHz.
+ * The description headers can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// interpolation coefficients
+static const int16_t kCoefficients48To32[2][8] = {
+ {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
+ {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
+};
+
+static const int16_t kCoefficients32To24[3][8] = {
+ {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
+ {386, -381, -2646, 19062, 19062, -2646, -381, 386},
+ {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
+};
+
+static const int16_t kCoefficients44To32[4][9] = {
+ {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
+ {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
+ {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
+ {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
+};
+
+// Resampling ratio: 2/3
+// input: int32_t (normalized, not saturated) :: size 3 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 2 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample48khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (3 input samples -> 2 output samples);
+ // process in sub blocks of size 3 samples.
+ int32_t tmp;
+ size_t m;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+ tmp += kCoefficients48To32[0][0] * In[0];
+ tmp += kCoefficients48To32[0][1] * In[1];
+ tmp += kCoefficients48To32[0][2] * In[2];
+ tmp += kCoefficients48To32[0][3] * In[3];
+ tmp += kCoefficients48To32[0][4] * In[4];
+ tmp += kCoefficients48To32[0][5] * In[5];
+ tmp += kCoefficients48To32[0][6] * In[6];
+ tmp += kCoefficients48To32[0][7] * In[7];
+ Out[0] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients48To32[1][0] * In[1];
+ tmp += kCoefficients48To32[1][1] * In[2];
+ tmp += kCoefficients48To32[1][2] * In[3];
+ tmp += kCoefficients48To32[1][3] * In[4];
+ tmp += kCoefficients48To32[1][4] * In[5];
+ tmp += kCoefficients48To32[1][5] * In[6];
+ tmp += kCoefficients48To32[1][6] * In[7];
+ tmp += kCoefficients48To32[1][7] * In[8];
+ Out[1] = tmp;
+
+ // update pointers
+ In += 3;
+ Out += 2;
+ }
+}
+
+// Resampling ratio: 3/4
+// input: int32_t (normalized, not saturated) :: size 4 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 3 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample32khzTo24khz(const int32_t *In, int32_t *Out, size_t K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (4 input samples -> 3 output samples);
+ // process in sub blocks of size 4 samples.
+ size_t m;
+ int32_t tmp;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[0][0] * In[0];
+ tmp += kCoefficients32To24[0][1] * In[1];
+ tmp += kCoefficients32To24[0][2] * In[2];
+ tmp += kCoefficients32To24[0][3] * In[3];
+ tmp += kCoefficients32To24[0][4] * In[4];
+ tmp += kCoefficients32To24[0][5] * In[5];
+ tmp += kCoefficients32To24[0][6] * In[6];
+ tmp += kCoefficients32To24[0][7] * In[7];
+ Out[0] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[1][0] * In[1];
+ tmp += kCoefficients32To24[1][1] * In[2];
+ tmp += kCoefficients32To24[1][2] * In[3];
+ tmp += kCoefficients32To24[1][3] * In[4];
+ tmp += kCoefficients32To24[1][4] * In[5];
+ tmp += kCoefficients32To24[1][5] * In[6];
+ tmp += kCoefficients32To24[1][6] * In[7];
+ tmp += kCoefficients32To24[1][7] * In[8];
+ Out[1] = tmp;
+
+ tmp = 1 << 14;
+ tmp += kCoefficients32To24[2][0] * In[2];
+ tmp += kCoefficients32To24[2][1] * In[3];
+ tmp += kCoefficients32To24[2][2] * In[4];
+ tmp += kCoefficients32To24[2][3] * In[5];
+ tmp += kCoefficients32To24[2][4] * In[6];
+ tmp += kCoefficients32To24[2][5] * In[7];
+ tmp += kCoefficients32To24[2][6] * In[8];
+ tmp += kCoefficients32To24[2][7] * In[9];
+ Out[2] = tmp;
+
+ // update pointers
+ In += 4;
+ Out += 3;
+ }
+}
+
+//
+// fractional resampling filters
+// Fout = 11/16 * Fin
+// Fout = 8/11 * Fin
+//
+
+// compute two inner-products and store them to output array
+static void WebRtcSpl_ResampDotProduct(const int32_t *in1, const int32_t *in2,
+ const int16_t *coef_ptr, int32_t *out1,
+ int32_t *out2)
+{
+ int32_t tmp1 = 16384;
+ int32_t tmp2 = 16384;
+ int16_t coef;
+
+ coef = coef_ptr[0];
+ tmp1 += coef * in1[0];
+ tmp2 += coef * in2[-0];
+
+ coef = coef_ptr[1];
+ tmp1 += coef * in1[1];
+ tmp2 += coef * in2[-1];
+
+ coef = coef_ptr[2];
+ tmp1 += coef * in1[2];
+ tmp2 += coef * in2[-2];
+
+ coef = coef_ptr[3];
+ tmp1 += coef * in1[3];
+ tmp2 += coef * in2[-3];
+
+ coef = coef_ptr[4];
+ tmp1 += coef * in1[4];
+ tmp2 += coef * in2[-4];
+
+ coef = coef_ptr[5];
+ tmp1 += coef * in1[5];
+ tmp2 += coef * in2[-5];
+
+ coef = coef_ptr[6];
+ tmp1 += coef * in1[6];
+ tmp2 += coef * in2[-6];
+
+ coef = coef_ptr[7];
+ tmp1 += coef * in1[7];
+ tmp2 += coef * in2[-7];
+
+ coef = coef_ptr[8];
+ *out1 = tmp1 + coef * in1[8];
+ *out2 = tmp2 + coef * in2[-8];
+}
+
+// Resampling ratio: 8/11
+// input: int32_t (normalized, not saturated) :: size 11 * K
+// output: int32_t (shifted 15 positions to the left, + offset 16384) :: size 8 * K
+// K: number of blocks
+
+void WebRtcSpl_Resample44khzTo32khz(const int32_t *In, int32_t *Out, size_t K)
+{
+ /////////////////////////////////////////////////////////////
+ // Filter operation:
+ //
+ // Perform resampling (11 input samples -> 8 output samples);
+ // process in sub blocks of size 11 samples.
+ int32_t tmp;
+ size_t m;
+
+ for (m = 0; m < K; m++)
+ {
+ tmp = 1 << 14;
+
+ // first output sample
+ Out[0] = ((int32_t)In[3] << 15) + tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ tmp += kCoefficients44To32[3][0] * In[5];
+ tmp += kCoefficients44To32[3][1] * In[6];
+ tmp += kCoefficients44To32[3][2] * In[7];
+ tmp += kCoefficients44To32[3][3] * In[8];
+ tmp += kCoefficients44To32[3][4] * In[9];
+ tmp += kCoefficients44To32[3][5] * In[10];
+ tmp += kCoefficients44To32[3][6] * In[11];
+ tmp += kCoefficients44To32[3][7] * In[12];
+ tmp += kCoefficients44To32[3][8] * In[13];
+ Out[4] = tmp;
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
+
+ // sum and accumulate filter coefficients and input samples
+ WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
+
+ // update pointers
+ In += 11;
+ Out += 8;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/signal_processing_unittest.cc b/third_party/libwebrtc/common_audio/signal_processing/signal_processing_unittest.cc
new file mode 100644
index 0000000000..80d605bc0b
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -0,0 +1,668 @@
+/*
+ * 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 <algorithm>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+static const size_t kVector16Size = 9;
+static const int16_t vector16[kVector16Size] = {1,
+ -15511,
+ 4323,
+ 1963,
+ WEBRTC_SPL_WORD16_MAX,
+ 0,
+ WEBRTC_SPL_WORD16_MIN + 5,
+ -3333,
+ 345};
+
+TEST(SplTest, MacroTest) {
+ // Macros with inputs.
+ int A = 10;
+ int B = 21;
+ int a = -3;
+ int b = WEBRTC_SPL_WORD32_MAX;
+
+ EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
+
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
+ EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
+
+ EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
+ EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
+ b = WEBRTC_SPL_WORD16_MAX >> 1;
+ EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
+
+ a = b;
+ b = -3;
+
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
+ EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
+ EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
+ EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
+
+ EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
+ EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
+
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
+ EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
+
+ // Shifting with negative numbers allowed
+ int shift_amount = 1; // Workaround compiler warning using variable here.
+ // Positive means left shift
+ EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
+
+ // Shifting with negative numbers not allowed
+ // We cannot do casting here due to signed/unsigned problem
+ EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
+
+ EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
+
+ EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
+
+ EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
+ EXPECT_EQ(1073676289,
+ WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MAX));
+ EXPECT_EQ(1073709055, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MAX,
+ WEBRTC_SPL_WORD32_MAX));
+ EXPECT_EQ(1073741824, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
+ WEBRTC_SPL_WORD32_MIN));
+#ifdef WEBRTC_ARCH_ARM_V7
+ EXPECT_EQ(-1073741824, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
+ WEBRTC_SPL_WORD32_MAX));
+#else
+ EXPECT_EQ(-1073741823, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
+ WEBRTC_SPL_WORD32_MAX));
+#endif
+}
+
+TEST(SplTest, InlineTest) {
+ int16_t a16 = 121;
+ int16_t b16 = -17;
+ int32_t a32 = 111121;
+ int32_t b32 = -1711;
+
+ EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(a32));
+
+ EXPECT_EQ(0, WebRtcSpl_NormW32(0));
+ EXPECT_EQ(31, WebRtcSpl_NormW32(-1));
+ EXPECT_EQ(0, WebRtcSpl_NormW32(WEBRTC_SPL_WORD32_MIN));
+ EXPECT_EQ(14, WebRtcSpl_NormW32(a32));
+
+ EXPECT_EQ(0, WebRtcSpl_NormW16(0));
+ EXPECT_EQ(15, WebRtcSpl_NormW16(-1));
+ EXPECT_EQ(0, WebRtcSpl_NormW16(WEBRTC_SPL_WORD16_MIN));
+ EXPECT_EQ(4, WebRtcSpl_NormW16(b32));
+ for (int ii = 0; ii < 15; ++ii) {
+ int16_t value = 1 << ii;
+ EXPECT_EQ(14 - ii, WebRtcSpl_NormW16(value));
+ EXPECT_EQ(15 - ii, WebRtcSpl_NormW16(-value));
+ }
+
+ EXPECT_EQ(0, WebRtcSpl_NormU32(0u));
+ EXPECT_EQ(0, WebRtcSpl_NormU32(0xffffffff));
+ EXPECT_EQ(15, WebRtcSpl_NormU32(static_cast<uint32_t>(a32)));
+
+ EXPECT_EQ(104, WebRtcSpl_AddSatW16(a16, b16));
+ EXPECT_EQ(138, WebRtcSpl_SubSatW16(a16, b16));
+}
+
+TEST(SplTest, AddSubSatW32) {
+ static constexpr int32_t kAddSubArgs[] = {
+ INT32_MIN, INT32_MIN + 1, -3, -2, -1, 0, 1, -1, 2,
+ 3, INT32_MAX - 1, INT32_MAX};
+ for (int32_t a : kAddSubArgs) {
+ for (int32_t b : kAddSubArgs) {
+ const int64_t sum = std::max<int64_t>(
+ INT32_MIN, std::min<int64_t>(INT32_MAX, static_cast<int64_t>(a) + b));
+ const int64_t diff = std::max<int64_t>(
+ INT32_MIN, std::min<int64_t>(INT32_MAX, static_cast<int64_t>(a) - b));
+ rtc::StringBuilder ss;
+ ss << a << " +/- " << b << ": sum " << sum << ", diff " << diff;
+ SCOPED_TRACE(ss.str());
+ EXPECT_EQ(sum, WebRtcSpl_AddSatW32(a, b));
+ EXPECT_EQ(diff, WebRtcSpl_SubSatW32(a, b));
+ }
+ }
+}
+
+TEST(SplTest, CountLeadingZeros32) {
+ EXPECT_EQ(32, WebRtcSpl_CountLeadingZeros32(0));
+ EXPECT_EQ(32, WebRtcSpl_CountLeadingZeros32_NotBuiltin(0));
+ for (int i = 0; i < 32; ++i) {
+ const uint32_t single_one = uint32_t{1} << i;
+ const uint32_t all_ones = 2 * single_one - 1;
+ EXPECT_EQ(31 - i, WebRtcSpl_CountLeadingZeros32(single_one));
+ EXPECT_EQ(31 - i, WebRtcSpl_CountLeadingZeros32_NotBuiltin(single_one));
+ EXPECT_EQ(31 - i, WebRtcSpl_CountLeadingZeros32(all_ones));
+ EXPECT_EQ(31 - i, WebRtcSpl_CountLeadingZeros32_NotBuiltin(all_ones));
+ }
+}
+
+TEST(SplTest, CountLeadingZeros64) {
+ EXPECT_EQ(64, WebRtcSpl_CountLeadingZeros64(0));
+ EXPECT_EQ(64, WebRtcSpl_CountLeadingZeros64_NotBuiltin(0));
+ for (int i = 0; i < 64; ++i) {
+ const uint64_t single_one = uint64_t{1} << i;
+ const uint64_t all_ones = 2 * single_one - 1;
+ EXPECT_EQ(63 - i, WebRtcSpl_CountLeadingZeros64(single_one));
+ EXPECT_EQ(63 - i, WebRtcSpl_CountLeadingZeros64_NotBuiltin(single_one));
+ EXPECT_EQ(63 - i, WebRtcSpl_CountLeadingZeros64(all_ones));
+ EXPECT_EQ(63 - i, WebRtcSpl_CountLeadingZeros64_NotBuiltin(all_ones));
+ }
+}
+
+TEST(SplTest, MathOperationsTest) {
+ int A = 1134567892;
+ int32_t num = 117;
+ int32_t den = -5;
+ uint16_t denU = 5;
+ EXPECT_EQ(33700, WebRtcSpl_Sqrt(A));
+ EXPECT_EQ(33683, WebRtcSpl_SqrtFloor(A));
+
+ EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (int16_t)den));
+ EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (int16_t)den));
+ EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
+ EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
+}
+
+TEST(SplTest, BasicArrayOperationsTest) {
+ const size_t kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ int16_t b16[kVectorSize];
+ int32_t b32[kVectorSize];
+
+ int16_t bTmp16[kVectorSize];
+ int32_t bTmp32[kVectorSize];
+
+ WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b16[kk]);
+ }
+ WebRtcSpl_ZerosArrayW16(b16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b16[kk]);
+ }
+ WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(3, b32[kk]);
+ }
+ WebRtcSpl_ZerosArrayW32(b32, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(0, b32[kk]);
+ }
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ bTmp16[kk] = (int16_t)kk;
+ bTmp32[kk] = (int32_t)kk;
+ }
+ WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[kk], bTmp16[kk]);
+ }
+ // WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
+ // for (int kk = 0; kk < kVectorSize; ++kk) {
+ // EXPECT_EQ(b32[kk], bTmp32[kk]);
+ // }
+ WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16);
+ for (size_t kk = 0; kk < 2; ++kk) {
+ EXPECT_EQ(static_cast<int16_t>(kk + 2), bTmp16[kk]);
+ }
+
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ b32[kk] = B[kk];
+ b16[kk] = (int16_t)B[kk];
+ }
+ WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk] >> 1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk] >> 1), bTmp16[kk]);
+ }
+ WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk] >> 1), bTmp32[kk]);
+ }
+
+ WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(b16[3 - kk], bTmp16[kk]);
+ }
+}
+
+TEST(SplTest, MinMaxOperationsTest) {
+ const size_t kVectorSize = 17;
+
+ // Vectors to test the cases where minimum values have to be caught
+ // outside of the unrolled loops in ARM-Neon.
+ int16_t vector16[kVectorSize] = {-1,
+ 7485,
+ 0,
+ 3333,
+ -18283,
+ 0,
+ 12334,
+ -29871,
+ 988,
+ -3333,
+ 345,
+ -456,
+ 222,
+ 999,
+ 888,
+ 8774,
+ WEBRTC_SPL_WORD16_MIN};
+ int32_t vector32[kVectorSize] = {-1,
+ 0,
+ 283211,
+ 3333,
+ 8712345,
+ 0,
+ -3333,
+ 89345,
+ -374585456,
+ 222,
+ 999,
+ 122345334,
+ -12389756,
+ -987329871,
+ 888,
+ -2,
+ WEBRTC_SPL_WORD32_MIN};
+
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MinValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MIN,
+ WebRtcSpl_MinValueW32(vector32, kVectorSize));
+ EXPECT_EQ(kVectorSize - 1, WebRtcSpl_MinIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(kVectorSize - 1, WebRtcSpl_MinIndexW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MaxAbsElementW16(vector16, kVectorSize));
+ int16_t min_value, max_value;
+ WebRtcSpl_MinMaxW16(vector16, kVectorSize, &min_value, &max_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN, min_value);
+ EXPECT_EQ(12334, max_value);
+
+ // Test the cases where maximum values have to be caught
+ // outside of the unrolled loops in ARM-Neon.
+ vector16[kVectorSize - 1] = WEBRTC_SPL_WORD16_MAX;
+ vector32[kVectorSize - 1] = WEBRTC_SPL_WORD32_MAX;
+
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxAbsValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MAX,
+ WebRtcSpl_MaxAbsValueW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MAX,
+ WebRtcSpl_MaxValueW32(vector32, kVectorSize));
+ EXPECT_EQ(kVectorSize - 1, WebRtcSpl_MaxAbsIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(kVectorSize - 1, WebRtcSpl_MaxIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(kVectorSize - 1, WebRtcSpl_MaxIndexW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxAbsElementW16(vector16, kVectorSize));
+ WebRtcSpl_MinMaxW16(vector16, kVectorSize, &min_value, &max_value);
+ EXPECT_EQ(-29871, min_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX, max_value);
+
+ // Test the cases where multiple maximum and minimum values are present.
+ vector16[1] = WEBRTC_SPL_WORD16_MAX;
+ vector16[6] = WEBRTC_SPL_WORD16_MIN;
+ vector16[11] = WEBRTC_SPL_WORD16_MIN;
+ vector32[1] = WEBRTC_SPL_WORD32_MAX;
+ vector32[6] = WEBRTC_SPL_WORD32_MIN;
+ vector32[11] = WEBRTC_SPL_WORD32_MIN;
+
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxAbsValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MinValueW16(vector16, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MAX,
+ WebRtcSpl_MaxAbsValueW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MAX,
+ WebRtcSpl_MaxValueW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD32_MIN,
+ WebRtcSpl_MinValueW32(vector32, kVectorSize));
+ EXPECT_EQ(6u, WebRtcSpl_MaxAbsIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(1u, WebRtcSpl_MaxIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(1u, WebRtcSpl_MaxIndexW32(vector32, kVectorSize));
+ EXPECT_EQ(6u, WebRtcSpl_MinIndexW16(vector16, kVectorSize));
+ EXPECT_EQ(6u, WebRtcSpl_MinIndexW32(vector32, kVectorSize));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MaxAbsElementW16(vector16, kVectorSize));
+ WebRtcSpl_MinMaxW16(vector16, kVectorSize, &min_value, &max_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN, min_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX, max_value);
+
+ // Test a one-element vector.
+ int16_t single_element_vector = 0;
+ EXPECT_EQ(0, WebRtcSpl_MaxAbsValueW16(&single_element_vector, 1));
+ EXPECT_EQ(0, WebRtcSpl_MaxValueW16(&single_element_vector, 1));
+ EXPECT_EQ(0, WebRtcSpl_MinValueW16(&single_element_vector, 1));
+ EXPECT_EQ(0u, WebRtcSpl_MaxAbsIndexW16(&single_element_vector, 1));
+ EXPECT_EQ(0u, WebRtcSpl_MaxIndexW16(&single_element_vector, 1));
+ EXPECT_EQ(0u, WebRtcSpl_MinIndexW16(&single_element_vector, 1));
+ EXPECT_EQ(0, WebRtcSpl_MaxAbsElementW16(&single_element_vector, 1));
+ WebRtcSpl_MinMaxW16(&single_element_vector, 1, &min_value, &max_value);
+ EXPECT_EQ(0, min_value);
+ EXPECT_EQ(0, max_value);
+
+ // Test a two-element vector with the values WEBRTC_SPL_WORD16_MIN and
+ // WEBRTC_SPL_WORD16_MAX.
+ int16_t two_element_vector[2] = {WEBRTC_SPL_WORD16_MIN,
+ WEBRTC_SPL_WORD16_MAX};
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxAbsValueW16(two_element_vector, 2));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX,
+ WebRtcSpl_MaxValueW16(two_element_vector, 2));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MinValueW16(two_element_vector, 2));
+ EXPECT_EQ(0u, WebRtcSpl_MaxAbsIndexW16(two_element_vector, 2));
+ EXPECT_EQ(1u, WebRtcSpl_MaxIndexW16(two_element_vector, 2));
+ EXPECT_EQ(0u, WebRtcSpl_MinIndexW16(two_element_vector, 2));
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
+ WebRtcSpl_MaxAbsElementW16(two_element_vector, 2));
+ WebRtcSpl_MinMaxW16(two_element_vector, 2, &min_value, &max_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MIN, min_value);
+ EXPECT_EQ(WEBRTC_SPL_WORD16_MAX, max_value);
+}
+
+TEST(SplTest, VectorOperationsTest) {
+ const size_t kVectorSize = 4;
+ int B[] = {4, 12, 133, 1100};
+ int16_t a16[kVectorSize];
+ int16_t b16[kVectorSize];
+ int16_t bTmp16[kVectorSize];
+
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ a16[kk] = B[kk];
+ b16[kk] = B[kk];
+ }
+
+ WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk] * 3 + 7) >> 2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((B[kk] * 3 + B[kk] * 2 + 2) >> 2, bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((B[kk] * 3 + B[kk] * 2 + 2) >> 2) + ((b16[kk] * 3 + 7) >> 2),
+ bTmp16[kk]);
+ }
+
+ WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk] * 13) >> 2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((b16[kk] * 13) >> 2, bTmp16[kk]);
+ }
+ WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(((a16[kk] * 13) >> 2) + ((b16[kk] * 7) >> 2), bTmp16[kk]);
+ }
+
+ WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
+ }
+ WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk] * b16[3 - kk]) >> 2, bTmp16[kk]);
+ }
+ WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ((a16[kk] * b16[kk]) >> 6, bTmp16[kk]);
+ }
+
+ WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
+ for (size_t kk = 0; kk < kVectorSize - 1; ++kk) {
+ EXPECT_EQ(32767, bTmp16[kk]);
+ }
+ EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
+
+ EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
+}
+
+TEST(SplTest, EstimatorsTest) {
+ const size_t kOrder = 2;
+ const int32_t unstable_filter[] = {4, 12, 133, 1100};
+ const int32_t stable_filter[] = {1100, 133, 12, 4};
+ int16_t lpc[kOrder + 2] = {0};
+ int16_t refl[kOrder + 2] = {0};
+ int16_t lpc_result[] = {4096, -497, 15, 0};
+ int16_t refl_result[] = {-3962, 123, 0, 0};
+
+ EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(unstable_filter, lpc, refl, kOrder));
+ EXPECT_EQ(1, WebRtcSpl_LevinsonDurbin(stable_filter, lpc, refl, kOrder));
+ for (size_t i = 0; i < kOrder + 2; ++i) {
+ EXPECT_EQ(lpc_result[i], lpc[i]);
+ EXPECT_EQ(refl_result[i], refl[i]);
+ }
+}
+
+TEST(SplTest, FilterTest) {
+ const size_t kVectorSize = 4;
+ const size_t kFilterOrder = 3;
+ int16_t A[] = {1, 2, 33, 100};
+ int16_t A5[] = {1, 2, 33, 100, -5};
+ int16_t B[] = {4, 12, 133, 110};
+ int16_t data_in[kVectorSize];
+ int16_t data_out[kVectorSize];
+ int16_t bTmp16Low[kVectorSize];
+ int16_t bState[kVectorSize];
+ int16_t bStateLow[kVectorSize];
+
+ WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
+ WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
+
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ data_in[kk] = A[kk];
+ data_out[kk] = 0;
+ }
+
+ // MA filters.
+ // Note that the input data has `kFilterOrder` states before the actual
+ // data (one sample).
+ WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
+ kFilterOrder + 1, 1);
+ EXPECT_EQ(0, data_out[0]);
+ // AR filters.
+ // Note that the output data has `kFilterOrder` states before the actual
+ // data (one sample).
+ WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
+ kFilterOrder + 1, 1);
+ EXPECT_EQ(0, data_out[kFilterOrder]);
+
+ EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5, 5, data_in, kVectorSize, bState,
+ kVectorSize, bStateLow, kVectorSize,
+ data_out, bTmp16Low, kVectorSize));
+}
+
+TEST(SplTest, RandTest) {
+ const int kVectorSize = 4;
+ int16_t BU[] = {3653, 12446, 8525, 30691};
+ int16_t b16[kVectorSize];
+ uint32_t bSeed = 100000;
+
+ EXPECT_EQ(7086, WebRtcSpl_RandU(&bSeed));
+ EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
+ EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
+ EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
+ for (int kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(BU[kk], b16[kk]);
+ }
+}
+
+TEST(SplTest, DotProductWithScaleTest) {
+ EXPECT_EQ(605362796, WebRtcSpl_DotProductWithScale(vector16, vector16,
+ kVector16Size, 2));
+}
+
+TEST(SplTest, CrossCorrelationTest) {
+ // Note the function arguments relation specificed by API.
+ const size_t kCrossCorrelationDimension = 3;
+ const int kShift = 2;
+ const int kStep = 1;
+ const size_t kSeqDimension = 6;
+
+ const int16_t kVector16[kVector16Size] = {
+ 1, 4323, 1963, WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MIN + 5, -3333,
+ -876, 8483, 142};
+ int32_t vector32[kCrossCorrelationDimension] = {0};
+
+ WebRtcSpl_CrossCorrelation(vector32, vector16, kVector16, kSeqDimension,
+ kCrossCorrelationDimension, kShift, kStep);
+
+ // WebRtcSpl_CrossCorrelationC() and WebRtcSpl_CrossCorrelationNeon()
+ // are not bit-exact.
+ const int32_t kExpected[kCrossCorrelationDimension] = {-266947903, -15579555,
+ -171282001};
+ const int32_t* expected = kExpected;
+#if !defined(MIPS32_LE)
+ const int32_t kExpectedNeon[kCrossCorrelationDimension] = {
+ -266947901, -15579553, -171281999};
+ if (WebRtcSpl_CrossCorrelation != WebRtcSpl_CrossCorrelationC) {
+ expected = kExpectedNeon;
+ }
+#endif
+ for (size_t i = 0; i < kCrossCorrelationDimension; ++i) {
+ EXPECT_EQ(expected[i], vector32[i]);
+ }
+}
+
+TEST(SplTest, AutoCorrelationTest) {
+ int scale = 0;
+ int32_t vector32[kVector16Size];
+ const int32_t expected[kVector16Size] = {302681398, 14223410, -121705063,
+ -85221647, -17104971, 61806945,
+ 6644603, -669329, 43};
+
+ EXPECT_EQ(kVector16Size,
+ WebRtcSpl_AutoCorrelation(vector16, kVector16Size,
+ kVector16Size - 1, vector32, &scale));
+ EXPECT_EQ(3, scale);
+ for (size_t i = 0; i < kVector16Size; ++i) {
+ EXPECT_EQ(expected[i], vector32[i]);
+ }
+}
+
+TEST(SplTest, SignalProcessingTest) {
+ const size_t kVectorSize = 4;
+ int A[] = {1, 2, 33, 100};
+ const int16_t kHanning[4] = {2399, 8192, 13985, 16384};
+ int16_t b16[kVectorSize];
+
+ int16_t bTmp16[kVectorSize];
+
+ int bScale = 0;
+
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+
+ // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
+ // WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
+ //// for (int kk = 0; kk < kVectorSize; ++kk) {
+ //// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+ //// }
+ // WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
+ //// for (int kk = 0; kk < kVectorSize; ++kk) {
+ //// EXPECT_EQ(a16[kk], b16[kk]);
+ //// }
+ // WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
+ //// for (int kk = 0; kk < kVectorSize; ++kk) {
+ //// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
+ //// }
+
+ WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ EXPECT_EQ(kHanning[kk], bTmp16[kk]);
+ }
+
+ for (size_t kk = 0; kk < kVectorSize; ++kk) {
+ b16[kk] = A[kk];
+ }
+ EXPECT_EQ(11094, WebRtcSpl_Energy(b16, kVectorSize, &bScale));
+ EXPECT_EQ(0, bScale);
+}
+
+TEST(SplTest, FFTTest) {
+ int16_t B[] = {1, 2, 33, 100, 2, 3, 34, 101, 3, 4, 35, 102, 4, 5, 36, 103};
+
+ EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
+ // for (int kk = 0; kk < 16; ++kk) {
+ // EXPECT_EQ(A[kk], B[kk]);
+ // }
+ EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
+ // for (int kk = 0; kk < 16; ++kk) {
+ // EXPECT_EQ(A[kk], B[kk]);
+ // }
+ WebRtcSpl_ComplexBitReverse(B, 3);
+ for (int kk = 0; kk < 16; ++kk) {
+ // EXPECT_EQ(A[kk], B[kk]);
+ }
+}
+
+TEST(SplTest, Resample48WithSaturationTest) {
+ // The test resamples 3*kBlockSize number of samples to 2*kBlockSize number
+ // of samples.
+ const size_t kBlockSize = 16;
+
+ // Saturated input vector of 48 samples.
+ const int32_t kVectorSaturated[3 * kBlockSize + 7] = {
+ -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+ -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+ -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
+ 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+ 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+ 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
+ 32767, 32767, 32767, 32767, 32767, 32767, 32767};
+
+ // All values in `out_vector` should be `kRefValue32kHz`.
+ const int32_t kRefValue32kHz1 = -1077493760;
+ const int32_t kRefValue32kHz2 = 1077493645;
+
+ // After bit shift with saturation, `out_vector_w16` is saturated.
+
+ const int16_t kRefValue16kHz1 = -32768;
+ const int16_t kRefValue16kHz2 = 32767;
+ // Vector for storing output.
+ int32_t out_vector[2 * kBlockSize];
+ int16_t out_vector_w16[2 * kBlockSize];
+
+ WebRtcSpl_Resample48khzTo32khz(kVectorSaturated, out_vector, kBlockSize);
+ WebRtcSpl_VectorBitShiftW32ToW16(out_vector_w16, 2 * kBlockSize, out_vector,
+ 15);
+
+ // Comparing output values against references. The values at position
+ // 12-15 are skipped to account for the filter lag.
+ for (size_t i = 0; i < 12; ++i) {
+ EXPECT_EQ(kRefValue32kHz1, out_vector[i]);
+ EXPECT_EQ(kRefValue16kHz1, out_vector_w16[i]);
+ }
+ for (size_t i = 16; i < 2 * kBlockSize; ++i) {
+ EXPECT_EQ(kRefValue32kHz2, out_vector[i]);
+ EXPECT_EQ(kRefValue16kHz2, out_vector_w16[i]);
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/spl_init.c b/third_party/libwebrtc/common_audio/signal_processing/spl_init.c
new file mode 100644
index 0000000000..cf37d47bec
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/spl_init.c
@@ -0,0 +1,69 @@
+/*
+ * 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
diff --git a/third_party/libwebrtc/common_audio/signal_processing/spl_inl.c b/third_party/libwebrtc/common_audio/signal_processing/spl_inl.c
new file mode 100644
index 0000000000..d09e308ed3
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/spl_inl.c
@@ -0,0 +1,24 @@
+/*
+ * Copyright (c) 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.
+ */
+
+#include <stdint.h>
+
+#include "common_audio/signal_processing/include/spl_inl.h"
+
+// Table used by WebRtcSpl_CountLeadingZeros32_NotBuiltin. For each uint32_t n
+// that's a sequence of 0 bits followed by a sequence of 1 bits, the entry at
+// index (n * 0x8c0b2891) >> 26 in this table gives the number of zero bits in
+// n.
+const int8_t kWebRtcSpl_CountLeadingZeros32_Table[64] = {
+ 32, 8, 17, -1, -1, 14, -1, -1, -1, 20, -1, -1, -1, 28, -1, 18,
+ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 0, 26, 25, 24,
+ 4, 11, 23, 31, 3, 7, 10, 16, 22, 30, -1, -1, 2, 6, 13, 9,
+ -1, 15, -1, 21, -1, 29, 19, -1, -1, -1, -1, -1, 1, 27, 5, 12,
+};
diff --git a/third_party/libwebrtc/common_audio/signal_processing/spl_sqrt.c b/third_party/libwebrtc/common_audio/signal_processing/spl_sqrt.c
new file mode 100644
index 0000000000..cf9448ac97
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/spl_sqrt.c
@@ -0,0 +1,194 @@
+/*
+ * 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_Sqrt().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+int32_t WebRtcSpl_SqrtLocal(int32_t in);
+
+int32_t WebRtcSpl_SqrtLocal(int32_t in)
+{
+
+ int16_t x_half, t16;
+ int32_t A, B, x2;
+
+ /* The following block performs:
+ y=in/2
+ x=y-2^30
+ x_half=x/2^31
+ t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ + 0.875*((x_half)^5)
+ */
+
+ B = in / 2;
+
+ B = B - ((int32_t)0x40000000); // B = in/2 - 1/2
+ x_half = (int16_t)(B >> 16); // x_half = x/2 = (in-1)/2
+ B = B + ((int32_t)0x40000000); // B = 1 + x/2
+ B = B + ((int32_t)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
+
+ x2 = ((int32_t)x_half) * ((int32_t)x_half) * 2; // A = (x/2)^2
+ A = -x2; // A = -(x/2)^2
+ B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
+
+ A >>= 16;
+ A = A * A * 2; // A = (x/2)^4
+ t16 = (int16_t)(A >> 16);
+ B += -20480 * t16 * 2; // B = B - 0.625*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
+
+ A = x_half * t16 * 2; // A = (x/2)^5
+ t16 = (int16_t)(A >> 16);
+ B += 28672 * t16 * 2; // B = B + 0.875*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+ t16 = (int16_t)(x2 >> 16);
+ A = x_half * t16 * 2; // A = x/2^3
+
+ B = B + (A >> 1); // B = B + 0.5*A
+ // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
+
+ B = B + ((int32_t)32768); // Round off bit
+
+ return B;
+}
+
+int32_t WebRtcSpl_Sqrt(int32_t value)
+{
+ /*
+ Algorithm:
+
+ Six term Taylor Series is used here to compute the square root of a number
+ y^0.5 = (1+x)^0.5 where x = y-1
+ = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
+ 0.5 <= x < 1
+
+ Example of how the algorithm works, with ut=sqrt(in), and
+ with in=73632 and ut=271 (even shift value case):
+
+ in=73632
+ y= in/131072
+ x=y-1
+ t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+ ut=t*(1/sqrt(2))*512
+
+ or:
+
+ in=73632
+ in2=73632*2^14
+ y= in2/2^31
+ x=y-1
+ t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
+ ut=t*(1/sqrt(2))
+ ut2=ut*2^9
+
+ which gives:
+
+ in = 73632
+ in2 = 1206386688
+ y = 0.56176757812500
+ x = -0.43823242187500
+ t = 0.74973506527313
+ ut = 0.53014274874797
+ ut2 = 2.714330873589594e+002
+
+ or:
+
+ in=73632
+ in2=73632*2^14
+ y=in2/2
+ x=y-2^30
+ x_half=x/2^31
+ t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
+ + 0.875*((x_half)^5)
+ ut=t*(1/sqrt(2))
+ ut2=ut*2^9
+
+ which gives:
+
+ in = 73632
+ in2 = 1206386688
+ y = 603193344
+ x = -470548480
+ x_half = -0.21911621093750
+ t = 0.74973506527313
+ ut = 0.53014274874797
+ ut2 = 2.714330873589594e+002
+
+ */
+
+ int16_t x_norm, nshift, t16, sh;
+ int32_t A;
+
+ int16_t k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
+
+ A = value;
+
+ // The convention in this function is to calculate sqrt(abs(A)). Negate the
+ // input if it is negative.
+ if (A < 0) {
+ if (A == WEBRTC_SPL_WORD32_MIN) {
+ // This number cannot be held in an int32_t after negating.
+ // Map it to the maximum positive value.
+ A = WEBRTC_SPL_WORD32_MAX;
+ } else {
+ A = -A;
+ }
+ } else if (A == 0) {
+ return 0; // sqrt(0) = 0
+ }
+
+ sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
+ A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
+ if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
+ {
+ A = A + ((int32_t)32768); // Round off bit
+ } else
+ {
+ A = WEBRTC_SPL_WORD32_MAX;
+ }
+
+ x_norm = (int16_t)(A >> 16); // x_norm = AH
+
+ nshift = (sh / 2);
+ RTC_DCHECK_GE(nshift, 0);
+
+ A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
+ A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
+ A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
+
+ if (2 * nshift == sh) {
+ // Even shift value case
+
+ t16 = (int16_t)(A >> 16); // t16 = AH
+
+ A = k_sqrt_2 * t16 * 2; // A = 1/sqrt(2)*t16
+ A = A + ((int32_t)32768); // Round off
+ A = A & ((int32_t)0x7fff0000); // Round off
+
+ A >>= 15; // A = A>>16
+
+ } else
+ {
+ A >>= 16; // A = A>>16
+ }
+
+ A = A & ((int32_t)0x0000ffff);
+ A >>= nshift; // De-normalize the result.
+
+ return A;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/splitting_filter.c b/third_party/libwebrtc/common_audio/signal_processing/splitting_filter.c
new file mode 100644
index 0000000000..27a0a2a8c9
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/splitting_filter.c
@@ -0,0 +1,211 @@
+/*
+ * 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 splitting filter functions.
+ *
+ */
+
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+// Maximum number of samples in a low/high-band frame.
+enum
+{
+ kMaxBandFrameLength = 320 // 10 ms at 64 kHz.
+};
+
+// QMF filter coefficients in Q16.
+static const uint16_t WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
+static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+// WebRtcSpl_AllPassQMF(...)
+//
+// Allpass filter used by the analysis and synthesis parts of the QMF filter.
+//
+// Input:
+// - in_data : Input data sequence (Q10)
+// - data_length : Length of data sequence (>2)
+// - filter_coefficients : Filter coefficients (length 3, Q16)
+//
+// Input & Output:
+// - filter_state : Filter state (length 6, Q10).
+//
+// Output:
+// - out_data : Output data sequence (Q10), length equal to
+// `data_length`
+//
+
+static void WebRtcSpl_AllPassQMF(int32_t* in_data,
+ size_t data_length,
+ int32_t* out_data,
+ const uint16_t* filter_coefficients,
+ int32_t* filter_state)
+{
+ // The procedure is to filter the input with three first order all pass
+ // filters (cascade operations).
+ //
+ // a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
+ // y[n] = ----------- ----------- ----------- x[n]
+ // 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
+ //
+ // The input vector `filter_coefficients` includes these three filter
+ // coefficients. The filter state contains the in_data state, in_data[-1],
+ // followed by the out_data state, out_data[-1]. This is repeated for each
+ // cascade. The first cascade filter will filter the `in_data` and store
+ // the output in `out_data`. The second will the take the `out_data` as
+ // input and make an intermediate storage in `in_data`, to save memory. The
+ // third, and final, cascade filter operation takes the `in_data` (which is
+ // the output from the previous cascade filter) and store the output in
+ // `out_data`. Note that the input vector values are changed during the
+ // process.
+ size_t k;
+ int32_t diff;
+ // First all-pass cascade; filter from in_data to out_data.
+
+ // Let y_i[n] indicate the output of cascade filter i (with filter
+ // coefficient a_i) at vector position n. Then the final output will be
+ // y[n] = y_3[n]
+
+ // First loop, use the states stored in memory.
+ // "diff" should be safe from wrap around since max values are 2^25
+ // diff = (x[0] - y_1[-1])
+ diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[1]);
+ // y_1[0] = x[-1] + a_1 * (x[0] - y_1[-1])
+ out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
+
+ // For the remaining loops, use previous values.
+ for (k = 1; k < data_length; k++)
+ {
+ // diff = (x[n] - y_1[n-1])
+ diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
+ // y_1[n] = x[n-1] + a_1 * (x[n] - y_1[n-1])
+ out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
+ }
+
+ // Update states.
+ filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
+ filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+
+ // Second all-pass cascade; filter from out_data to in_data.
+ // diff = (y_1[0] - y_2[-1])
+ diff = WebRtcSpl_SubSatW32(out_data[0], filter_state[3]);
+ // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+ in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
+ for (k = 1; k < data_length; k++)
+ {
+ // diff = (y_1[n] - y_2[n-1])
+ diff = WebRtcSpl_SubSatW32(out_data[k], in_data[k - 1]);
+ // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
+ in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
+ }
+
+ filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
+ filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+
+ // Third all-pass cascade; filter from in_data to out_data.
+ // diff = (y_2[0] - y[-1])
+ diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[5]);
+ // y[0] = y_2[-1] + a_3 * (y_2[0] - y[-1])
+ out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
+ for (k = 1; k < data_length; k++)
+ {
+ // diff = (y_2[n] - y[n-1])
+ diff = WebRtcSpl_SubSatW32(in_data[k], out_data[k - 1]);
+ // y[n] = y_2[n-1] + a_3 * (y_2[n] - y[n-1])
+ out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
+ }
+ filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
+ filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
+}
+
+void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
+ int16_t* low_band, int16_t* high_band,
+ int32_t* filter_state1, int32_t* filter_state2)
+{
+ size_t i;
+ int16_t k;
+ int32_t tmp;
+ int32_t half_in1[kMaxBandFrameLength];
+ int32_t half_in2[kMaxBandFrameLength];
+ int32_t filter1[kMaxBandFrameLength];
+ int32_t filter2[kMaxBandFrameLength];
+ const size_t band_length = in_data_length / 2;
+ RTC_DCHECK_EQ(0, in_data_length % 2);
+ RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
+
+ // Split even and odd samples. Also shift them to Q10.
+ for (i = 0, k = 0; i < band_length; i++, k += 2)
+ {
+ half_in2[i] = ((int32_t)in_data[k]) * (1 << 10);
+ half_in1[i] = ((int32_t)in_data[k + 1]) * (1 << 10);
+ }
+
+ // All pass filter even and odd samples, independently.
+ WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
+ WebRtcSpl_kAllPassFilter1, filter_state1);
+ WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
+ WebRtcSpl_kAllPassFilter2, filter_state2);
+
+ // Take the sum and difference of filtered version of odd and even
+ // branches to get upper & lower band.
+ for (i = 0; i < band_length; i++)
+ {
+ tmp = (filter1[i] + filter2[i] + 1024) >> 11;
+ low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+
+ tmp = (filter1[i] - filter2[i] + 1024) >> 11;
+ high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
+ }
+}
+
+void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
+ size_t band_length, int16_t* out_data,
+ int32_t* filter_state1, int32_t* filter_state2)
+{
+ int32_t tmp;
+ int32_t half_in1[kMaxBandFrameLength];
+ int32_t half_in2[kMaxBandFrameLength];
+ int32_t filter1[kMaxBandFrameLength];
+ int32_t filter2[kMaxBandFrameLength];
+ size_t i;
+ int16_t k;
+ RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
+
+ // Obtain the sum and difference channels out of upper and lower-band channels.
+ // Also shift to Q10 domain.
+ for (i = 0; i < band_length; i++)
+ {
+ tmp = (int32_t)low_band[i] + (int32_t)high_band[i];
+ half_in1[i] = tmp * (1 << 10);
+ tmp = (int32_t)low_band[i] - (int32_t)high_band[i];
+ half_in2[i] = tmp * (1 << 10);
+ }
+
+ // all-pass filter the sum and difference channels
+ WebRtcSpl_AllPassQMF(half_in1, band_length, filter1,
+ WebRtcSpl_kAllPassFilter2, filter_state1);
+ WebRtcSpl_AllPassQMF(half_in2, band_length, filter2,
+ WebRtcSpl_kAllPassFilter1, filter_state2);
+
+ // The filtered signals are even and odd samples of the output. Combine
+ // them. The signals are Q10 should shift them back to Q0 and take care of
+ // saturation.
+ for (i = 0, k = 0; i < band_length; i++)
+ {
+ tmp = (filter2[i] + 512) >> 10;
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+
+ tmp = (filter1[i] + 512) >> 10;
+ out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
+ }
+
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/third_party/libwebrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
new file mode 100644
index 0000000000..a77fd4063f
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
@@ -0,0 +1,35 @@
+/*
+ * 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_SqrtOfOneMinusXSquared().
+ * The description header can be found in signal_processing_library.h
+ *
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, size_t vector_length,
+ int16_t *yQ15)
+{
+ int32_t sq;
+ size_t m;
+ int16_t tmp;
+
+ for (m = 0; m < vector_length; m++)
+ {
+ tmp = xQ15[m];
+ sq = tmp * tmp; // x^2 in Q30
+ sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
+ sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
+ yQ15[m] = (int16_t)sq;
+ }
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations.c b/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations.c
new file mode 100644
index 0000000000..7307dc78ff
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations.c
@@ -0,0 +1,165 @@
+/*
+ * 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;
+}
diff --git a/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations_mips.c b/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations_mips.c
new file mode 100644
index 0000000000..ba2d26d422
--- /dev/null
+++ b/third_party/libwebrtc/common_audio/signal_processing/vector_scaling_operations_mips.c
@@ -0,0 +1,57 @@
+/*
+ * 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.
+ */
+
+
+/*
+ * This file contains implementations of the functions
+ * WebRtcSpl_ScaleAndAddVectorsWithRound_mips()
+ */
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(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) {
+ int16_t r0 = 0, r1 = 0;
+ int16_t *in1 = (int16_t*)in_vector1;
+ int16_t *in2 = (int16_t*)in_vector2;
+ int16_t *out = out_vector;
+ size_t i = 0;
+ int value32 = 0;
+
+ if (in_vector1 == NULL || in_vector2 == NULL || out_vector == NULL ||
+ length == 0 || right_shifts < 0) {
+ return -1;
+ }
+ for (i = 0; i < length; i++) {
+ __asm __volatile (
+ "lh %[r0], 0(%[in1]) \n\t"
+ "lh %[r1], 0(%[in2]) \n\t"
+ "mult %[r0], %[in_vector1_scale] \n\t"
+ "madd %[r1], %[in_vector2_scale] \n\t"
+ "extrv_r.w %[value32], $ac0, %[right_shifts] \n\t"
+ "addiu %[in1], %[in1], 2 \n\t"
+ "addiu %[in2], %[in2], 2 \n\t"
+ "sh %[value32], 0(%[out]) \n\t"
+ "addiu %[out], %[out], 2 \n\t"
+ : [value32] "=&r" (value32), [out] "+r" (out), [in1] "+r" (in1),
+ [in2] "+r" (in2), [r0] "=&r" (r0), [r1] "=&r" (r1)
+ : [in_vector1_scale] "r" (in_vector1_scale),
+ [in_vector2_scale] "r" (in_vector2_scale),
+ [right_shifts] "r" (right_shifts)
+ : "hi", "lo", "memory"
+ );
+ }
+ return 0;
+}