Adding upstream version 14.2.upstream/14.2upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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