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380 lines
10 KiB
C++
380 lines
10 KiB
C++
/* aec.h
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*
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* Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
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* All Rights Reserved.
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* Author: Andre Adrian
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*
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* Acoustic Echo Cancellation Leaky NLMS-pw algorithm
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*
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* Version 0.3 filter created with www.dsptutor.freeuk.com
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* Version 0.3.1 Allow change of stability parameter delta
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* Version 0.4 Leaky Normalized LMS - pre whitening algorithm
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*/
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#ifndef _AEC_H /* include only once */
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <pulsecore/macro.h>
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#include <pulse/xmalloc.h>
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#define WIDEB 2
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// use double if your CPU does software-emulation of float
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#define REAL float
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/* dB Values */
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#define M0dB 1.0f
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#define M3dB 0.71f
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#define M6dB 0.50f
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#define M9dB 0.35f
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#define M12dB 0.25f
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#define M18dB 0.125f
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#define M24dB 0.063f
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/* dB values for 16bit PCM */
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/* MxdB_PCM = 32767 * 10 ^(x / 20) */
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#define M10dB_PCM 10362.0f
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#define M20dB_PCM 3277.0f
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#define M25dB_PCM 1843.0f
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#define M30dB_PCM 1026.0f
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#define M35dB_PCM 583.0f
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#define M40dB_PCM 328.0f
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#define M45dB_PCM 184.0f
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#define M50dB_PCM 104.0f
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#define M55dB_PCM 58.0f
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#define M60dB_PCM 33.0f
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#define M65dB_PCM 18.0f
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#define M70dB_PCM 10.0f
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#define M75dB_PCM 6.0f
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#define M80dB_PCM 3.0f
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#define M85dB_PCM 2.0f
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#define M90dB_PCM 1.0f
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#define MAXPCM 32767.0f
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/* Design constants (Change to fine tune the algorithms */
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/* The following values are for hardware AEC and studio quality
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* microphone */
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/* NLMS filter length in taps (samples). A longer filter length gives
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* better Echo Cancellation, but maybe slower convergence speed and
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* needs more CPU power (Order of NLMS is linear) */
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#define NLMS_LEN (100*WIDEB*8)
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/* Vector w visualization length in taps (samples).
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* Must match argv value for wdisplay.tcl */
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#define DUMP_LEN (40*WIDEB*8)
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/* minimum energy in xf. Range: M70dB_PCM to M50dB_PCM. Should be equal
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* to microphone ambient Noise level */
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#define NoiseFloor M55dB_PCM
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/* Leaky hangover in taps.
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*/
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#define Thold (60 * WIDEB * 8)
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// Adrian soft decision DTD
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// left point. X is ratio, Y is stepsize
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#define STEPX1 1.0
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#define STEPY1 1.0
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// right point. STEPX2=2.0 is good double talk, 3.0 is good single talk.
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#define STEPX2 2.5
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#define STEPY2 0
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#define ALPHAFAST (1.0f / 100.0f)
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#define ALPHASLOW (1.0f / 20000.0f)
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/* Ageing multiplier for LMS memory vector w */
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#define Leaky 0.9999f
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/* Double Talk Detector Speaker/Microphone Threshold. Range <=1
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* Large value (M0dB) is good for Single-Talk Echo cancellation,
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* small value (M12dB) is good for Doulbe-Talk AEC */
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#define GeigelThreshold M6dB
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/* for Non Linear Processor. Range >0 to 1. Large value (M0dB) is good
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* for Double-Talk, small value (M12dB) is good for Single-Talk */
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#define NLPAttenuation M12dB
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/* Below this line there are no more design constants */
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typedef struct IIR_HP IIR_HP;
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/* Exponential Smoothing or IIR Infinite Impulse Response Filter */
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struct IIR_HP {
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REAL x;
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};
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static IIR_HP* IIR_HP_init(void) {
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IIR_HP *i = pa_xnew(IIR_HP, 1);
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i->x = 0.0f;
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return i;
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}
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static REAL IIR_HP_highpass(IIR_HP *i, REAL in) {
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const REAL a0 = 0.01f; /* controls Transfer Frequency */
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/* Highpass = Signal - Lowpass. Lowpass = Exponential Smoothing */
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i->x += a0 * (in - i->x);
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return in - i->x;
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};
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typedef struct FIR_HP_300Hz FIR_HP_300Hz;
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#if WIDEB==1
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/* 17 taps FIR Finite Impulse Response filter
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* Coefficients calculated with
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* www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
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*/
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class FIR_HP_300Hz {
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REAL z[18];
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public:
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FIR_HP_300Hz() {
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memset(this, 0, sizeof(FIR_HP_300Hz));
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}
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REAL highpass(REAL in) {
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const REAL a[18] = {
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// Kaiser Window FIR Filter, Filter type: High pass
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// Passband: 300.0 - 4000.0 Hz, Order: 16
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// Transition band: 75.0 Hz, Stopband attenuation: 10.0 dB
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-0.034870606, -0.039650206, -0.044063766, -0.04800318,
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-0.051370874, -0.054082647, -0.056070227, -0.057283327,
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0.8214126, -0.057283327, -0.056070227, -0.054082647,
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-0.051370874, -0.04800318, -0.044063766, -0.039650206,
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-0.034870606, 0.0
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};
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memmove(z + 1, z, 17 * sizeof(REAL));
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z[0] = in;
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REAL sum0 = 0.0, sum1 = 0.0;
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int j;
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for (j = 0; j < 18; j += 2) {
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// optimize: partial loop unrolling
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sum0 += a[j] * z[j];
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sum1 += a[j + 1] * z[j + 1];
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}
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return sum0 + sum1;
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}
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};
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#else
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/* 35 taps FIR Finite Impulse Response filter
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* Passband 150Hz to 4kHz for 8kHz sample rate, 300Hz to 8kHz for 16kHz
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* sample rate.
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* Coefficients calculated with
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* www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
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*/
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struct FIR_HP_300Hz {
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REAL z[36];
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};
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static FIR_HP_300Hz* FIR_HP_300Hz_init(void) {
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FIR_HP_300Hz *ret = pa_xnew(FIR_HP_300Hz, 1);
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memset(ret, 0, sizeof(FIR_HP_300Hz));
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return ret;
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}
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static REAL FIR_HP_300Hz_highpass(FIR_HP_300Hz *f, REAL in) {
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REAL sum0 = 0.0, sum1 = 0.0;
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int j;
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const REAL a[36] = {
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// Kaiser Window FIR Filter, Filter type: High pass
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// Passband: 150.0 - 4000.0 Hz, Order: 34
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// Transition band: 34.0 Hz, Stopband attenuation: 10.0 dB
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-0.016165324, -0.017454365, -0.01871232, -0.019931411,
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-0.021104068, -0.022222936, -0.02328091, -0.024271343,
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-0.025187887, -0.02602462, -0.026776174, -0.027437767,
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-0.028004972, -0.028474221, -0.028842418, -0.029107114,
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-0.02926664, 0.8524841, -0.02926664, -0.029107114,
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-0.028842418, -0.028474221, -0.028004972, -0.027437767,
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-0.026776174, -0.02602462, -0.025187887, -0.024271343,
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-0.02328091, -0.022222936, -0.021104068, -0.019931411,
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-0.01871232, -0.017454365, -0.016165324, 0.0
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};
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memmove(f->z + 1, f->z, 35 * sizeof(REAL));
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f->z[0] = in;
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for (j = 0; j < 36; j += 2) {
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// optimize: partial loop unrolling
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sum0 += a[j] * f->z[j];
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sum1 += a[j + 1] * f->z[j + 1];
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}
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return sum0 + sum1;
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}
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#endif
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typedef struct IIR1 IIR1;
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/* Recursive single pole IIR Infinite Impulse response High-pass filter
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*
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* Reference: The Scientist and Engineer's Guide to Digital Processing
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*
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* output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
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*
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* X = exp(-2.0 * pi * Fc)
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* A0 = (1 + X) / 2
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* A1 = -(1 + X) / 2
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* B1 = X
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* Fc = cutoff freq / sample rate
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*/
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struct IIR1 {
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REAL in0, out0;
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REAL a0, a1, b1;
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};
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#if 0
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IIR1() {
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memset(this, 0, sizeof(IIR1));
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}
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#endif
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static IIR1* IIR1_init(REAL Fc) {
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IIR1 *i = pa_xnew(IIR1, 1);
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i->b1 = expf(-2.0f * M_PI * Fc);
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i->a0 = (1.0f + i->b1) / 2.0f;
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i->a1 = -(i->a0);
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i->in0 = 0.0f;
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i->out0 = 0.0f;
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return i;
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}
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static REAL IIR1_highpass(IIR1 *i, REAL in) {
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REAL out = i->a0 * in + i->a1 * i->in0 + i->b1 * i->out0;
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i->in0 = in;
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i->out0 = out;
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return out;
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}
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#if 0
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/* Recursive two pole IIR Infinite Impulse Response filter
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* Coefficients calculated with
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* http://www.dsptutor.freeuk.com/IIRFilterDesign/IIRFiltDes102.html
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*/
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class IIR2 {
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REAL x[2], y[2];
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public:
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IIR2() {
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memset(this, 0, sizeof(IIR2));
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}
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REAL highpass(REAL in) {
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// Butterworth IIR filter, Filter type: HP
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// Passband: 2000 - 4000.0 Hz, Order: 2
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const REAL a[] = { 0.29289323f, -0.58578646f, 0.29289323f };
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const REAL b[] = { 1.3007072E-16f, 0.17157288f };
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REAL out =
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a[0] * in + a[1] * x[0] + a[2] * x[1] - b[0] * y[0] - b[1] * y[1];
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x[1] = x[0];
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x[0] = in;
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y[1] = y[0];
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y[0] = out;
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return out;
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}
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};
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#endif
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// Extention in taps to reduce mem copies
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#define NLMS_EXT (10*8)
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// block size in taps to optimize DTD calculation
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#define DTD_LEN 16
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typedef struct AEC AEC;
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struct AEC {
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// Time domain Filters
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IIR_HP *acMic, *acSpk; // DC-level remove Highpass)
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FIR_HP_300Hz *cutoff; // 150Hz cut-off Highpass
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REAL gain; // Mic signal amplify
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IIR1 *Fx, *Fe; // pre-whitening Highpass for x, e
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// Adrian soft decision DTD (Double Talk Detector)
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REAL dfast, xfast;
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REAL dslow, xslow;
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// NLMS-pw
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REAL x[NLMS_LEN + NLMS_EXT]; // tap delayed loudspeaker signal
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REAL xf[NLMS_LEN + NLMS_EXT]; // pre-whitening tap delayed signal
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REAL w_arr[NLMS_LEN + (16 / sizeof(REAL))]; // tap weights
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REAL *w; // this will be a 16-byte aligned pointer into w_arr
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int j; // optimize: less memory copies
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double dotp_xf_xf; // double to avoid loss of precision
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float delta; // noise floor to stabilize NLMS
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// AES
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float aes_y2; // not in use!
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// w vector visualization
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REAL ws[DUMP_LEN]; // tap weights sums
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int fdwdisplay; // TCP file descriptor
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int dumpcnt; // wdisplay output counter
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// variables are public for visualization
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int hangover;
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float stepsize;
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// vfuncs that are picked based on processor features available
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REAL (*dotp) (REAL[], REAL[]);
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};
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/* Double-Talk Detector
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*
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* in d: microphone sample (PCM as REALing point value)
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* in x: loudspeaker sample (PCM as REALing point value)
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* return: from 0 for doubletalk to 1.0 for single talk
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*/
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static float AEC_dtd(AEC *a, REAL d, REAL x);
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static void AEC_leaky(AEC *a);
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/* Normalized Least Mean Square Algorithm pre-whitening (NLMS-pw)
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* The LMS algorithm was developed by Bernard Widrow
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* book: Haykin, Adaptive Filter Theory, 4. edition, Prentice Hall, 2002
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*
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* in d: microphone sample (16bit PCM value)
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* in x_: loudspeaker sample (16bit PCM value)
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* in stepsize: NLMS adaptation variable
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* return: echo cancelled microphone sample
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*/
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static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize);
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AEC* AEC_init(int RATE, int have_vector);
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/* Acoustic Echo Cancellation and Suppression of one sample
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* in d: microphone signal with echo
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* in x: loudspeaker signal
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* return: echo cancelled microphone signal
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*/
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int AEC_doAEC(AEC *a, int d_, int x_);
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PA_GCC_UNUSED static float AEC_getambient(AEC *a) {
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return a->dfast;
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};
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static void AEC_setambient(AEC *a, float Min_xf) {
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a->dotp_xf_xf -= a->delta; // subtract old delta
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a->delta = (NLMS_LEN-1) * Min_xf * Min_xf;
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a->dotp_xf_xf += a->delta; // add new delta
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};
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PA_GCC_UNUSED static void AEC_setgain(AEC *a, float gain_) {
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a->gain = gain_;
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};
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#if 0
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void AEC_openwdisplay(AEC *a);
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#endif
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PA_GCC_UNUSED static void AEC_setaes(AEC *a, float aes_y2_) {
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a->aes_y2 = aes_y2_;
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};
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#define _AEC_H
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#endif
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