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[pulseaudio] / src / modules / echo-cancel / adrian-aec.h
1 /* aec.h
2 *
3 * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
4 * All Rights Reserved.
5 * Author: Andre Adrian
6 *
7 * Acoustic Echo Cancellation Leaky NLMS-pw algorithm
8 *
9 * Version 0.3 filter created with www.dsptutor.freeuk.com
10 * Version 0.3.1 Allow change of stability parameter delta
11 * Version 0.4 Leaky Normalized LMS - pre whitening algorithm
12 */
13
14 #ifndef _AEC_H /* include only once */
15
16 #ifdef HAVE_CONFIG_H
17 #include <config.h>
18 #endif
19
20 #include <pulse/gccmacro.h>
21 #include <pulse/xmalloc.h>
22
23 #include <pulsecore/macro.h>
24
25 #define WIDEB 2
26
27 // use double if your CPU does software-emulation of float
28 #define REAL float
29
30 /* dB Values */
31 #define M0dB 1.0f
32 #define M3dB 0.71f
33 #define M6dB 0.50f
34 #define M9dB 0.35f
35 #define M12dB 0.25f
36 #define M18dB 0.125f
37 #define M24dB 0.063f
38
39 /* dB values for 16bit PCM */
40 /* MxdB_PCM = 32767 * 10 ^(x / 20) */
41 #define M10dB_PCM 10362.0f
42 #define M20dB_PCM 3277.0f
43 #define M25dB_PCM 1843.0f
44 #define M30dB_PCM 1026.0f
45 #define M35dB_PCM 583.0f
46 #define M40dB_PCM 328.0f
47 #define M45dB_PCM 184.0f
48 #define M50dB_PCM 104.0f
49 #define M55dB_PCM 58.0f
50 #define M60dB_PCM 33.0f
51 #define M65dB_PCM 18.0f
52 #define M70dB_PCM 10.0f
53 #define M75dB_PCM 6.0f
54 #define M80dB_PCM 3.0f
55 #define M85dB_PCM 2.0f
56 #define M90dB_PCM 1.0f
57
58 #define MAXPCM 32767.0f
59
60 /* Design constants (Change to fine tune the algorithms */
61
62 /* The following values are for hardware AEC and studio quality
63 * microphone */
64
65 /* NLMS filter length in taps (samples). A longer filter length gives
66 * better Echo Cancellation, but maybe slower convergence speed and
67 * needs more CPU power (Order of NLMS is linear) */
68 #define NLMS_LEN (100*WIDEB*8)
69
70 /* Vector w visualization length in taps (samples).
71 * Must match argv value for wdisplay.tcl */
72 #define DUMP_LEN (40*WIDEB*8)
73
74 /* minimum energy in xf. Range: M70dB_PCM to M50dB_PCM. Should be equal
75 * to microphone ambient Noise level */
76 #define NoiseFloor M55dB_PCM
77
78 /* Leaky hangover in taps.
79 */
80 #define Thold (60 * WIDEB * 8)
81
82 // Adrian soft decision DTD
83 // left point. X is ratio, Y is stepsize
84 #define STEPX1 1.0
85 #define STEPY1 1.0
86 // right point. STEPX2=2.0 is good double talk, 3.0 is good single talk.
87 #define STEPX2 2.5
88 #define STEPY2 0
89 #define ALPHAFAST (1.0f / 100.0f)
90 #define ALPHASLOW (1.0f / 20000.0f)
91
92
93
94 /* Ageing multiplier for LMS memory vector w */
95 #define Leaky 0.9999f
96
97 /* Double Talk Detector Speaker/Microphone Threshold. Range <=1
98 * Large value (M0dB) is good for Single-Talk Echo cancellation,
99 * small value (M12dB) is good for Doulbe-Talk AEC */
100 #define GeigelThreshold M6dB
101
102 /* for Non Linear Processor. Range >0 to 1. Large value (M0dB) is good
103 * for Double-Talk, small value (M12dB) is good for Single-Talk */
104 #define NLPAttenuation M12dB
105
106 /* Below this line there are no more design constants */
107
108 typedef struct IIR_HP IIR_HP;
109
110 /* Exponential Smoothing or IIR Infinite Impulse Response Filter */
111 struct IIR_HP {
112 REAL x;
113 };
114
115 static IIR_HP* IIR_HP_init(void) {
116 IIR_HP *i = pa_xnew(IIR_HP, 1);
117 i->x = 0.0f;
118 return i;
119 }
120
121 static REAL IIR_HP_highpass(IIR_HP *i, REAL in) {
122 const REAL a0 = 0.01f; /* controls Transfer Frequency */
123 /* Highpass = Signal - Lowpass. Lowpass = Exponential Smoothing */
124 i->x += a0 * (in - i->x);
125 return in - i->x;
126 }
127
128 typedef struct FIR_HP_300Hz FIR_HP_300Hz;
129
130 #if WIDEB==1
131 /* 17 taps FIR Finite Impulse Response filter
132 * Coefficients calculated with
133 * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
134 */
135 class FIR_HP_300Hz {
136 REAL z[18];
137
138 public:
139 FIR_HP_300Hz() {
140 memset(this, 0, sizeof(FIR_HP_300Hz));
141 }
142
143 REAL highpass(REAL in) {
144 const REAL a[18] = {
145 // Kaiser Window FIR Filter, Filter type: High pass
146 // Passband: 300.0 - 4000.0 Hz, Order: 16
147 // Transition band: 75.0 Hz, Stopband attenuation: 10.0 dB
148 -0.034870606, -0.039650206, -0.044063766, -0.04800318,
149 -0.051370874, -0.054082647, -0.056070227, -0.057283327,
150 0.8214126, -0.057283327, -0.056070227, -0.054082647,
151 -0.051370874, -0.04800318, -0.044063766, -0.039650206,
152 -0.034870606, 0.0
153 };
154 memmove(z + 1, z, 17 * sizeof(REAL));
155 z[0] = in;
156 REAL sum0 = 0.0, sum1 = 0.0;
157 int j;
158
159 for (j = 0; j < 18; j += 2) {
160 // optimize: partial loop unrolling
161 sum0 += a[j] * z[j];
162 sum1 += a[j + 1] * z[j + 1];
163 }
164 return sum0 + sum1;
165 }
166 };
167
168 #else
169
170 /* 35 taps FIR Finite Impulse Response filter
171 * Passband 150Hz to 4kHz for 8kHz sample rate, 300Hz to 8kHz for 16kHz
172 * sample rate.
173 * Coefficients calculated with
174 * www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
175 */
176 struct FIR_HP_300Hz {
177 REAL z[36];
178 };
179
180 static FIR_HP_300Hz* FIR_HP_300Hz_init(void) {
181 FIR_HP_300Hz *ret = pa_xnew(FIR_HP_300Hz, 1);
182 memset(ret, 0, sizeof(FIR_HP_300Hz));
183 return ret;
184 }
185
186 static REAL FIR_HP_300Hz_highpass(FIR_HP_300Hz *f, REAL in) {
187 REAL sum0 = 0.0, sum1 = 0.0;
188 int j;
189 const REAL a[36] = {
190 // Kaiser Window FIR Filter, Filter type: High pass
191 // Passband: 150.0 - 4000.0 Hz, Order: 34
192 // Transition band: 34.0 Hz, Stopband attenuation: 10.0 dB
193 -0.016165324, -0.017454365, -0.01871232, -0.019931411,
194 -0.021104068, -0.022222936, -0.02328091, -0.024271343,
195 -0.025187887, -0.02602462, -0.026776174, -0.027437767,
196 -0.028004972, -0.028474221, -0.028842418, -0.029107114,
197 -0.02926664, 0.8524841, -0.02926664, -0.029107114,
198 -0.028842418, -0.028474221, -0.028004972, -0.027437767,
199 -0.026776174, -0.02602462, -0.025187887, -0.024271343,
200 -0.02328091, -0.022222936, -0.021104068, -0.019931411,
201 -0.01871232, -0.017454365, -0.016165324, 0.0
202 };
203 memmove(f->z + 1, f->z, 35 * sizeof(REAL));
204 f->z[0] = in;
205
206 for (j = 0; j < 36; j += 2) {
207 // optimize: partial loop unrolling
208 sum0 += a[j] * f->z[j];
209 sum1 += a[j + 1] * f->z[j + 1];
210 }
211 return sum0 + sum1;
212 }
213 #endif
214
215 typedef struct IIR1 IIR1;
216
217 /* Recursive single pole IIR Infinite Impulse response High-pass filter
218 *
219 * Reference: The Scientist and Engineer's Guide to Digital Processing
220 *
221 * output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
222 *
223 * X = exp(-2.0 * pi * Fc)
224 * A0 = (1 + X) / 2
225 * A1 = -(1 + X) / 2
226 * B1 = X
227 * Fc = cutoff freq / sample rate
228 */
229 struct IIR1 {
230 REAL in0, out0;
231 REAL a0, a1, b1;
232 };
233
234 #if 0
235 IIR1() {
236 memset(this, 0, sizeof(IIR1));
237 }
238 #endif
239
240 static IIR1* IIR1_init(REAL Fc) {
241 IIR1 *i = pa_xnew(IIR1, 1);
242 i->b1 = expf(-2.0f * M_PI * Fc);
243 i->a0 = (1.0f + i->b1) / 2.0f;
244 i->a1 = -(i->a0);
245 i->in0 = 0.0f;
246 i->out0 = 0.0f;
247 return i;
248 }
249
250 static REAL IIR1_highpass(IIR1 *i, REAL in) {
251 REAL out = i->a0 * in + i->a1 * i->in0 + i->b1 * i->out0;
252 i->in0 = in;
253 i->out0 = out;
254 return out;
255 }
256
257
258 #if 0
259 /* Recursive two pole IIR Infinite Impulse Response filter
260 * Coefficients calculated with
261 * http://www.dsptutor.freeuk.com/IIRFilterDesign/IIRFiltDes102.html
262 */
263 class IIR2 {
264 REAL x[2], y[2];
265
266 public:
267 IIR2() {
268 memset(this, 0, sizeof(IIR2));
269 }
270
271 REAL highpass(REAL in) {
272 // Butterworth IIR filter, Filter type: HP
273 // Passband: 2000 - 4000.0 Hz, Order: 2
274 const REAL a[] = { 0.29289323f, -0.58578646f, 0.29289323f };
275 const REAL b[] = { 1.3007072E-16f, 0.17157288f };
276 REAL out =
277 a[0] * in + a[1] * x[0] + a[2] * x[1] - b[0] * y[0] - b[1] * y[1];
278
279 x[1] = x[0];
280 x[0] = in;
281 y[1] = y[0];
282 y[0] = out;
283 return out;
284 }
285 };
286 #endif
287
288
289 // Extention in taps to reduce mem copies
290 #define NLMS_EXT (10*8)
291
292 // block size in taps to optimize DTD calculation
293 #define DTD_LEN 16
294
295 typedef struct AEC AEC;
296
297 struct AEC {
298 // Time domain Filters
299 IIR_HP *acMic, *acSpk; // DC-level remove Highpass)
300 FIR_HP_300Hz *cutoff; // 150Hz cut-off Highpass
301 REAL gain; // Mic signal amplify
302 IIR1 *Fx, *Fe; // pre-whitening Highpass for x, e
303
304 // Adrian soft decision DTD (Double Talk Detector)
305 REAL dfast, xfast;
306 REAL dslow, xslow;
307
308 // NLMS-pw
309 REAL x[NLMS_LEN + NLMS_EXT]; // tap delayed loudspeaker signal
310 REAL xf[NLMS_LEN + NLMS_EXT]; // pre-whitening tap delayed signal
311 REAL w_arr[NLMS_LEN + (16 / sizeof(REAL))]; // tap weights
312 REAL *w; // this will be a 16-byte aligned pointer into w_arr
313 int j; // optimize: less memory copies
314 double dotp_xf_xf; // double to avoid loss of precision
315 float delta; // noise floor to stabilize NLMS
316
317 // AES
318 float aes_y2; // not in use!
319
320 // w vector visualization
321 REAL ws[DUMP_LEN]; // tap weights sums
322 int fdwdisplay; // TCP file descriptor
323 int dumpcnt; // wdisplay output counter
324
325 // variables are public for visualization
326 int hangover;
327 float stepsize;
328
329 // vfuncs that are picked based on processor features available
330 REAL (*dotp) (REAL[], REAL[]);
331 };
332
333 /* Double-Talk Detector
334 *
335 * in d: microphone sample (PCM as REALing point value)
336 * in x: loudspeaker sample (PCM as REALing point value)
337 * return: from 0 for doubletalk to 1.0 for single talk
338 */
339 static float AEC_dtd(AEC *a, REAL d, REAL x);
340
341 static void AEC_leaky(AEC *a);
342
343 /* Normalized Least Mean Square Algorithm pre-whitening (NLMS-pw)
344 * The LMS algorithm was developed by Bernard Widrow
345 * book: Haykin, Adaptive Filter Theory, 4. edition, Prentice Hall, 2002
346 *
347 * in d: microphone sample (16bit PCM value)
348 * in x_: loudspeaker sample (16bit PCM value)
349 * in stepsize: NLMS adaptation variable
350 * return: echo cancelled microphone sample
351 */
352 static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize);
353
354 AEC* AEC_init(int RATE, int have_vector);
355 void AEC_done(AEC *a);
356
357 /* Acoustic Echo Cancellation and Suppression of one sample
358 * in d: microphone signal with echo
359 * in x: loudspeaker signal
360 * return: echo cancelled microphone signal
361 */
362 int AEC_doAEC(AEC *a, int d_, int x_);
363
364 PA_GCC_UNUSED static float AEC_getambient(AEC *a) {
365 return a->dfast;
366 }
367 static void AEC_setambient(AEC *a, float Min_xf) {
368 a->dotp_xf_xf -= a->delta; // subtract old delta
369 a->delta = (NLMS_LEN-1) * Min_xf * Min_xf;
370 a->dotp_xf_xf += a->delta; // add new delta
371 }
372 PA_GCC_UNUSED static void AEC_setgain(AEC *a, float gain_) {
373 a->gain = gain_;
374 }
375 #if 0
376 void AEC_openwdisplay(AEC *a);
377 #endif
378 PA_GCC_UNUSED static void AEC_setaes(AEC *a, float aes_y2_) {
379 a->aes_y2 = aes_y2_;
380 }
381
382 #define _AEC_H
383 #endif