mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-30 21:45:20 +00:00
d83a3d71bc
Merge in RedPhone // FREEBIE
180 lines
5.9 KiB
C
180 lines
5.9 KiB
C
/*
|
|
* 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 "webrtc/common_audio/vad/vad_sp.h"
|
|
|
|
#include <assert.h>
|
|
|
|
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
|
#include "webrtc/common_audio/vad/vad_core.h"
|
|
#include "webrtc/typedefs.h"
|
|
|
|
// Allpass filter coefficients, upper and lower, in Q13.
|
|
// Upper: 0.64, Lower: 0.17.
|
|
static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13.
|
|
static const int16_t kSmoothingDown = 6553; // 0.2 in Q15.
|
|
static const int16_t kSmoothingUp = 32439; // 0.99 in Q15.
|
|
|
|
// TODO(bjornv): Move this function to vad_filterbank.c.
|
|
// Downsampling filter based on splitting filter and allpass functions.
|
|
void WebRtcVad_Downsampling(const int16_t* signal_in,
|
|
int16_t* signal_out,
|
|
int32_t* filter_state,
|
|
int in_length) {
|
|
int16_t tmp16_1 = 0, tmp16_2 = 0;
|
|
int32_t tmp32_1 = filter_state[0];
|
|
int32_t tmp32_2 = filter_state[1];
|
|
int n = 0;
|
|
int half_length = (in_length >> 1); // Downsampling by 2 gives half length.
|
|
|
|
// Filter coefficients in Q13, filter state in Q0.
|
|
for (n = 0; n < half_length; n++) {
|
|
// All-pass filtering upper branch.
|
|
tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
|
|
WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
|
|
*signal_out = tmp16_1;
|
|
tmp32_1 = (int32_t) (*signal_in++) -
|
|
WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
|
|
|
|
// All-pass filtering lower branch.
|
|
tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
|
|
WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
|
|
*signal_out++ += tmp16_2;
|
|
tmp32_2 = (int32_t) (*signal_in++) -
|
|
WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
|
|
}
|
|
// Store the filter states.
|
|
filter_state[0] = tmp32_1;
|
|
filter_state[1] = tmp32_2;
|
|
}
|
|
|
|
// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
|
|
// smallest values the last 100 frames. Then calculates and returns the median
|
|
// of the five smallest values.
|
|
int16_t WebRtcVad_FindMinimum(VadInstT* self,
|
|
int16_t feature_value,
|
|
int channel) {
|
|
int i = 0, j = 0;
|
|
int position = -1;
|
|
// Offset to beginning of the 16 minimum values in memory.
|
|
const int offset = (channel << 4);
|
|
int16_t current_median = 1600;
|
|
int16_t alpha = 0;
|
|
int32_t tmp32 = 0;
|
|
// Pointer to memory for the 16 minimum values and the age of each value of
|
|
// the |channel|.
|
|
int16_t* age = &self->index_vector[offset];
|
|
int16_t* smallest_values = &self->low_value_vector[offset];
|
|
|
|
assert(channel < kNumChannels);
|
|
|
|
// Each value in |smallest_values| is getting 1 loop older. Update |age|, and
|
|
// remove old values.
|
|
for (i = 0; i < 16; i++) {
|
|
if (age[i] != 100) {
|
|
age[i]++;
|
|
} else {
|
|
// Too old value. Remove from memory and shift larger values downwards.
|
|
for (j = i; j < 16; j++) {
|
|
smallest_values[j] = smallest_values[j + 1];
|
|
age[j] = age[j + 1];
|
|
}
|
|
age[15] = 101;
|
|
smallest_values[15] = 10000;
|
|
}
|
|
}
|
|
|
|
// Check if |feature_value| is smaller than any of the values in
|
|
// |smallest_values|. If so, find the |position| where to insert the new value
|
|
// (|feature_value|).
|
|
if (feature_value < smallest_values[7]) {
|
|
if (feature_value < smallest_values[3]) {
|
|
if (feature_value < smallest_values[1]) {
|
|
if (feature_value < smallest_values[0]) {
|
|
position = 0;
|
|
} else {
|
|
position = 1;
|
|
}
|
|
} else if (feature_value < smallest_values[2]) {
|
|
position = 2;
|
|
} else {
|
|
position = 3;
|
|
}
|
|
} else if (feature_value < smallest_values[5]) {
|
|
if (feature_value < smallest_values[4]) {
|
|
position = 4;
|
|
} else {
|
|
position = 5;
|
|
}
|
|
} else if (feature_value < smallest_values[6]) {
|
|
position = 6;
|
|
} else {
|
|
position = 7;
|
|
}
|
|
} else if (feature_value < smallest_values[15]) {
|
|
if (feature_value < smallest_values[11]) {
|
|
if (feature_value < smallest_values[9]) {
|
|
if (feature_value < smallest_values[8]) {
|
|
position = 8;
|
|
} else {
|
|
position = 9;
|
|
}
|
|
} else if (feature_value < smallest_values[10]) {
|
|
position = 10;
|
|
} else {
|
|
position = 11;
|
|
}
|
|
} else if (feature_value < smallest_values[13]) {
|
|
if (feature_value < smallest_values[12]) {
|
|
position = 12;
|
|
} else {
|
|
position = 13;
|
|
}
|
|
} else if (feature_value < smallest_values[14]) {
|
|
position = 14;
|
|
} else {
|
|
position = 15;
|
|
}
|
|
}
|
|
|
|
// If we have detected a new small value, insert it at the correct position
|
|
// and shift larger values up.
|
|
if (position > -1) {
|
|
for (i = 15; i > position; i--) {
|
|
smallest_values[i] = smallest_values[i - 1];
|
|
age[i] = age[i - 1];
|
|
}
|
|
smallest_values[position] = feature_value;
|
|
age[position] = 1;
|
|
}
|
|
|
|
// Get |current_median|.
|
|
if (self->frame_counter > 2) {
|
|
current_median = smallest_values[2];
|
|
} else if (self->frame_counter > 0) {
|
|
current_median = smallest_values[0];
|
|
}
|
|
|
|
// Smooth the median value.
|
|
if (self->frame_counter > 0) {
|
|
if (current_median < self->mean_value[channel]) {
|
|
alpha = kSmoothingDown; // 0.2 in Q15.
|
|
} else {
|
|
alpha = kSmoothingUp; // 0.99 in Q15.
|
|
}
|
|
}
|
|
tmp32 = WEBRTC_SPL_MUL_16_16(alpha + 1, self->mean_value[channel]);
|
|
tmp32 += WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, current_median);
|
|
tmp32 += 16384;
|
|
self->mean_value[channel] = (int16_t) (tmp32 >> 15);
|
|
|
|
return self->mean_value[channel];
|
|
}
|