mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
90 lines
3.0 KiB
C
90 lines
3.0 KiB
C
/*
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* Copyright (c) 2011 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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/******************************************************************
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iLBC Speech Coder ANSI-C Source Code
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WebRtcIlbcfix_HpOutput.c
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******************************************************************/
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#include "defines.h"
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/*----------------------------------------------------------------*
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* high-pass filter of output and *2 with saturation
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*---------------------------------------------------------------*/
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void WebRtcIlbcfix_HpOutput(
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int16_t *signal, /* (i/o) signal vector */
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int16_t *ba, /* (i) B- and A-coefficients (2:nd order)
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{b[0] b[1] b[2] -a[1] -a[2]} a[0]
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is assumed to be 1.0 */
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int16_t *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
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yhi[n-2] ylow[n-2] */
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int16_t *x, /* (i/o) Filter state x[n-1] x[n-2] */
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int16_t len) /* (i) Number of samples to filter */
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{
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int i;
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int32_t tmpW32;
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int32_t tmpW32b;
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for (i=0; i<len; i++) {
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/*
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y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
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+ (-a[1])*y[i-1] + (-a[2])*y[i-2];
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*/
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tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */
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tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */
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tmpW32 = (tmpW32>>15);
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tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */
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tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */
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tmpW32 = (tmpW32<<1);
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tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] */
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tmpW32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); /* b[1]*x[i-1] */
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tmpW32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); /* b[2]*x[i-2] */
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/* Update state (input part) */
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x[1] = x[0];
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x[0] = signal[i];
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/* Rounding in Q(12-1), i.e. add 2^10 */
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tmpW32b = tmpW32 + 1024;
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/* Saturate (to 2^26) so that the HP filtered signal does not overflow */
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tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
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/* Convert back to Q0 and multiply with 2 */
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signal[i] = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11);
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/* Update state (filtered part) */
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y[2] = y[0];
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y[3] = y[1];
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/* upshift tmpW32 by 3 with saturation */
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if (tmpW32>268435455) {
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tmpW32 = WEBRTC_SPL_WORD32_MAX;
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} else if (tmpW32<-268435456) {
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tmpW32 = WEBRTC_SPL_WORD32_MIN;
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} else {
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tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
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}
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y[0] = (int16_t)(tmpW32 >> 16);
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y[1] = (int16_t)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((int32_t)y[0], 16))>>1);
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}
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return;
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}
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