/* 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 #endif #include #include #include #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 Doulbe-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 // Extention 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); /* 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