mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
177 lines
5.7 KiB
C
177 lines
5.7 KiB
C
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "common_audio/vad/vad_sp.h"
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#include "rtc_base/checks.h"
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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#include "common_audio/vad/vad_core.h"
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// Allpass filter coefficients, upper and lower, in Q13.
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// Upper: 0.64, Lower: 0.17.
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static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13.
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static const int16_t kSmoothingDown = 6553; // 0.2 in Q15.
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static const int16_t kSmoothingUp = 32439; // 0.99 in Q15.
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// TODO(bjornv): Move this function to vad_filterbank.c.
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// Downsampling filter based on splitting filter and allpass functions.
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void WebRtcVad_Downsampling(const int16_t* signal_in,
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int16_t* signal_out,
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int32_t* filter_state,
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size_t in_length) {
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int16_t tmp16_1 = 0, tmp16_2 = 0;
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int32_t tmp32_1 = filter_state[0];
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int32_t tmp32_2 = filter_state[1];
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size_t n = 0;
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// Downsampling by 2 gives half length.
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size_t half_length = (in_length >> 1);
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// Filter coefficients in Q13, filter state in Q0.
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for (n = 0; n < half_length; n++) {
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// All-pass filtering upper branch.
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tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
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((kAllPassCoefsQ13[0] * *signal_in) >> 14));
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*signal_out = tmp16_1;
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tmp32_1 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[0] * tmp16_1) >> 12);
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// All-pass filtering lower branch.
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tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
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((kAllPassCoefsQ13[1] * *signal_in) >> 14));
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*signal_out++ += tmp16_2;
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tmp32_2 = (int32_t)(*signal_in++) - ((kAllPassCoefsQ13[1] * tmp16_2) >> 12);
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}
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// Store the filter states.
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filter_state[0] = tmp32_1;
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filter_state[1] = tmp32_2;
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}
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// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
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// smallest values the last 100 frames. Then calculates and returns the median
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// of the five smallest values.
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int16_t WebRtcVad_FindMinimum(VadInstT* self,
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int16_t feature_value,
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int channel) {
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int i = 0, j = 0;
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int position = -1;
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// Offset to beginning of the 16 minimum values in memory.
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const int offset = (channel << 4);
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int16_t current_median = 1600;
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int16_t alpha = 0;
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int32_t tmp32 = 0;
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// Pointer to memory for the 16 minimum values and the age of each value of
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// the |channel|.
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int16_t* age = &self->index_vector[offset];
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int16_t* smallest_values = &self->low_value_vector[offset];
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RTC_DCHECK_LT(channel, kNumChannels);
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// Each value in |smallest_values| is getting 1 loop older. Update |age|, and
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// remove old values.
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for (i = 0; i < 16; i++) {
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if (age[i] != 100) {
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age[i]++;
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} else {
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// Too old value. Remove from memory and shift larger values downwards.
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for (j = i; j < 16; j++) {
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smallest_values[j] = smallest_values[j + 1];
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age[j] = age[j + 1];
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}
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age[15] = 101;
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smallest_values[15] = 10000;
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}
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}
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// Check if |feature_value| is smaller than any of the values in
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// |smallest_values|. If so, find the |position| where to insert the new value
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// (|feature_value|).
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if (feature_value < smallest_values[7]) {
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if (feature_value < smallest_values[3]) {
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if (feature_value < smallest_values[1]) {
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if (feature_value < smallest_values[0]) {
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position = 0;
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} else {
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position = 1;
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}
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} else if (feature_value < smallest_values[2]) {
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position = 2;
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} else {
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position = 3;
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}
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} else if (feature_value < smallest_values[5]) {
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if (feature_value < smallest_values[4]) {
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position = 4;
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} else {
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position = 5;
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}
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} else if (feature_value < smallest_values[6]) {
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position = 6;
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} else {
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position = 7;
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}
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} else if (feature_value < smallest_values[15]) {
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if (feature_value < smallest_values[11]) {
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if (feature_value < smallest_values[9]) {
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if (feature_value < smallest_values[8]) {
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position = 8;
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} else {
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position = 9;
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}
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} else if (feature_value < smallest_values[10]) {
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position = 10;
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} else {
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position = 11;
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}
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} else if (feature_value < smallest_values[13]) {
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if (feature_value < smallest_values[12]) {
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position = 12;
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} else {
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position = 13;
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}
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} else if (feature_value < smallest_values[14]) {
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position = 14;
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} else {
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position = 15;
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}
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}
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// If we have detected a new small value, insert it at the correct position
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// and shift larger values up.
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if (position > -1) {
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for (i = 15; i > position; i--) {
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smallest_values[i] = smallest_values[i - 1];
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age[i] = age[i - 1];
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}
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smallest_values[position] = feature_value;
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age[position] = 1;
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}
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// Get |current_median|.
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if (self->frame_counter > 2) {
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current_median = smallest_values[2];
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} else if (self->frame_counter > 0) {
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current_median = smallest_values[0];
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}
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// Smooth the median value.
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if (self->frame_counter > 0) {
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if (current_median < self->mean_value[channel]) {
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alpha = kSmoothingDown; // 0.2 in Q15.
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} else {
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alpha = kSmoothingUp; // 0.99 in Q15.
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}
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}
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tmp32 = (alpha + 1) * self->mean_value[channel];
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tmp32 += (WEBRTC_SPL_WORD16_MAX - alpha) * current_median;
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tmp32 += 16384;
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self->mean_value[channel] = (int16_t) (tmp32 >> 15);
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return self->mean_value[channel];
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}
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