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Diffstat (limited to 'src/modules/echo-cancel/adrian-aec.h')
| -rw-r--r-- | src/modules/echo-cancel/adrian-aec.h | 383 |
1 files changed, 383 insertions, 0 deletions
diff --git a/src/modules/echo-cancel/adrian-aec.h b/src/modules/echo-cancel/adrian-aec.h new file mode 100644 index 0000000..3a31fd8 --- /dev/null +++ b/src/modules/echo-cancel/adrian-aec.h @@ -0,0 +1,383 @@ +/* aec.h + * + * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005). + * All Rights Reserved. + * Author: Andre Adrian + * + * Acoustic Echo Cancellation Leaky NLMS-pw algorithm + * + * Version 0.3 filter created with www.dsptutor.freeuk.com + * Version 0.3.1 Allow change of stability parameter delta + * Version 0.4 Leaky Normalized LMS - pre whitening algorithm + */ + +#ifndef _AEC_H /* include only once */ + +#ifdef HAVE_CONFIG_H +#include <config.h> +#endif + +#include <pulse/gccmacro.h> +#include <pulse/xmalloc.h> + +#include <pulsecore/macro.h> + +#define WIDEB 2 + +// use double if your CPU does software-emulation of float +#define REAL float + +/* dB Values */ +#define M0dB 1.0f +#define M3dB 0.71f +#define M6dB 0.50f +#define M9dB 0.35f +#define M12dB 0.25f +#define M18dB 0.125f +#define M24dB 0.063f + +/* dB values for 16bit PCM */ +/* MxdB_PCM = 32767 * 10 ^(x / 20) */ +#define M10dB_PCM 10362.0f +#define M20dB_PCM 3277.0f +#define M25dB_PCM 1843.0f +#define M30dB_PCM 1026.0f +#define M35dB_PCM 583.0f +#define M40dB_PCM 328.0f +#define M45dB_PCM 184.0f +#define M50dB_PCM 104.0f +#define M55dB_PCM 58.0f +#define M60dB_PCM 33.0f +#define M65dB_PCM 18.0f +#define M70dB_PCM 10.0f +#define M75dB_PCM 6.0f +#define M80dB_PCM 3.0f +#define M85dB_PCM 2.0f +#define M90dB_PCM 1.0f + +#define MAXPCM 32767.0f + +/* Design constants (Change to fine tune the algorithms */ + +/* The following values are for hardware AEC and studio quality + * microphone */ + +/* NLMS filter length in taps (samples). A longer filter length gives + * better Echo Cancellation, but maybe slower convergence speed and + * needs more CPU power (Order of NLMS is linear) */ +#define NLMS_LEN (100*WIDEB*8) + +/* Vector w visualization length in taps (samples). + * Must match argv value for wdisplay.tcl */ +#define DUMP_LEN (40*WIDEB*8) + +/* minimum energy in xf. Range: M70dB_PCM to M50dB_PCM. Should be equal + * to microphone ambient Noise level */ +#define NoiseFloor M55dB_PCM + +/* Leaky hangover in taps. + */ +#define Thold (60 * WIDEB * 8) + +// Adrian soft decision DTD +// left point. X is ratio, Y is stepsize +#define STEPX1 1.0 +#define STEPY1 1.0 +// right point. STEPX2=2.0 is good double talk, 3.0 is good single talk. +#define STEPX2 2.5 +#define STEPY2 0 +#define ALPHAFAST (1.0f / 100.0f) +#define ALPHASLOW (1.0f / 20000.0f) + + + +/* Ageing multiplier for LMS memory vector w */ +#define Leaky 0.9999f + +/* Double Talk Detector Speaker/Microphone Threshold. Range <=1 + * Large value (M0dB) is good for Single-Talk Echo cancellation, + * small value (M12dB) is good for Double-Talk AEC */ +#define GeigelThreshold M6dB + +/* for Non Linear Processor. Range >0 to 1. Large value (M0dB) is good + * for Double-Talk, small value (M12dB) is good for Single-Talk */ +#define NLPAttenuation M12dB + +/* Below this line there are no more design constants */ + +typedef struct IIR_HP IIR_HP; + +/* Exponential Smoothing or IIR Infinite Impulse Response Filter */ +struct IIR_HP { + REAL x; +}; + +static IIR_HP* IIR_HP_init(void) { + IIR_HP *i = pa_xnew(IIR_HP, 1); + i->x = 0.0f; + return i; + } + +static REAL IIR_HP_highpass(IIR_HP *i, REAL in) { + const REAL a0 = 0.01f; /* controls Transfer Frequency */ + /* Highpass = Signal - Lowpass. Lowpass = Exponential Smoothing */ + i->x += a0 * (in - i->x); + return in - i->x; + } + +typedef struct FIR_HP_300Hz FIR_HP_300Hz; + +#if WIDEB==1 +/* 17 taps FIR Finite Impulse Response filter + * Coefficients calculated with + * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html + */ +class FIR_HP_300Hz { + REAL z[18]; + +public: + FIR_HP_300Hz() { + memset(this, 0, sizeof(FIR_HP_300Hz)); + } + + REAL highpass(REAL in) { + const REAL a[18] = { + // Kaiser Window FIR Filter, Filter type: High pass + // Passband: 300.0 - 4000.0 Hz, Order: 16 + // Transition band: 75.0 Hz, Stopband attenuation: 10.0 dB + -0.034870606, -0.039650206, -0.044063766, -0.04800318, + -0.051370874, -0.054082647, -0.056070227, -0.057283327, + 0.8214126, -0.057283327, -0.056070227, -0.054082647, + -0.051370874, -0.04800318, -0.044063766, -0.039650206, + -0.034870606, 0.0 + }; + memmove(z + 1, z, 17 * sizeof(REAL)); + z[0] = in; + REAL sum0 = 0.0, sum1 = 0.0; + int j; + + for (j = 0; j < 18; j += 2) { + // optimize: partial loop unrolling + sum0 += a[j] * z[j]; + sum1 += a[j + 1] * z[j + 1]; + } + return sum0 + sum1; + } +}; + +#else + +/* 35 taps FIR Finite Impulse Response filter + * Passband 150Hz to 4kHz for 8kHz sample rate, 300Hz to 8kHz for 16kHz + * sample rate. + * Coefficients calculated with + * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html + */ +struct FIR_HP_300Hz { + REAL z[36]; +}; + +static FIR_HP_300Hz* FIR_HP_300Hz_init(void) { + FIR_HP_300Hz *ret = pa_xnew(FIR_HP_300Hz, 1); + memset(ret, 0, sizeof(FIR_HP_300Hz)); + return ret; + } + +static REAL FIR_HP_300Hz_highpass(FIR_HP_300Hz *f, REAL in) { + REAL sum0 = 0.0, sum1 = 0.0; + int j; + const REAL a[36] = { + // Kaiser Window FIR Filter, Filter type: High pass + // Passband: 150.0 - 4000.0 Hz, Order: 34 + // Transition band: 34.0 Hz, Stopband attenuation: 10.0 dB + -0.016165324, -0.017454365, -0.01871232, -0.019931411, + -0.021104068, -0.022222936, -0.02328091, -0.024271343, + -0.025187887, -0.02602462, -0.026776174, -0.027437767, + -0.028004972, -0.028474221, -0.028842418, -0.029107114, + -0.02926664, 0.8524841, -0.02926664, -0.029107114, + -0.028842418, -0.028474221, -0.028004972, -0.027437767, + -0.026776174, -0.02602462, -0.025187887, -0.024271343, + -0.02328091, -0.022222936, -0.021104068, -0.019931411, + -0.01871232, -0.017454365, -0.016165324, 0.0 + }; + memmove(f->z + 1, f->z, 35 * sizeof(REAL)); + f->z[0] = in; + + for (j = 0; j < 36; j += 2) { + // optimize: partial loop unrolling + sum0 += a[j] * f->z[j]; + sum1 += a[j + 1] * f->z[j + 1]; + } + return sum0 + sum1; + } +#endif + +typedef struct IIR1 IIR1; + +/* Recursive single pole IIR Infinite Impulse response High-pass filter + * + * Reference: The Scientist and Engineer's Guide to Digital Processing + * + * output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1] + * + * X = exp(-2.0 * pi * Fc) + * A0 = (1 + X) / 2 + * A1 = -(1 + X) / 2 + * B1 = X + * Fc = cutoff freq / sample rate + */ +struct IIR1 { + REAL in0, out0; + REAL a0, a1, b1; +}; + +#if 0 + IIR1() { + memset(this, 0, sizeof(IIR1)); + } +#endif + +static IIR1* IIR1_init(REAL Fc) { + IIR1 *i = pa_xnew(IIR1, 1); + i->b1 = expf(-2.0f * M_PI * Fc); + i->a0 = (1.0f + i->b1) / 2.0f; + i->a1 = -(i->a0); + i->in0 = 0.0f; + i->out0 = 0.0f; + return i; + } + +static REAL IIR1_highpass(IIR1 *i, REAL in) { + REAL out = i->a0 * in + i->a1 * i->in0 + i->b1 * i->out0; + i->in0 = in; + i->out0 = out; + return out; + } + + +#if 0 +/* Recursive two pole IIR Infinite Impulse Response filter + * Coefficients calculated with + * http://www.dsptutor.freeuk.com/IIRFilterDesign/IIRFiltDes102.html + */ +class IIR2 { + REAL x[2], y[2]; + +public: + IIR2() { + memset(this, 0, sizeof(IIR2)); + } + + REAL highpass(REAL in) { + // Butterworth IIR filter, Filter type: HP + // Passband: 2000 - 4000.0 Hz, Order: 2 + const REAL a[] = { 0.29289323f, -0.58578646f, 0.29289323f }; + const REAL b[] = { 1.3007072E-16f, 0.17157288f }; + REAL out = + a[0] * in + a[1] * x[0] + a[2] * x[1] - b[0] * y[0] - b[1] * y[1]; + + x[1] = x[0]; + x[0] = in; + y[1] = y[0]; + y[0] = out; + return out; + } +}; +#endif + + +// Extension in taps to reduce mem copies +#define NLMS_EXT (10*8) + +// block size in taps to optimize DTD calculation +#define DTD_LEN 16 + +typedef struct AEC AEC; + +struct AEC { + // Time domain Filters + IIR_HP *acMic, *acSpk; // DC-level remove Highpass) + FIR_HP_300Hz *cutoff; // 150Hz cut-off Highpass + REAL gain; // Mic signal amplify + IIR1 *Fx, *Fe; // pre-whitening Highpass for x, e + + // Adrian soft decision DTD (Double Talk Detector) + REAL dfast, xfast; + REAL dslow, xslow; + + // NLMS-pw + REAL x[NLMS_LEN + NLMS_EXT]; // tap delayed loudspeaker signal + REAL xf[NLMS_LEN + NLMS_EXT]; // pre-whitening tap delayed signal + REAL w_arr[NLMS_LEN + (16 / sizeof(REAL))]; // tap weights + REAL *w; // this will be a 16-byte aligned pointer into w_arr + int j; // optimize: less memory copies + double dotp_xf_xf; // double to avoid loss of precision + float delta; // noise floor to stabilize NLMS + + // AES + float aes_y2; // not in use! + + // w vector visualization + REAL ws[DUMP_LEN]; // tap weights sums + int fdwdisplay; // TCP file descriptor + int dumpcnt; // wdisplay output counter + + // variables are public for visualization + int hangover; + float stepsize; + + // vfuncs that are picked based on processor features available + REAL (*dotp) (REAL[], REAL[]); +}; + +/* Double-Talk Detector + * + * in d: microphone sample (PCM as REALing point value) + * in x: loudspeaker sample (PCM as REALing point value) + * return: from 0 for doubletalk to 1.0 for single talk + */ +static float AEC_dtd(AEC *a, REAL d, REAL x); + +static void AEC_leaky(AEC *a); + +/* Normalized Least Mean Square Algorithm pre-whitening (NLMS-pw) + * The LMS algorithm was developed by Bernard Widrow + * book: Haykin, Adaptive Filter Theory, 4. edition, Prentice Hall, 2002 + * + * in d: microphone sample (16bit PCM value) + * in x_: loudspeaker sample (16bit PCM value) + * in stepsize: NLMS adaptation variable + * return: echo cancelled microphone sample + */ +static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize); + +AEC* AEC_init(int RATE, int have_vector); +void AEC_done(AEC *a); + +/* Acoustic Echo Cancellation and Suppression of one sample + * in d: microphone signal with echo + * in x: loudspeaker signal + * return: echo cancelled microphone signal + */ + int AEC_doAEC(AEC *a, int d_, int x_); + +PA_GCC_UNUSED static float AEC_getambient(AEC *a) { + return a->dfast; + } +static void AEC_setambient(AEC *a, float Min_xf) { + a->dotp_xf_xf -= a->delta; // subtract old delta + a->delta = (NLMS_LEN-1) * Min_xf * Min_xf; + a->dotp_xf_xf += a->delta; // add new delta + } +PA_GCC_UNUSED static void AEC_setgain(AEC *a, float gain_) { + a->gain = gain_; + } +#if 0 + void AEC_openwdisplay(AEC *a); +#endif +PA_GCC_UNUSED static void AEC_setaes(AEC *a, float aes_y2_) { + a->aes_y2 = aes_y2_; + } + +#define _AEC_H +#endif |