summaryrefslogtreecommitdiff
path: root/src/modules
diff options
context:
space:
mode:
Diffstat (limited to 'src/modules')
-rw-r--r--src/modules/echo-cancel/adrian-aec.c233
-rw-r--r--src/modules/echo-cancel/adrian-aec.h370
-rw-r--r--src/modules/echo-cancel/adrian-license.txt17
-rw-r--r--src/modules/echo-cancel/adrian.c121
-rw-r--r--src/modules/echo-cancel/adrian.h31
-rw-r--r--src/modules/echo-cancel/echo-cancel.h14
-rw-r--r--src/modules/echo-cancel/module-echo-cancel.c8
7 files changed, 794 insertions, 0 deletions
diff --git a/src/modules/echo-cancel/adrian-aec.c b/src/modules/echo-cancel/adrian-aec.c
new file mode 100644
index 00000000..69107c75
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-aec.c
@@ -0,0 +1,233 @@
+/* aec.cpp
+ *
+ * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
+ * All Rights Reserved.
+ *
+ * Acoustic Echo Cancellation 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
+ */
+
+#include <math.h>
+#include <string.h>
+
+#include <pulse/xmalloc.h>
+
+#include "adrian-aec.h"
+
+/* Vector Dot Product */
+static REAL dotp(REAL a[], REAL b[])
+{
+ REAL sum0 = 0.0, sum1 = 0.0;
+ int j;
+
+ for (j = 0; j < NLMS_LEN; j += 2) {
+ // optimize: partial loop unrolling
+ sum0 += a[j] * b[j];
+ sum1 += a[j + 1] * b[j + 1];
+ }
+ return sum0 + sum1;
+}
+
+
+AEC* AEC_init(int RATE)
+{
+ AEC *a = pa_xnew(AEC, 1);
+ a->hangover = 0;
+ memset(a->x, 0, sizeof(a->x));
+ memset(a->xf, 0, sizeof(a->xf));
+ memset(a->w, 0, sizeof(a->w));
+ a->j = NLMS_EXT;
+ a->delta = 0.0f;
+ AEC_setambient(a, NoiseFloor);
+ a->dfast = a->dslow = M75dB_PCM;
+ a->xfast = a->xslow = M80dB_PCM;
+ a->gain = 1.0f;
+ a->Fx = IIR1_init(2000.0f/RATE);
+ a->Fe = IIR1_init(2000.0f/RATE);
+ a->cutoff = FIR_HP_300Hz_init();
+ a->acMic = IIR_HP_init();
+ a->acSpk = IIR_HP_init();
+
+ a->aes_y2 = M0dB;
+
+ a->fdwdisplay = -1;
+ a->dumpcnt = 0;
+ memset(a->ws, 0, sizeof(a->ws));
+
+ return a;
+}
+
+// Adrian soft decision DTD
+// (Dual Average Near-End to Far-End signal Ratio DTD)
+// This algorithm uses exponential smoothing with differnt
+// ageing parameters to get fast and slow near-end and far-end
+// signal averages. The ratio of NFRs term
+// (dfast / xfast) / (dslow / xslow) is used to compute the stepsize
+// A ratio value of 2.5 is mapped to stepsize 0, a ratio of 0 is
+// mapped to 1.0 with a limited linear function.
+static float AEC_dtd(AEC *a, REAL d, REAL x)
+{
+ float stepsize;
+ float ratio, M;
+
+ // fast near-end and far-end average
+ a->dfast += ALPHAFAST * (fabsf(d) - a->dfast);
+ a->xfast += ALPHAFAST * (fabsf(x) - a->xfast);
+
+ // slow near-end and far-end average
+ a->dslow += ALPHASLOW * (fabsf(d) - a->dslow);
+ a->xslow += ALPHASLOW * (fabsf(x) - a->xslow);
+
+ if (a->xfast < M70dB_PCM) {
+ return 0.0; // no Spk signal
+ }
+
+ if (a->dfast < M70dB_PCM) {
+ return 0.0; // no Mic signal
+ }
+
+ // ratio of NFRs
+ ratio = (a->dfast * a->xslow) / (a->dslow * a->xfast);
+
+ // begrenzte lineare Kennlinie
+ M = (STEPY2 - STEPY1) / (STEPX2 - STEPX1);
+ if (ratio < STEPX1) {
+ stepsize = STEPY1;
+ } else if (ratio > STEPX2) {
+ stepsize = STEPY2;
+ } else {
+ // Punktrichtungsform einer Geraden
+ stepsize = M * (ratio - STEPX1) + STEPY1;
+ }
+
+ return stepsize;
+}
+
+
+static void AEC_leaky(AEC *a)
+// The xfast signal is used to charge the hangover timer to Thold.
+// When hangover expires (no Spk signal for some time) the vector w
+// is erased. This is my implementation of Leaky NLMS.
+{
+ if (a->xfast >= M70dB_PCM) {
+ // vector w is valid for hangover Thold time
+ a->hangover = Thold;
+ } else {
+ if (a->hangover > 1) {
+ --(a->hangover);
+ } else if (1 == a->hangover) {
+ --(a->hangover);
+ // My Leaky NLMS is to erase vector w when hangover expires
+ memset(a->w, 0, sizeof(a->w));
+ }
+ }
+}
+
+
+#if 0
+void AEC::openwdisplay() {
+ // open TCP connection to program wdisplay.tcl
+ fdwdisplay = socket_async("127.0.0.1", 50999);
+};
+#endif
+
+
+static REAL AEC_nlms_pw(AEC *a, REAL d, REAL x_, float stepsize)
+{
+ REAL e;
+ REAL ef;
+ a->x[a->j] = x_;
+ a->xf[a->j] = IIR1_highpass(a->Fx, x_); // pre-whitening of x
+
+ // calculate error value
+ // (mic signal - estimated mic signal from spk signal)
+ e = d;
+ if (a->hangover > 0) {
+ e -= dotp(a->w, a->x + a->j);
+ }
+ ef = IIR1_highpass(a->Fe, e); // pre-whitening of e
+
+ // optimize: iterative dotp(xf, xf)
+ a->dotp_xf_xf += (a->xf[a->j] * a->xf[a->j] - a->xf[a->j + NLMS_LEN - 1] * a->xf[a->j + NLMS_LEN - 1]);
+
+ if (stepsize > 0.0) {
+ // calculate variable step size
+ REAL mikro_ef = stepsize * ef / a->dotp_xf_xf;
+
+ // update tap weights (filter learning)
+ int i;
+ for (i = 0; i < NLMS_LEN; i += 2) {
+ // optimize: partial loop unrolling
+ a->w[i] += mikro_ef * a->xf[i + a->j];
+ a->w[i + 1] += mikro_ef * a->xf[i + a->j + 1];
+ }
+ }
+
+ if (--(a->j) < 0) {
+ // optimize: decrease number of memory copies
+ a->j = NLMS_EXT;
+ memmove(a->x + a->j + 1, a->x, (NLMS_LEN - 1) * sizeof(REAL));
+ memmove(a->xf + a->j + 1, a->xf, (NLMS_LEN - 1) * sizeof(REAL));
+ }
+
+ // Saturation
+ if (e > MAXPCM) {
+ return MAXPCM;
+ } else if (e < -MAXPCM) {
+ return -MAXPCM;
+ } else {
+ return e;
+ }
+}
+
+
+int AEC_doAEC(AEC *a, int d_, int x_)
+{
+ REAL d = (REAL) d_;
+ REAL x = (REAL) x_;
+
+ // Mic Highpass Filter - to remove DC
+ d = IIR_HP_highpass(a->acMic, d);
+
+ // Mic Highpass Filter - cut-off below 300Hz
+ d = FIR_HP_300Hz_highpass(a->cutoff, d);
+
+ // Amplify, for e.g. Soundcards with -6dB max. volume
+ d *= a->gain;
+
+ // Spk Highpass Filter - to remove DC
+ x = IIR_HP_highpass(a->acSpk, x);
+
+ // Double Talk Detector
+ a->stepsize = AEC_dtd(a, d, x);
+
+ // Leaky (ageing of vector w)
+ AEC_leaky(a);
+
+ // Acoustic Echo Cancellation
+ d = AEC_nlms_pw(a, d, x, a->stepsize);
+
+#if 0
+ if (fdwdisplay >= 0) {
+ if (++dumpcnt >= (WIDEB*RATE/10)) {
+ // wdisplay creates 10 dumps per seconds = large CPU load!
+ dumpcnt = 0;
+ write(fdwdisplay, ws, DUMP_LEN*sizeof(float));
+ // we don't check return value. This is not production quality!!!
+ memset(ws, 0, sizeof(ws));
+ } else {
+ int i;
+ for (i = 0; i < DUMP_LEN; i += 2) {
+ // optimize: partial loop unrolling
+ ws[i] += w[i];
+ ws[i + 1] += w[i + 1];
+ }
+ }
+ }
+#endif
+
+ return (int) d;
+}
diff --git a/src/modules/echo-cancel/adrian-aec.h b/src/modules/echo-cancel/adrian-aec.h
new file mode 100644
index 00000000..1f5b090a
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-aec.h
@@ -0,0 +1,370 @@
+/* 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 */
+
+#define WIDEB 2
+
+// use double if your CPU does software-emulation of float
+typedef float REAL;
+
+/* 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[NLMS_LEN]; // tap weights
+ 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;
+};
+
+/* 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);
+
+/* 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_);
+
+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
+ };
+static void AEC_setgain(AEC *a, float gain_) {
+ a->gain = gain_;
+ };
+#if 0
+ void AEC_openwdisplay(AEC *a);
+#endif
+static void AEC_setaes(AEC *a, float aes_y2_) {
+ a->aes_y2 = aes_y2_;
+ };
+static double AEC_max_dotp_xf_xf(AEC *a, double u);
+
+#define _AEC_H
+#endif
diff --git a/src/modules/echo-cancel/adrian-license.txt b/src/modules/echo-cancel/adrian-license.txt
new file mode 100644
index 00000000..7c06efd0
--- /dev/null
+++ b/src/modules/echo-cancel/adrian-license.txt
@@ -0,0 +1,17 @@
+ Copyright (C) DFS Deutsche Flugsicherung (2004). All Rights Reserved.
+
+ You are allowed to use this source code in any open source or closed
+ source software you want. You are allowed to use the algorithms for a
+ hardware solution. You are allowed to modify the source code.
+ You are not allowed to remove the name of the author from this memo or
+ from the source code files. You are not allowed to monopolize the
+ source code or the algorithms behind the source code as your
+ intellectual property. This source code is free of royalty and comes
+ with no warranty.
+
+--- The following does not apply to the PulseAudio module ---
+
+ Please see g711/gen-lic.txt for the ITU-T G.711 codec copyright.
+ Please see gsm/gen-lic.txt for the ITU-T GSM codec copyright.
+ Please see ilbc/COPYRIGHT and ilbc/NOTICE for the IETF iLBC codec
+ copyright.
diff --git a/src/modules/echo-cancel/adrian.c b/src/modules/echo-cancel/adrian.c
new file mode 100644
index 00000000..86c22cb3
--- /dev/null
+++ b/src/modules/echo-cancel/adrian.c
@@ -0,0 +1,121 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ Contributor: Wim Taymans <wim.taymans@gmail.com>
+
+ The actual implementation is taken from the sources at
+ http://andreadrian.de/intercom/ - for the license, look for
+ adrian-license.txt in the same directory as this file.
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <pulsecore/modargs.h>
+#include "echo-cancel.h"
+
+/* should be between 10-20 ms */
+#define DEFAULT_FRAME_SIZE_MS 20
+
+static const char* const valid_modargs[] = {
+ "frame_size_ms",
+ NULL
+};
+
+static void pa_adrian_ec_fixate_spec(pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map)
+{
+ source_ss->format = PA_SAMPLE_S16LE;
+ source_ss->channels = 1;
+ pa_channel_map_init_mono(source_map);
+
+ *sink_ss = *source_ss;
+ *sink_map = *source_map;
+}
+
+pa_bool_t pa_adrian_ec_init(pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ const char *args)
+{
+ int framelen, rate;
+ uint32_t frame_size_ms;
+ pa_modargs *ma;
+
+ if (!(ma = pa_modargs_new(args, valid_modargs))) {
+ pa_log("Failed to parse submodule arguments.");
+ goto fail;
+ }
+
+ frame_size_ms = DEFAULT_FRAME_SIZE_MS;
+ if (pa_modargs_get_value_u32(ma, "frame_size_ms", &frame_size_ms) < 0 || frame_size_ms < 1 || frame_size_ms > 200) {
+ pa_log("Invalid frame_size_ms specification");
+ goto fail;
+ }
+
+ pa_adrian_ec_fixate_spec(source_ss, source_map, sink_ss, sink_map);
+
+ rate = source_ss->rate;
+ framelen = (rate * frame_size_ms) / 1000;
+
+ ec->params.priv.adrian.blocksize = framelen * pa_frame_size (source_ss);
+
+ pa_log_debug ("Using framelen %d, blocksize %lld, channels %d, rate %d", framelen, (long long) ec->params.priv.adrian.blocksize, source_ss->channels, source_ss->rate);
+
+ ec->params.priv.adrian.aec = AEC_init(rate);
+ if (!ec->params.priv.adrian.aec)
+ goto fail;
+
+ pa_modargs_free(ma);
+ return TRUE;
+
+fail:
+ if (ma)
+ pa_modargs_free(ma);
+ return FALSE;
+}
+
+void pa_adrian_ec_run(pa_echo_canceller *ec, uint8_t *rec, uint8_t *play, uint8_t *out)
+{
+ unsigned int i;
+
+ for (i = 0; i < ec->params.priv.adrian.blocksize; i += 2) {
+ /* We know it's S16LE mono data */
+ int r = (((int8_t) rec[i + 1]) << 8) | rec[i];
+ int p = (((int8_t) play[i + 1]) << 8) | play[i];
+ int res;
+
+ res = AEC_doAEC(ec->params.priv.adrian.aec, r, p);
+ out[i] = (uint8_t) (res & 0xff);
+ out[i + 1] = (uint8_t) ((res >> 8) & 0xff);
+ }
+}
+
+void pa_adrian_ec_done(pa_echo_canceller *ec)
+{
+ pa_xfree(ec->params.priv.adrian.aec);
+ ec->params.priv.adrian.aec = NULL;
+}
+
+uint32_t pa_adrian_ec_get_block_size(pa_echo_canceller *ec)
+{
+ return ec->params.priv.adrian.blocksize;
+}
diff --git a/src/modules/echo-cancel/adrian.h b/src/modules/echo-cancel/adrian.h
new file mode 100644
index 00000000..d02e934d
--- /dev/null
+++ b/src/modules/echo-cancel/adrian.h
@@ -0,0 +1,31 @@
+/***
+ This file is part of PulseAudio.
+
+ Copyright 2010 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ The actual implementation is taken from the sources at
+ http://andreadrian.de/intercom/ - for the license, look for
+ adrian-license.txt in the same directory as this file.
+
+ PulseAudio is free software; you can redistribute it and/or modify
+ it under the terms of the GNU Lesser General Public License as published
+ by the Free Software Foundation; either version 2.1 of the License,
+ or (at your option) any later version.
+
+ PulseAudio is distributed in the hope that it will be useful, but
+ WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+/* Forward declarations */
+
+typedef struct AEC AEC;
+
+AEC* AEC_init(int RATE);
+int AEC_doAEC(AEC *a, int d_, int x_);
diff --git a/src/modules/echo-cancel/echo-cancel.h b/src/modules/echo-cancel/echo-cancel.h
index 205c4d14..65e0e240 100644
--- a/src/modules/echo-cancel/echo-cancel.h
+++ b/src/modules/echo-cancel/echo-cancel.h
@@ -28,6 +28,7 @@
#include <pulsecore/macro.h>
#include <speex/speex_echo.h>
+#include "adrian.h"
/* Common data structures */
@@ -39,6 +40,10 @@ struct pa_echo_canceller_params {
uint32_t blocksize;
SpeexEchoState *state;
} speex;
+ struct {
+ uint32_t blocksize;
+ AEC *aec;
+ } adrian;
/* each canceller-specific structure goes here */
} priv;
};
@@ -67,3 +72,12 @@ pa_bool_t pa_speex_ec_init(pa_echo_canceller *ec,
void pa_speex_ec_run(pa_echo_canceller *ec, uint8_t *rec, uint8_t *play, uint8_t *out);
void pa_speex_ec_done(pa_echo_canceller *ec);
uint32_t pa_speex_ec_get_block_size(pa_echo_canceller *ec);
+
+/* Adrian Andre's echo canceller */
+pa_bool_t pa_adrian_ec_init(pa_echo_canceller *ec,
+ pa_sample_spec *source_ss, pa_channel_map *source_map,
+ pa_sample_spec *sink_ss, pa_channel_map *sink_map,
+ const char *args);
+void pa_adrian_ec_run(pa_echo_canceller *ec, uint8_t *rec, uint8_t *play, uint8_t *out);
+void pa_adrian_ec_done(pa_echo_canceller *ec);
+uint32_t pa_adrian_ec_get_block_size(pa_echo_canceller *ec);
diff --git a/src/modules/echo-cancel/module-echo-cancel.c b/src/modules/echo-cancel/module-echo-cancel.c
index 6a88167b..75f74d34 100644
--- a/src/modules/echo-cancel/module-echo-cancel.c
+++ b/src/modules/echo-cancel/module-echo-cancel.c
@@ -82,6 +82,7 @@ PA_MODULE_USAGE(
/* NOTE: Make sure the enum and ec_table are maintained in the correct order */
enum {
PA_ECHO_CANCELLER_SPEEX,
+ PA_ECHO_CANCELLER_ADRIAN,
};
#define DEFAULT_ECHO_CANCELLER PA_ECHO_CANCELLER_SPEEX
@@ -94,6 +95,13 @@ static const pa_echo_canceller ec_table[] = {
.done = pa_speex_ec_done,
.get_block_size = pa_speex_ec_get_block_size,
},
+ {
+ /* Adrian Andre's NLMS implementation */
+ .init = pa_adrian_ec_init,
+ .run = pa_adrian_ec_run,
+ .done = pa_adrian_ec_done,
+ .get_block_size = pa_adrian_ec_get_block_size,
+ },
};
#define DEFAULT_ADJUST_TIME_USEC (1*PA_USEC_PER_SEC)
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-28 05:15:14 +0000