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547b6a570d
This is a relatively minor bump, but it's the first step towards bringing this current. Change-Id: Iab6c9b0c77f0ba705280434ea74b513364719499
1285 lines
43 KiB
C
1285 lines
43 KiB
C
/* Copyright (C) 2003-2008 Jean-Marc Valin
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File: mdf.c
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Echo canceller based on the MDF algorithm (see below)
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions are
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met:
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1. Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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2. Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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3. The name of the author may not be used to endorse or promote products
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derived from this software without specific prior written permission.
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THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
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INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
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HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
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STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
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ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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POSSIBILITY OF SUCH DAMAGE.
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*/
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/*
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The echo canceller is based on the MDF algorithm described in:
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J. S. Soo, K. K. Pang Multidelay block frequency adaptive filter,
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IEEE Trans. Acoust. Speech Signal Process., Vol. ASSP-38, No. 2,
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February 1990.
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We use the Alternatively Updated MDF (AUMDF) variant. Robustness to
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double-talk is achieved using a variable learning rate as described in:
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Valin, J.-M., On Adjusting the Learning Rate in Frequency Domain Echo
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Cancellation With Double-Talk. IEEE Transactions on Audio,
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Speech and Language Processing, Vol. 15, No. 3, pp. 1030-1034, 2007.
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http://people.xiph.org/~jm/papers/valin_taslp2006.pdf
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There is no explicit double-talk detection, but a continuous variation
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in the learning rate based on residual echo, double-talk and background
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noise.
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About the fixed-point version:
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All the signals are represented with 16-bit words. The filter weights
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are represented with 32-bit words, but only the top 16 bits are used
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in most cases. The lower 16 bits are completely unreliable (due to the
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fact that the update is done only on the top bits), but help in the
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adaptation -- probably by removing a "threshold effect" due to
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quantization (rounding going to zero) when the gradient is small.
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Another kludge that seems to work good: when performing the weight
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update, we only move half the way toward the "goal" this seems to
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reduce the effect of quantization noise in the update phase. This
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can be seen as applying a gradient descent on a "soft constraint"
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instead of having a hard constraint.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config-speex.h"
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#endif
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#include "arch.h"
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#include "speex/speex_echo.h"
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#include "fftwrap.h"
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#include "pseudofloat.h"
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#include "math_approx.h"
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#include "os_support.h"
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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#ifdef FIXED_POINT
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#define WEIGHT_SHIFT 11
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#define NORMALIZE_SCALEDOWN 5
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#define NORMALIZE_SCALEUP 3
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#else
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#define WEIGHT_SHIFT 0
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#endif
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#ifdef FIXED_POINT
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#define WORD2INT(x) ((x) < -32767 ? -32768 : ((x) > 32766 ? 32767 : (x)))
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#else
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#define WORD2INT(x) ((x) < -32767.5f ? -32768 : ((x) > 32766.5f ? 32767 : floor(.5+(x))))
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#endif
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/* If enabled, the AEC will use a foreground filter and a background filter to be more robust to double-talk
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and difficult signals in general. The cost is an extra FFT and a matrix-vector multiply */
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#define TWO_PATH
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#ifdef FIXED_POINT
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static const spx_float_t MIN_LEAK = {20972, -22};
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/* Constants for the two-path filter */
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static const spx_float_t VAR1_SMOOTH = {23593, -16};
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static const spx_float_t VAR2_SMOOTH = {23675, -15};
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static const spx_float_t VAR1_UPDATE = {16384, -15};
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static const spx_float_t VAR2_UPDATE = {16384, -16};
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static const spx_float_t VAR_BACKTRACK = {16384, -12};
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#define TOP16(x) ((x)>>16)
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#else
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static const spx_float_t MIN_LEAK = .005f;
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/* Constants for the two-path filter */
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static const spx_float_t VAR1_SMOOTH = .36f;
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static const spx_float_t VAR2_SMOOTH = .7225f;
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static const spx_float_t VAR1_UPDATE = .5f;
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static const spx_float_t VAR2_UPDATE = .25f;
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static const spx_float_t VAR_BACKTRACK = 4.f;
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#define TOP16(x) (x)
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#endif
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#define PLAYBACK_DELAY 2
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void speex_echo_get_residual(SpeexEchoState *st, spx_word32_t *Yout, int len);
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/** Speex echo cancellation state. */
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struct SpeexEchoState_ {
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int frame_size; /**< Number of samples processed each time */
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int window_size;
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int M;
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int cancel_count;
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int adapted;
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int saturated;
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int screwed_up;
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int C; /** Number of input channels (microphones) */
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int K; /** Number of output channels (loudspeakers) */
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spx_int32_t sampling_rate;
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spx_word16_t spec_average;
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spx_word16_t beta0;
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spx_word16_t beta_max;
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spx_word32_t sum_adapt;
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spx_word16_t leak_estimate;
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spx_word16_t *e; /* scratch */
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spx_word16_t *x; /* Far-end input buffer (2N) */
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spx_word16_t *X; /* Far-end buffer (M+1 frames) in frequency domain */
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spx_word16_t *input; /* scratch */
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spx_word16_t *y; /* scratch */
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spx_word16_t *last_y;
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spx_word16_t *Y; /* scratch */
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spx_word16_t *E;
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spx_word32_t *PHI; /* scratch */
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spx_word32_t *W; /* (Background) filter weights */
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#ifdef TWO_PATH
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spx_word16_t *foreground; /* Foreground filter weights */
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spx_word32_t Davg1; /* 1st recursive average of the residual power difference */
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spx_word32_t Davg2; /* 2nd recursive average of the residual power difference */
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spx_float_t Dvar1; /* Estimated variance of 1st estimator */
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spx_float_t Dvar2; /* Estimated variance of 2nd estimator */
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#endif
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spx_word32_t *power; /* Power of the far-end signal */
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spx_float_t *power_1;/* Inverse power of far-end */
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spx_word16_t *wtmp; /* scratch */
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#ifdef FIXED_POINT
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spx_word16_t *wtmp2; /* scratch */
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#endif
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spx_word32_t *Rf; /* scratch */
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spx_word32_t *Yf; /* scratch */
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spx_word32_t *Xf; /* scratch */
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spx_word32_t *Eh;
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spx_word32_t *Yh;
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spx_float_t Pey;
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spx_float_t Pyy;
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spx_word16_t *window;
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spx_word16_t *prop;
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void *fft_table;
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spx_word16_t *memX, *memD, *memE;
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spx_word16_t preemph;
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spx_word16_t notch_radius;
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spx_mem_t *notch_mem;
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/* NOTE: If you only use speex_echo_cancel() and want to save some memory, remove this */
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spx_int16_t *play_buf;
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int play_buf_pos;
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int play_buf_started;
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};
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static inline void filter_dc_notch16(const spx_int16_t *in, spx_word16_t radius, spx_word16_t *out, int len, spx_mem_t *mem, int stride)
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{
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int i;
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spx_word16_t den2;
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#ifdef FIXED_POINT
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den2 = MULT16_16_Q15(radius,radius) + MULT16_16_Q15(QCONST16(.7,15),MULT16_16_Q15(32767-radius,32767-radius));
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#else
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den2 = radius*radius + .7*(1-radius)*(1-radius);
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#endif
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/*printf ("%d %d %d %d %d %d\n", num[0], num[1], num[2], den[0], den[1], den[2]);*/
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for (i=0;i<len;i++)
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{
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spx_word16_t vin = in[i*stride];
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spx_word32_t vout = mem[0] + SHL32(EXTEND32(vin),15);
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#ifdef FIXED_POINT
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mem[0] = mem[1] + SHL32(SHL32(-EXTEND32(vin),15) + MULT16_32_Q15(radius,vout),1);
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#else
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mem[0] = mem[1] + 2*(-vin + radius*vout);
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#endif
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mem[1] = SHL32(EXTEND32(vin),15) - MULT16_32_Q15(den2,vout);
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out[i] = SATURATE32(PSHR32(MULT16_32_Q15(radius,vout),15),32767);
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}
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}
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/* This inner product is slightly different from the codec version because of fixed-point */
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static inline spx_word32_t mdf_inner_prod(const spx_word16_t *x, const spx_word16_t *y, int len)
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{
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spx_word32_t sum=0;
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len >>= 1;
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while(len--)
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{
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spx_word32_t part=0;
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part = MAC16_16(part,*x++,*y++);
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part = MAC16_16(part,*x++,*y++);
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/* HINT: If you had a 40-bit accumulator, you could shift only at the end */
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sum = ADD32(sum,SHR32(part,6));
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}
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return sum;
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}
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/** Compute power spectrum of a half-complex (packed) vector */
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static inline void power_spectrum(const spx_word16_t *X, spx_word32_t *ps, int N)
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{
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int i, j;
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ps[0]=MULT16_16(X[0],X[0]);
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for (i=1,j=1;i<N-1;i+=2,j++)
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{
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ps[j] = MULT16_16(X[i],X[i]) + MULT16_16(X[i+1],X[i+1]);
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}
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ps[j]=MULT16_16(X[i],X[i]);
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}
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/** Compute power spectrum of a half-complex (packed) vector and accumulate */
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static inline void power_spectrum_accum(const spx_word16_t *X, spx_word32_t *ps, int N)
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{
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int i, j;
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ps[0]+=MULT16_16(X[0],X[0]);
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for (i=1,j=1;i<N-1;i+=2,j++)
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{
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ps[j] += MULT16_16(X[i],X[i]) + MULT16_16(X[i+1],X[i+1]);
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}
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ps[j]+=MULT16_16(X[i],X[i]);
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}
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/** Compute cross-power spectrum of a half-complex (packed) vectors and add to acc */
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#ifdef FIXED_POINT
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static inline void spectral_mul_accum(const spx_word16_t *X, const spx_word32_t *Y, spx_word16_t *acc, int N, int M)
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{
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int i,j;
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spx_word32_t tmp1=0,tmp2=0;
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for (j=0;j<M;j++)
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{
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tmp1 = MAC16_16(tmp1, X[j*N],TOP16(Y[j*N]));
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}
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acc[0] = PSHR32(tmp1,WEIGHT_SHIFT);
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for (i=1;i<N-1;i+=2)
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{
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tmp1 = tmp2 = 0;
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for (j=0;j<M;j++)
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{
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tmp1 = SUB32(MAC16_16(tmp1, X[j*N+i],TOP16(Y[j*N+i])), MULT16_16(X[j*N+i+1],TOP16(Y[j*N+i+1])));
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tmp2 = MAC16_16(MAC16_16(tmp2, X[j*N+i+1],TOP16(Y[j*N+i])), X[j*N+i], TOP16(Y[j*N+i+1]));
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}
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acc[i] = PSHR32(tmp1,WEIGHT_SHIFT);
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acc[i+1] = PSHR32(tmp2,WEIGHT_SHIFT);
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}
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tmp1 = tmp2 = 0;
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for (j=0;j<M;j++)
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{
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tmp1 = MAC16_16(tmp1, X[(j+1)*N-1],TOP16(Y[(j+1)*N-1]));
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}
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acc[N-1] = PSHR32(tmp1,WEIGHT_SHIFT);
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}
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static inline void spectral_mul_accum16(const spx_word16_t *X, const spx_word16_t *Y, spx_word16_t *acc, int N, int M)
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{
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int i,j;
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spx_word32_t tmp1=0,tmp2=0;
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for (j=0;j<M;j++)
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{
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tmp1 = MAC16_16(tmp1, X[j*N],Y[j*N]);
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}
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acc[0] = PSHR32(tmp1,WEIGHT_SHIFT);
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for (i=1;i<N-1;i+=2)
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{
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tmp1 = tmp2 = 0;
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for (j=0;j<M;j++)
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{
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tmp1 = SUB32(MAC16_16(tmp1, X[j*N+i],Y[j*N+i]), MULT16_16(X[j*N+i+1],Y[j*N+i+1]));
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tmp2 = MAC16_16(MAC16_16(tmp2, X[j*N+i+1],Y[j*N+i]), X[j*N+i], Y[j*N+i+1]);
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}
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acc[i] = PSHR32(tmp1,WEIGHT_SHIFT);
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acc[i+1] = PSHR32(tmp2,WEIGHT_SHIFT);
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}
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tmp1 = tmp2 = 0;
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for (j=0;j<M;j++)
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{
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tmp1 = MAC16_16(tmp1, X[(j+1)*N-1],Y[(j+1)*N-1]);
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}
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acc[N-1] = PSHR32(tmp1,WEIGHT_SHIFT);
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}
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#else
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static inline void spectral_mul_accum(const spx_word16_t *X, const spx_word32_t *Y, spx_word16_t *acc, int N, int M)
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{
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int i,j;
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for (i=0;i<N;i++)
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acc[i] = 0;
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for (j=0;j<M;j++)
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{
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acc[0] += X[0]*Y[0];
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for (i=1;i<N-1;i+=2)
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{
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acc[i] += (X[i]*Y[i] - X[i+1]*Y[i+1]);
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acc[i+1] += (X[i+1]*Y[i] + X[i]*Y[i+1]);
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}
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acc[i] += X[i]*Y[i];
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X += N;
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Y += N;
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}
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}
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#define spectral_mul_accum16 spectral_mul_accum
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#endif
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/** Compute weighted cross-power spectrum of a half-complex (packed) vector with conjugate */
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static inline void weighted_spectral_mul_conj(const spx_float_t *w, const spx_float_t p, const spx_word16_t *X, const spx_word16_t *Y, spx_word32_t *prod, int N)
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{
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int i, j;
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spx_float_t W;
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W = FLOAT_AMULT(p, w[0]);
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prod[0] = FLOAT_MUL32(W,MULT16_16(X[0],Y[0]));
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for (i=1,j=1;i<N-1;i+=2,j++)
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{
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W = FLOAT_AMULT(p, w[j]);
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prod[i] = FLOAT_MUL32(W,MAC16_16(MULT16_16(X[i],Y[i]), X[i+1],Y[i+1]));
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prod[i+1] = FLOAT_MUL32(W,MAC16_16(MULT16_16(-X[i+1],Y[i]), X[i],Y[i+1]));
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}
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W = FLOAT_AMULT(p, w[j]);
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prod[i] = FLOAT_MUL32(W,MULT16_16(X[i],Y[i]));
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}
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static inline void mdf_adjust_prop(const spx_word32_t *W, int N, int M, int P, spx_word16_t *prop)
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{
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int i, j, p;
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spx_word16_t max_sum = 1;
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spx_word32_t prop_sum = 1;
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for (i=0;i<M;i++)
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{
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spx_word32_t tmp = 1;
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for (p=0;p<P;p++)
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for (j=0;j<N;j++)
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tmp += MULT16_16(EXTRACT16(SHR32(W[p*N*M + i*N+j],18)), EXTRACT16(SHR32(W[p*N*M + i*N+j],18)));
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#ifdef FIXED_POINT
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/* Just a security in case an overflow were to occur */
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tmp = MIN32(ABS32(tmp), 536870912);
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#endif
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prop[i] = spx_sqrt(tmp);
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if (prop[i] > max_sum)
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max_sum = prop[i];
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}
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for (i=0;i<M;i++)
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{
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prop[i] += MULT16_16_Q15(QCONST16(.1f,15),max_sum);
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prop_sum += EXTEND32(prop[i]);
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}
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for (i=0;i<M;i++)
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{
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prop[i] = DIV32(MULT16_16(QCONST16(.99f,15), prop[i]),prop_sum);
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/*printf ("%f ", prop[i]);*/
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}
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/*printf ("\n");*/
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}
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#ifdef DUMP_ECHO_CANCEL_DATA
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#include <stdio.h>
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static FILE *rFile=NULL, *pFile=NULL, *oFile=NULL;
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static void dump_audio(const spx_int16_t *rec, const spx_int16_t *play, const spx_int16_t *out, int len)
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{
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if (!(rFile && pFile && oFile))
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{
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speex_fatal("Dump files not open");
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}
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fwrite(rec, sizeof(spx_int16_t), len, rFile);
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fwrite(play, sizeof(spx_int16_t), len, pFile);
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fwrite(out, sizeof(spx_int16_t), len, oFile);
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}
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#endif
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/** Creates a new echo canceller state */
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EXPORT SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length)
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{
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return speex_echo_state_init_mc(frame_size, filter_length, 1, 1);
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}
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EXPORT SpeexEchoState *speex_echo_state_init_mc(int frame_size, int filter_length, int nb_mic, int nb_speakers)
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{
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int i,N,M, C, K;
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SpeexEchoState *st = (SpeexEchoState *)speex_alloc(sizeof(SpeexEchoState));
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st->K = nb_speakers;
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st->C = nb_mic;
|
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C=st->C;
|
|
K=st->K;
|
|
#ifdef DUMP_ECHO_CANCEL_DATA
|
|
if (rFile || pFile || oFile)
|
|
speex_fatal("Opening dump files twice");
|
|
rFile = fopen("aec_rec.sw", "wb");
|
|
pFile = fopen("aec_play.sw", "wb");
|
|
oFile = fopen("aec_out.sw", "wb");
|
|
#endif
|
|
|
|
st->frame_size = frame_size;
|
|
st->window_size = 2*frame_size;
|
|
N = st->window_size;
|
|
M = st->M = (filter_length+st->frame_size-1)/frame_size;
|
|
st->cancel_count=0;
|
|
st->sum_adapt = 0;
|
|
st->saturated = 0;
|
|
st->screwed_up = 0;
|
|
/* This is the default sampling rate */
|
|
st->sampling_rate = 8000;
|
|
st->spec_average = DIV32_16(SHL32(EXTEND32(st->frame_size), 15), st->sampling_rate);
|
|
#ifdef FIXED_POINT
|
|
st->beta0 = DIV32_16(SHL32(EXTEND32(st->frame_size), 16), st->sampling_rate);
|
|
st->beta_max = DIV32_16(SHL32(EXTEND32(st->frame_size), 14), st->sampling_rate);
|
|
#else
|
|
st->beta0 = (2.0f*st->frame_size)/st->sampling_rate;
|
|
st->beta_max = (.5f*st->frame_size)/st->sampling_rate;
|
|
#endif
|
|
st->leak_estimate = 0;
|
|
|
|
st->fft_table = spx_fft_init(N);
|
|
|
|
st->e = (spx_word16_t*)speex_alloc(C*N*sizeof(spx_word16_t));
|
|
st->x = (spx_word16_t*)speex_alloc(K*N*sizeof(spx_word16_t));
|
|
st->input = (spx_word16_t*)speex_alloc(C*st->frame_size*sizeof(spx_word16_t));
|
|
st->y = (spx_word16_t*)speex_alloc(C*N*sizeof(spx_word16_t));
|
|
st->last_y = (spx_word16_t*)speex_alloc(C*N*sizeof(spx_word16_t));
|
|
st->Yf = (spx_word32_t*)speex_alloc((st->frame_size+1)*sizeof(spx_word32_t));
|
|
st->Rf = (spx_word32_t*)speex_alloc((st->frame_size+1)*sizeof(spx_word32_t));
|
|
st->Xf = (spx_word32_t*)speex_alloc((st->frame_size+1)*sizeof(spx_word32_t));
|
|
st->Yh = (spx_word32_t*)speex_alloc((st->frame_size+1)*sizeof(spx_word32_t));
|
|
st->Eh = (spx_word32_t*)speex_alloc((st->frame_size+1)*sizeof(spx_word32_t));
|
|
|
|
st->X = (spx_word16_t*)speex_alloc(K*(M+1)*N*sizeof(spx_word16_t));
|
|
st->Y = (spx_word16_t*)speex_alloc(C*N*sizeof(spx_word16_t));
|
|
st->E = (spx_word16_t*)speex_alloc(C*N*sizeof(spx_word16_t));
|
|
st->W = (spx_word32_t*)speex_alloc(C*K*M*N*sizeof(spx_word32_t));
|
|
#ifdef TWO_PATH
|
|
st->foreground = (spx_word16_t*)speex_alloc(M*N*C*K*sizeof(spx_word16_t));
|
|
#endif
|
|
st->PHI = (spx_word32_t*)speex_alloc(N*sizeof(spx_word32_t));
|
|
st->power = (spx_word32_t*)speex_alloc((frame_size+1)*sizeof(spx_word32_t));
|
|
st->power_1 = (spx_float_t*)speex_alloc((frame_size+1)*sizeof(spx_float_t));
|
|
st->window = (spx_word16_t*)speex_alloc(N*sizeof(spx_word16_t));
|
|
st->prop = (spx_word16_t*)speex_alloc(M*sizeof(spx_word16_t));
|
|
st->wtmp = (spx_word16_t*)speex_alloc(N*sizeof(spx_word16_t));
|
|
#ifdef FIXED_POINT
|
|
st->wtmp2 = (spx_word16_t*)speex_alloc(N*sizeof(spx_word16_t));
|
|
for (i=0;i<N>>1;i++)
|
|
{
|
|
st->window[i] = (16383-SHL16(spx_cos(DIV32_16(MULT16_16(25736,i<<1),N)),1));
|
|
st->window[N-i-1] = st->window[i];
|
|
}
|
|
#else
|
|
for (i=0;i<N;i++)
|
|
st->window[i] = .5-.5*cos(2*M_PI*i/N);
|
|
#endif
|
|
for (i=0;i<=st->frame_size;i++)
|
|
st->power_1[i] = FLOAT_ONE;
|
|
for (i=0;i<N*M*K*C;i++)
|
|
st->W[i] = 0;
|
|
{
|
|
spx_word32_t sum = 0;
|
|
/* Ratio of ~10 between adaptation rate of first and last block */
|
|
spx_word16_t decay = SHR32(spx_exp(NEG16(DIV32_16(QCONST16(2.4,11),M))),1);
|
|
st->prop[0] = QCONST16(.7, 15);
|
|
sum = EXTEND32(st->prop[0]);
|
|
for (i=1;i<M;i++)
|
|
{
|
|
st->prop[i] = MULT16_16_Q15(st->prop[i-1], decay);
|
|
sum = ADD32(sum, EXTEND32(st->prop[i]));
|
|
}
|
|
for (i=M-1;i>=0;i--)
|
|
{
|
|
st->prop[i] = DIV32(MULT16_16(QCONST16(.8f,15), st->prop[i]),sum);
|
|
}
|
|
}
|
|
|
|
st->memX = (spx_word16_t*)speex_alloc(K*sizeof(spx_word16_t));
|
|
st->memD = (spx_word16_t*)speex_alloc(C*sizeof(spx_word16_t));
|
|
st->memE = (spx_word16_t*)speex_alloc(C*sizeof(spx_word16_t));
|
|
st->preemph = QCONST16(.9,15);
|
|
if (st->sampling_rate<12000)
|
|
st->notch_radius = QCONST16(.9, 15);
|
|
else if (st->sampling_rate<24000)
|
|
st->notch_radius = QCONST16(.982, 15);
|
|
else
|
|
st->notch_radius = QCONST16(.992, 15);
|
|
|
|
st->notch_mem = (spx_mem_t*)speex_alloc(2*C*sizeof(spx_mem_t));
|
|
st->adapted = 0;
|
|
st->Pey = st->Pyy = FLOAT_ONE;
|
|
|
|
#ifdef TWO_PATH
|
|
st->Davg1 = st->Davg2 = 0;
|
|
st->Dvar1 = st->Dvar2 = FLOAT_ZERO;
|
|
#endif
|
|
|
|
st->play_buf = (spx_int16_t*)speex_alloc(K*(PLAYBACK_DELAY+1)*st->frame_size*sizeof(spx_int16_t));
|
|
st->play_buf_pos = PLAYBACK_DELAY*st->frame_size;
|
|
st->play_buf_started = 0;
|
|
|
|
return st;
|
|
}
|
|
|
|
/** Resets echo canceller state */
|
|
EXPORT void speex_echo_state_reset(SpeexEchoState *st)
|
|
{
|
|
int i, M, N, C, K;
|
|
st->cancel_count=0;
|
|
st->screwed_up = 0;
|
|
N = st->window_size;
|
|
M = st->M;
|
|
C=st->C;
|
|
K=st->K;
|
|
for (i=0;i<N*M;i++)
|
|
st->W[i] = 0;
|
|
#ifdef TWO_PATH
|
|
for (i=0;i<N*M;i++)
|
|
st->foreground[i] = 0;
|
|
#endif
|
|
for (i=0;i<N*(M+1);i++)
|
|
st->X[i] = 0;
|
|
for (i=0;i<=st->frame_size;i++)
|
|
{
|
|
st->power[i] = 0;
|
|
st->power_1[i] = FLOAT_ONE;
|
|
st->Eh[i] = 0;
|
|
st->Yh[i] = 0;
|
|
}
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
st->last_y[i] = 0;
|
|
}
|
|
for (i=0;i<N*C;i++)
|
|
{
|
|
st->E[i] = 0;
|
|
}
|
|
for (i=0;i<N*K;i++)
|
|
{
|
|
st->x[i] = 0;
|
|
}
|
|
for (i=0;i<2*C;i++)
|
|
st->notch_mem[i] = 0;
|
|
for (i=0;i<C;i++)
|
|
st->memD[i]=st->memE[i]=0;
|
|
for (i=0;i<K;i++)
|
|
st->memX[i]=0;
|
|
|
|
st->saturated = 0;
|
|
st->adapted = 0;
|
|
st->sum_adapt = 0;
|
|
st->Pey = st->Pyy = FLOAT_ONE;
|
|
#ifdef TWO_PATH
|
|
st->Davg1 = st->Davg2 = 0;
|
|
st->Dvar1 = st->Dvar2 = FLOAT_ZERO;
|
|
#endif
|
|
for (i=0;i<3*st->frame_size;i++)
|
|
st->play_buf[i] = 0;
|
|
st->play_buf_pos = PLAYBACK_DELAY*st->frame_size;
|
|
st->play_buf_started = 0;
|
|
|
|
}
|
|
|
|
/** Destroys an echo canceller state */
|
|
EXPORT void speex_echo_state_destroy(SpeexEchoState *st)
|
|
{
|
|
spx_fft_destroy(st->fft_table);
|
|
|
|
speex_free(st->e);
|
|
speex_free(st->x);
|
|
speex_free(st->input);
|
|
speex_free(st->y);
|
|
speex_free(st->last_y);
|
|
speex_free(st->Yf);
|
|
speex_free(st->Rf);
|
|
speex_free(st->Xf);
|
|
speex_free(st->Yh);
|
|
speex_free(st->Eh);
|
|
|
|
speex_free(st->X);
|
|
speex_free(st->Y);
|
|
speex_free(st->E);
|
|
speex_free(st->W);
|
|
#ifdef TWO_PATH
|
|
speex_free(st->foreground);
|
|
#endif
|
|
speex_free(st->PHI);
|
|
speex_free(st->power);
|
|
speex_free(st->power_1);
|
|
speex_free(st->window);
|
|
speex_free(st->prop);
|
|
speex_free(st->wtmp);
|
|
#ifdef FIXED_POINT
|
|
speex_free(st->wtmp2);
|
|
#endif
|
|
speex_free(st->memX);
|
|
speex_free(st->memD);
|
|
speex_free(st->memE);
|
|
speex_free(st->notch_mem);
|
|
|
|
speex_free(st->play_buf);
|
|
speex_free(st);
|
|
|
|
#ifdef DUMP_ECHO_CANCEL_DATA
|
|
fclose(rFile);
|
|
fclose(pFile);
|
|
fclose(oFile);
|
|
rFile = pFile = oFile = NULL;
|
|
#endif
|
|
}
|
|
|
|
EXPORT void speex_echo_capture(SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out)
|
|
{
|
|
int i;
|
|
/*speex_warning_int("capture with fill level ", st->play_buf_pos/st->frame_size);*/
|
|
st->play_buf_started = 1;
|
|
if (st->play_buf_pos>=st->frame_size)
|
|
{
|
|
speex_echo_cancellation(st, rec, st->play_buf, out);
|
|
st->play_buf_pos -= st->frame_size;
|
|
for (i=0;i<st->play_buf_pos;i++)
|
|
st->play_buf[i] = st->play_buf[i+st->frame_size];
|
|
} else {
|
|
speex_warning("No playback frame available (your application is buggy and/or got xruns)");
|
|
if (st->play_buf_pos!=0)
|
|
{
|
|
speex_warning("internal playback buffer corruption?");
|
|
st->play_buf_pos = 0;
|
|
}
|
|
for (i=0;i<st->frame_size;i++)
|
|
out[i] = rec[i];
|
|
}
|
|
}
|
|
|
|
EXPORT void speex_echo_playback(SpeexEchoState *st, const spx_int16_t *play)
|
|
{
|
|
/*speex_warning_int("playback with fill level ", st->play_buf_pos/st->frame_size);*/
|
|
if (!st->play_buf_started)
|
|
{
|
|
speex_warning("discarded first playback frame");
|
|
return;
|
|
}
|
|
if (st->play_buf_pos<=PLAYBACK_DELAY*st->frame_size)
|
|
{
|
|
int i;
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->play_buf[st->play_buf_pos+i] = play[i];
|
|
st->play_buf_pos += st->frame_size;
|
|
if (st->play_buf_pos <= (PLAYBACK_DELAY-1)*st->frame_size)
|
|
{
|
|
speex_warning("Auto-filling the buffer (your application is buggy and/or got xruns)");
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->play_buf[st->play_buf_pos+i] = play[i];
|
|
st->play_buf_pos += st->frame_size;
|
|
}
|
|
} else {
|
|
speex_warning("Had to discard a playback frame (your application is buggy and/or got xruns)");
|
|
}
|
|
}
|
|
|
|
/** Performs echo cancellation on a frame (deprecated, last arg now ignored) */
|
|
EXPORT void speex_echo_cancel(SpeexEchoState *st, const spx_int16_t *in, const spx_int16_t *far_end, spx_int16_t *out, spx_int32_t *Yout)
|
|
{
|
|
speex_echo_cancellation(st, in, far_end, out);
|
|
}
|
|
|
|
/** Performs echo cancellation on a frame */
|
|
EXPORT void speex_echo_cancellation(SpeexEchoState *st, const spx_int16_t *in, const spx_int16_t *far_end, spx_int16_t *out)
|
|
{
|
|
int i,j, chan, speak;
|
|
int N,M, C, K;
|
|
spx_word32_t Syy,See,Sxx,Sdd, Sff;
|
|
#ifdef TWO_PATH
|
|
spx_word32_t Dbf;
|
|
int update_foreground;
|
|
#endif
|
|
spx_word32_t Sey;
|
|
spx_word16_t ss, ss_1;
|
|
spx_float_t Pey = FLOAT_ONE, Pyy=FLOAT_ONE;
|
|
spx_float_t alpha, alpha_1;
|
|
spx_word16_t RER;
|
|
spx_word32_t tmp32;
|
|
|
|
N = st->window_size;
|
|
M = st->M;
|
|
C = st->C;
|
|
K = st->K;
|
|
|
|
st->cancel_count++;
|
|
#ifdef FIXED_POINT
|
|
ss=DIV32_16(11469,M);
|
|
ss_1 = SUB16(32767,ss);
|
|
#else
|
|
ss=.35/M;
|
|
ss_1 = 1-ss;
|
|
#endif
|
|
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
/* Apply a notch filter to make sure DC doesn't end up causing problems */
|
|
filter_dc_notch16(in+chan, st->notch_radius, st->input+chan*st->frame_size, st->frame_size, st->notch_mem+2*chan, C);
|
|
/* Copy input data to buffer and apply pre-emphasis */
|
|
/* Copy input data to buffer */
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
spx_word32_t tmp32;
|
|
/* FIXME: This core has changed a bit, need to merge properly */
|
|
tmp32 = SUB32(EXTEND32(st->input[chan*st->frame_size+i]), EXTEND32(MULT16_16_P15(st->preemph, st->memD[chan])));
|
|
#ifdef FIXED_POINT
|
|
if (tmp32 > 32767)
|
|
{
|
|
tmp32 = 32767;
|
|
if (st->saturated == 0)
|
|
st->saturated = 1;
|
|
}
|
|
if (tmp32 < -32767)
|
|
{
|
|
tmp32 = -32767;
|
|
if (st->saturated == 0)
|
|
st->saturated = 1;
|
|
}
|
|
#endif
|
|
st->memD[chan] = st->input[chan*st->frame_size+i];
|
|
st->input[chan*st->frame_size+i] = EXTRACT16(tmp32);
|
|
}
|
|
}
|
|
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
spx_word32_t tmp32;
|
|
st->x[speak*N+i] = st->x[speak*N+i+st->frame_size];
|
|
tmp32 = SUB32(EXTEND32(far_end[i*K+speak]), EXTEND32(MULT16_16_P15(st->preemph, st->memX[speak])));
|
|
#ifdef FIXED_POINT
|
|
/*FIXME: If saturation occurs here, we need to freeze adaptation for M frames (not just one) */
|
|
if (tmp32 > 32767)
|
|
{
|
|
tmp32 = 32767;
|
|
st->saturated = M+1;
|
|
}
|
|
if (tmp32 < -32767)
|
|
{
|
|
tmp32 = -32767;
|
|
st->saturated = M+1;
|
|
}
|
|
#endif
|
|
st->x[speak*N+i+st->frame_size] = EXTRACT16(tmp32);
|
|
st->memX[speak] = far_end[i*K+speak];
|
|
}
|
|
}
|
|
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
/* Shift memory: this could be optimized eventually*/
|
|
for (j=M-1;j>=0;j--)
|
|
{
|
|
for (i=0;i<N;i++)
|
|
st->X[(j+1)*N*K+speak*N+i] = st->X[j*N*K+speak*N+i];
|
|
}
|
|
/* Convert x (echo input) to frequency domain */
|
|
spx_fft(st->fft_table, st->x+speak*N, &st->X[speak*N]);
|
|
}
|
|
|
|
Sxx = 0;
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
Sxx += mdf_inner_prod(st->x+speak*N+st->frame_size, st->x+speak*N+st->frame_size, st->frame_size);
|
|
power_spectrum_accum(st->X+speak*N, st->Xf, N);
|
|
}
|
|
|
|
Sff = 0;
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
#ifdef TWO_PATH
|
|
/* Compute foreground filter */
|
|
spectral_mul_accum16(st->X, st->foreground+chan*N*K*M, st->Y+chan*N, N, M*K);
|
|
spx_ifft(st->fft_table, st->Y+chan*N, st->e+chan*N);
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->e[chan*N+i] = SUB16(st->input[chan*st->frame_size+i], st->e[chan*N+i+st->frame_size]);
|
|
Sff += mdf_inner_prod(st->e+chan*N, st->e+chan*N, st->frame_size);
|
|
#endif
|
|
}
|
|
|
|
/* Adjust proportional adaption rate */
|
|
/* FIXME: Adjust that for C, K*/
|
|
if (st->adapted)
|
|
mdf_adjust_prop (st->W, N, M, C*K, st->prop);
|
|
/* Compute weight gradient */
|
|
if (st->saturated == 0)
|
|
{
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
for (j=M-1;j>=0;j--)
|
|
{
|
|
weighted_spectral_mul_conj(st->power_1, FLOAT_SHL(PSEUDOFLOAT(st->prop[j]),-15), &st->X[(j+1)*N*K+speak*N], st->E+chan*N, st->PHI, N);
|
|
for (i=0;i<N;i++)
|
|
st->W[chan*N*K*M + j*N*K + speak*N + i] += st->PHI[i];
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
st->saturated--;
|
|
}
|
|
|
|
/* FIXME: MC conversion required */
|
|
/* Update weight to prevent circular convolution (MDF / AUMDF) */
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
for (j=0;j<M;j++)
|
|
{
|
|
/* This is a variant of the Alternatively Updated MDF (AUMDF) */
|
|
/* Remove the "if" to make this an MDF filter */
|
|
if (j==0 || st->cancel_count%(M-1) == j-1)
|
|
{
|
|
#ifdef FIXED_POINT
|
|
for (i=0;i<N;i++)
|
|
st->wtmp2[i] = EXTRACT16(PSHR32(st->W[chan*N*K*M + j*N*K + speak*N + i],NORMALIZE_SCALEDOWN+16));
|
|
spx_ifft(st->fft_table, st->wtmp2, st->wtmp);
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
st->wtmp[i]=0;
|
|
}
|
|
for (i=st->frame_size;i<N;i++)
|
|
{
|
|
st->wtmp[i]=SHL16(st->wtmp[i],NORMALIZE_SCALEUP);
|
|
}
|
|
spx_fft(st->fft_table, st->wtmp, st->wtmp2);
|
|
/* The "-1" in the shift is a sort of kludge that trades less efficient update speed for decrease noise */
|
|
for (i=0;i<N;i++)
|
|
st->W[chan*N*K*M + j*N*K + speak*N + i] -= SHL32(EXTEND32(st->wtmp2[i]),16+NORMALIZE_SCALEDOWN-NORMALIZE_SCALEUP-1);
|
|
#else
|
|
spx_ifft(st->fft_table, &st->W[chan*N*K*M + j*N*K + speak*N], st->wtmp);
|
|
for (i=st->frame_size;i<N;i++)
|
|
{
|
|
st->wtmp[i]=0;
|
|
}
|
|
spx_fft(st->fft_table, st->wtmp, &st->W[chan*N*K*M + j*N*K + speak*N]);
|
|
#endif
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* So we can use power_spectrum_accum */
|
|
for (i=0;i<=st->frame_size;i++)
|
|
st->Rf[i] = st->Yf[i] = st->Xf[i] = 0;
|
|
|
|
Dbf = 0;
|
|
See = 0;
|
|
#ifdef TWO_PATH
|
|
/* Difference in response, this is used to estimate the variance of our residual power estimate */
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
spectral_mul_accum(st->X, st->W+chan*N*K*M, st->Y+chan*N, N, M*K);
|
|
spx_ifft(st->fft_table, st->Y+chan*N, st->y+chan*N);
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->e[chan*N+i] = SUB16(st->e[chan*N+i+st->frame_size], st->y[chan*N+i+st->frame_size]);
|
|
Dbf += 10+mdf_inner_prod(st->e+chan*N, st->e+chan*N, st->frame_size);
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->e[chan*N+i] = SUB16(st->input[chan*st->frame_size+i], st->y[chan*N+i+st->frame_size]);
|
|
See += mdf_inner_prod(st->e+chan*N, st->e+chan*N, st->frame_size);
|
|
}
|
|
#endif
|
|
|
|
#ifndef TWO_PATH
|
|
Sff = See;
|
|
#endif
|
|
|
|
#ifdef TWO_PATH
|
|
/* Logic for updating the foreground filter */
|
|
|
|
/* For two time windows, compute the mean of the energy difference, as well as the variance */
|
|
st->Davg1 = ADD32(MULT16_32_Q15(QCONST16(.6f,15),st->Davg1), MULT16_32_Q15(QCONST16(.4f,15),SUB32(Sff,See)));
|
|
st->Davg2 = ADD32(MULT16_32_Q15(QCONST16(.85f,15),st->Davg2), MULT16_32_Q15(QCONST16(.15f,15),SUB32(Sff,See)));
|
|
st->Dvar1 = FLOAT_ADD(FLOAT_MULT(VAR1_SMOOTH, st->Dvar1), FLOAT_MUL32U(MULT16_32_Q15(QCONST16(.4f,15),Sff), MULT16_32_Q15(QCONST16(.4f,15),Dbf)));
|
|
st->Dvar2 = FLOAT_ADD(FLOAT_MULT(VAR2_SMOOTH, st->Dvar2), FLOAT_MUL32U(MULT16_32_Q15(QCONST16(.15f,15),Sff), MULT16_32_Q15(QCONST16(.15f,15),Dbf)));
|
|
|
|
/* Equivalent float code:
|
|
st->Davg1 = .6*st->Davg1 + .4*(Sff-See);
|
|
st->Davg2 = .85*st->Davg2 + .15*(Sff-See);
|
|
st->Dvar1 = .36*st->Dvar1 + .16*Sff*Dbf;
|
|
st->Dvar2 = .7225*st->Dvar2 + .0225*Sff*Dbf;
|
|
*/
|
|
|
|
update_foreground = 0;
|
|
/* Check if we have a statistically significant reduction in the residual echo */
|
|
/* Note that this is *not* Gaussian, so we need to be careful about the longer tail */
|
|
if (FLOAT_GT(FLOAT_MUL32U(SUB32(Sff,See),ABS32(SUB32(Sff,See))), FLOAT_MUL32U(Sff,Dbf)))
|
|
update_foreground = 1;
|
|
else if (FLOAT_GT(FLOAT_MUL32U(st->Davg1, ABS32(st->Davg1)), FLOAT_MULT(VAR1_UPDATE,(st->Dvar1))))
|
|
update_foreground = 1;
|
|
else if (FLOAT_GT(FLOAT_MUL32U(st->Davg2, ABS32(st->Davg2)), FLOAT_MULT(VAR2_UPDATE,(st->Dvar2))))
|
|
update_foreground = 1;
|
|
|
|
/* Do we update? */
|
|
if (update_foreground)
|
|
{
|
|
st->Davg1 = st->Davg2 = 0;
|
|
st->Dvar1 = st->Dvar2 = FLOAT_ZERO;
|
|
/* Copy background filter to foreground filter */
|
|
for (i=0;i<N*M*C*K;i++)
|
|
st->foreground[i] = EXTRACT16(PSHR32(st->W[i],16));
|
|
/* Apply a smooth transition so as to not introduce blocking artifacts */
|
|
for (chan = 0; chan < C; chan++)
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->e[chan*N+i+st->frame_size] = MULT16_16_Q15(st->window[i+st->frame_size],st->e[chan*N+i+st->frame_size]) + MULT16_16_Q15(st->window[i],st->y[chan*N+i+st->frame_size]);
|
|
} else {
|
|
int reset_background=0;
|
|
/* Otherwise, check if the background filter is significantly worse */
|
|
if (FLOAT_GT(FLOAT_MUL32U(NEG32(SUB32(Sff,See)),ABS32(SUB32(Sff,See))), FLOAT_MULT(VAR_BACKTRACK,FLOAT_MUL32U(Sff,Dbf))))
|
|
reset_background = 1;
|
|
if (FLOAT_GT(FLOAT_MUL32U(NEG32(st->Davg1), ABS32(st->Davg1)), FLOAT_MULT(VAR_BACKTRACK,st->Dvar1)))
|
|
reset_background = 1;
|
|
if (FLOAT_GT(FLOAT_MUL32U(NEG32(st->Davg2), ABS32(st->Davg2)), FLOAT_MULT(VAR_BACKTRACK,st->Dvar2)))
|
|
reset_background = 1;
|
|
if (reset_background)
|
|
{
|
|
/* Copy foreground filter to background filter */
|
|
for (i=0;i<N*M*C*K;i++)
|
|
st->W[i] = SHL32(EXTEND32(st->foreground[i]),16);
|
|
/* We also need to copy the output so as to get correct adaptation */
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->y[chan*N+i+st->frame_size] = st->e[chan*N+i+st->frame_size];
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->e[chan*N+i] = SUB16(st->input[chan*st->frame_size+i], st->y[chan*N+i+st->frame_size]);
|
|
}
|
|
See = Sff;
|
|
st->Davg1 = st->Davg2 = 0;
|
|
st->Dvar1 = st->Dvar2 = FLOAT_ZERO;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
Sey = Syy = Sdd = 0;
|
|
for (chan = 0; chan < C; chan++)
|
|
{
|
|
/* Compute error signal (for the output with de-emphasis) */
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
spx_word32_t tmp_out;
|
|
#ifdef TWO_PATH
|
|
tmp_out = SUB32(EXTEND32(st->input[chan*st->frame_size+i]), EXTEND32(st->e[chan*N+i+st->frame_size]));
|
|
#else
|
|
tmp_out = SUB32(EXTEND32(st->input[chan*st->frame_size+i]), EXTEND32(st->y[chan*N+i+st->frame_size]));
|
|
#endif
|
|
tmp_out = ADD32(tmp_out, EXTEND32(MULT16_16_P15(st->preemph, st->memE[chan])));
|
|
/* This is an arbitrary test for saturation in the microphone signal */
|
|
if (in[i*C+chan] <= -32000 || in[i*C+chan] >= 32000)
|
|
{
|
|
if (st->saturated == 0)
|
|
st->saturated = 1;
|
|
}
|
|
out[i*C+chan] = WORD2INT(tmp_out);
|
|
st->memE[chan] = tmp_out;
|
|
}
|
|
|
|
#ifdef DUMP_ECHO_CANCEL_DATA
|
|
dump_audio(in, far_end, out, st->frame_size);
|
|
#endif
|
|
|
|
/* Compute error signal (filter update version) */
|
|
for (i=0;i<st->frame_size;i++)
|
|
{
|
|
st->e[chan*N+i+st->frame_size] = st->e[chan*N+i];
|
|
st->e[chan*N+i] = 0;
|
|
}
|
|
|
|
/* Compute a bunch of correlations */
|
|
/* FIXME: bad merge */
|
|
Sey += mdf_inner_prod(st->e+chan*N+st->frame_size, st->y+chan*N+st->frame_size, st->frame_size);
|
|
Syy += mdf_inner_prod(st->y+chan*N+st->frame_size, st->y+chan*N+st->frame_size, st->frame_size);
|
|
Sdd += mdf_inner_prod(st->input+chan*st->frame_size, st->input+chan*st->frame_size, st->frame_size);
|
|
|
|
/* Convert error to frequency domain */
|
|
spx_fft(st->fft_table, st->e+chan*N, st->E+chan*N);
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->y[i+chan*N] = 0;
|
|
spx_fft(st->fft_table, st->y+chan*N, st->Y+chan*N);
|
|
|
|
/* Compute power spectrum of echo (X), error (E) and filter response (Y) */
|
|
power_spectrum_accum(st->E+chan*N, st->Rf, N);
|
|
power_spectrum_accum(st->Y+chan*N, st->Yf, N);
|
|
|
|
}
|
|
|
|
/*printf ("%f %f %f %f\n", Sff, See, Syy, Sdd, st->update_cond);*/
|
|
|
|
/* Do some sanity check */
|
|
if (!(Syy>=0 && Sxx>=0 && See >= 0)
|
|
#ifndef FIXED_POINT
|
|
|| !(Sff < N*1e9 && Syy < N*1e9 && Sxx < N*1e9)
|
|
#endif
|
|
)
|
|
{
|
|
/* Things have gone really bad */
|
|
st->screwed_up += 50;
|
|
for (i=0;i<st->frame_size*C;i++)
|
|
out[i] = 0;
|
|
} else if (SHR32(Sff, 2) > ADD32(Sdd, SHR32(MULT16_16(N, 10000),6)))
|
|
{
|
|
/* AEC seems to add lots of echo instead of removing it, let's see if it will improve */
|
|
st->screwed_up++;
|
|
} else {
|
|
/* Everything's fine */
|
|
st->screwed_up=0;
|
|
}
|
|
if (st->screwed_up>=50)
|
|
{
|
|
speex_warning("The echo canceller started acting funny and got slapped (reset). It swears it will behave now.");
|
|
speex_echo_state_reset(st);
|
|
return;
|
|
}
|
|
|
|
/* Add a small noise floor to make sure not to have problems when dividing */
|
|
See = MAX32(See, SHR32(MULT16_16(N, 100),6));
|
|
|
|
for (speak = 0; speak < K; speak++)
|
|
{
|
|
Sxx += mdf_inner_prod(st->x+speak*N+st->frame_size, st->x+speak*N+st->frame_size, st->frame_size);
|
|
power_spectrum_accum(st->X+speak*N, st->Xf, N);
|
|
}
|
|
|
|
|
|
/* Smooth far end energy estimate over time */
|
|
for (j=0;j<=st->frame_size;j++)
|
|
st->power[j] = MULT16_32_Q15(ss_1,st->power[j]) + 1 + MULT16_32_Q15(ss,st->Xf[j]);
|
|
|
|
/* Compute filtered spectra and (cross-)correlations */
|
|
for (j=st->frame_size;j>=0;j--)
|
|
{
|
|
spx_float_t Eh, Yh;
|
|
Eh = PSEUDOFLOAT(st->Rf[j] - st->Eh[j]);
|
|
Yh = PSEUDOFLOAT(st->Yf[j] - st->Yh[j]);
|
|
Pey = FLOAT_ADD(Pey,FLOAT_MULT(Eh,Yh));
|
|
Pyy = FLOAT_ADD(Pyy,FLOAT_MULT(Yh,Yh));
|
|
#ifdef FIXED_POINT
|
|
st->Eh[j] = MAC16_32_Q15(MULT16_32_Q15(SUB16(32767,st->spec_average),st->Eh[j]), st->spec_average, st->Rf[j]);
|
|
st->Yh[j] = MAC16_32_Q15(MULT16_32_Q15(SUB16(32767,st->spec_average),st->Yh[j]), st->spec_average, st->Yf[j]);
|
|
#else
|
|
st->Eh[j] = (1-st->spec_average)*st->Eh[j] + st->spec_average*st->Rf[j];
|
|
st->Yh[j] = (1-st->spec_average)*st->Yh[j] + st->spec_average*st->Yf[j];
|
|
#endif
|
|
}
|
|
|
|
Pyy = FLOAT_SQRT(Pyy);
|
|
Pey = FLOAT_DIVU(Pey,Pyy);
|
|
|
|
/* Compute correlation updatete rate */
|
|
tmp32 = MULT16_32_Q15(st->beta0,Syy);
|
|
if (tmp32 > MULT16_32_Q15(st->beta_max,See))
|
|
tmp32 = MULT16_32_Q15(st->beta_max,See);
|
|
alpha = FLOAT_DIV32(tmp32, See);
|
|
alpha_1 = FLOAT_SUB(FLOAT_ONE, alpha);
|
|
/* Update correlations (recursive average) */
|
|
st->Pey = FLOAT_ADD(FLOAT_MULT(alpha_1,st->Pey) , FLOAT_MULT(alpha,Pey));
|
|
st->Pyy = FLOAT_ADD(FLOAT_MULT(alpha_1,st->Pyy) , FLOAT_MULT(alpha,Pyy));
|
|
if (FLOAT_LT(st->Pyy, FLOAT_ONE))
|
|
st->Pyy = FLOAT_ONE;
|
|
/* We don't really hope to get better than 33 dB (MIN_LEAK-3dB) attenuation anyway */
|
|
if (FLOAT_LT(st->Pey, FLOAT_MULT(MIN_LEAK,st->Pyy)))
|
|
st->Pey = FLOAT_MULT(MIN_LEAK,st->Pyy);
|
|
if (FLOAT_GT(st->Pey, st->Pyy))
|
|
st->Pey = st->Pyy;
|
|
/* leak_estimate is the linear regression result */
|
|
st->leak_estimate = FLOAT_EXTRACT16(FLOAT_SHL(FLOAT_DIVU(st->Pey, st->Pyy),14));
|
|
/* This looks like a stupid bug, but it's right (because we convert from Q14 to Q15) */
|
|
if (st->leak_estimate > 16383)
|
|
st->leak_estimate = 32767;
|
|
else
|
|
st->leak_estimate = SHL16(st->leak_estimate,1);
|
|
/*printf ("%f\n", st->leak_estimate);*/
|
|
|
|
/* Compute Residual to Error Ratio */
|
|
#ifdef FIXED_POINT
|
|
tmp32 = MULT16_32_Q15(st->leak_estimate,Syy);
|
|
tmp32 = ADD32(SHR32(Sxx,13), ADD32(tmp32, SHL32(tmp32,1)));
|
|
/* Check for y in e (lower bound on RER) */
|
|
{
|
|
spx_float_t bound = PSEUDOFLOAT(Sey);
|
|
bound = FLOAT_DIVU(FLOAT_MULT(bound, bound), PSEUDOFLOAT(ADD32(1,Syy)));
|
|
if (FLOAT_GT(bound, PSEUDOFLOAT(See)))
|
|
tmp32 = See;
|
|
else if (tmp32 < FLOAT_EXTRACT32(bound))
|
|
tmp32 = FLOAT_EXTRACT32(bound);
|
|
}
|
|
if (tmp32 > SHR32(See,1))
|
|
tmp32 = SHR32(See,1);
|
|
RER = FLOAT_EXTRACT16(FLOAT_SHL(FLOAT_DIV32(tmp32,See),15));
|
|
#else
|
|
RER = (.0001*Sxx + 3.*MULT16_32_Q15(st->leak_estimate,Syy)) / See;
|
|
/* Check for y in e (lower bound on RER) */
|
|
if (RER < Sey*Sey/(1+See*Syy))
|
|
RER = Sey*Sey/(1+See*Syy);
|
|
if (RER > .5)
|
|
RER = .5;
|
|
#endif
|
|
|
|
/* We consider that the filter has had minimal adaptation if the following is true*/
|
|
if (!st->adapted && st->sum_adapt > SHL32(EXTEND32(M),15) && MULT16_32_Q15(st->leak_estimate,Syy) > MULT16_32_Q15(QCONST16(.03f,15),Syy))
|
|
{
|
|
st->adapted = 1;
|
|
}
|
|
|
|
if (st->adapted)
|
|
{
|
|
/* Normal learning rate calculation once we're past the minimal adaptation phase */
|
|
for (i=0;i<=st->frame_size;i++)
|
|
{
|
|
spx_word32_t r, e;
|
|
/* Compute frequency-domain adaptation mask */
|
|
r = MULT16_32_Q15(st->leak_estimate,SHL32(st->Yf[i],3));
|
|
e = SHL32(st->Rf[i],3)+1;
|
|
#ifdef FIXED_POINT
|
|
if (r>SHR32(e,1))
|
|
r = SHR32(e,1);
|
|
#else
|
|
if (r>.5*e)
|
|
r = .5*e;
|
|
#endif
|
|
r = MULT16_32_Q15(QCONST16(.7,15),r) + MULT16_32_Q15(QCONST16(.3,15),(spx_word32_t)(MULT16_32_Q15(RER,e)));
|
|
/*st->power_1[i] = adapt_rate*r/(e*(1+st->power[i]));*/
|
|
st->power_1[i] = FLOAT_SHL(FLOAT_DIV32_FLOAT(r,FLOAT_MUL32U(e,st->power[i]+10)),WEIGHT_SHIFT+16);
|
|
}
|
|
} else {
|
|
/* Temporary adaption rate if filter is not yet adapted enough */
|
|
spx_word16_t adapt_rate=0;
|
|
|
|
if (Sxx > SHR32(MULT16_16(N, 1000),6))
|
|
{
|
|
tmp32 = MULT16_32_Q15(QCONST16(.25f, 15), Sxx);
|
|
#ifdef FIXED_POINT
|
|
if (tmp32 > SHR32(See,2))
|
|
tmp32 = SHR32(See,2);
|
|
#else
|
|
if (tmp32 > .25*See)
|
|
tmp32 = .25*See;
|
|
#endif
|
|
adapt_rate = FLOAT_EXTRACT16(FLOAT_SHL(FLOAT_DIV32(tmp32, See),15));
|
|
}
|
|
for (i=0;i<=st->frame_size;i++)
|
|
st->power_1[i] = FLOAT_SHL(FLOAT_DIV32(EXTEND32(adapt_rate),ADD32(st->power[i],10)),WEIGHT_SHIFT+1);
|
|
|
|
|
|
/* How much have we adapted so far? */
|
|
st->sum_adapt = ADD32(st->sum_adapt,adapt_rate);
|
|
}
|
|
|
|
/* FIXME: MC conversion required */
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->last_y[i] = st->last_y[st->frame_size+i];
|
|
if (st->adapted)
|
|
{
|
|
/* If the filter is adapted, take the filtered echo */
|
|
for (i=0;i<st->frame_size;i++)
|
|
st->last_y[st->frame_size+i] = in[i]-out[i];
|
|
} else {
|
|
/* If filter isn't adapted yet, all we can do is take the far end signal directly */
|
|
/* moved earlier: for (i=0;i<N;i++)
|
|
st->last_y[i] = st->x[i];*/
|
|
}
|
|
|
|
}
|
|
|
|
/* Compute spectrum of estimated echo for use in an echo post-filter */
|
|
void speex_echo_get_residual(SpeexEchoState *st, spx_word32_t *residual_echo, int len)
|
|
{
|
|
int i;
|
|
spx_word16_t leak2;
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|
int N;
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|
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N = st->window_size;
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|
|
|
/* Apply hanning window (should pre-compute it)*/
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|
for (i=0;i<N;i++)
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st->y[i] = MULT16_16_Q15(st->window[i],st->last_y[i]);
|
|
|
|
/* Compute power spectrum of the echo */
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|
spx_fft(st->fft_table, st->y, st->Y);
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power_spectrum(st->Y, residual_echo, N);
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|
|
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#ifdef FIXED_POINT
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if (st->leak_estimate > 16383)
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|
leak2 = 32767;
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|
else
|
|
leak2 = SHL16(st->leak_estimate, 1);
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#else
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|
if (st->leak_estimate>.5)
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|
leak2 = 1;
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|
else
|
|
leak2 = 2*st->leak_estimate;
|
|
#endif
|
|
/* Estimate residual echo */
|
|
for (i=0;i<=st->frame_size;i++)
|
|
residual_echo[i] = (spx_int32_t)MULT16_32_Q15(leak2,residual_echo[i]);
|
|
|
|
}
|
|
|
|
EXPORT int speex_echo_ctl(SpeexEchoState *st, int request, void *ptr)
|
|
{
|
|
switch(request)
|
|
{
|
|
|
|
case SPEEX_ECHO_GET_FRAME_SIZE:
|
|
(*(int*)ptr) = st->frame_size;
|
|
break;
|
|
case SPEEX_ECHO_SET_SAMPLING_RATE:
|
|
st->sampling_rate = (*(int*)ptr);
|
|
st->spec_average = DIV32_16(SHL32(EXTEND32(st->frame_size), 15), st->sampling_rate);
|
|
#ifdef FIXED_POINT
|
|
st->beta0 = DIV32_16(SHL32(EXTEND32(st->frame_size), 16), st->sampling_rate);
|
|
st->beta_max = DIV32_16(SHL32(EXTEND32(st->frame_size), 14), st->sampling_rate);
|
|
#else
|
|
st->beta0 = (2.0f*st->frame_size)/st->sampling_rate;
|
|
st->beta_max = (.5f*st->frame_size)/st->sampling_rate;
|
|
#endif
|
|
if (st->sampling_rate<12000)
|
|
st->notch_radius = QCONST16(.9, 15);
|
|
else if (st->sampling_rate<24000)
|
|
st->notch_radius = QCONST16(.982, 15);
|
|
else
|
|
st->notch_radius = QCONST16(.992, 15);
|
|
break;
|
|
case SPEEX_ECHO_GET_SAMPLING_RATE:
|
|
(*(int*)ptr) = st->sampling_rate;
|
|
break;
|
|
case SPEEX_ECHO_GET_IMPULSE_RESPONSE_SIZE:
|
|
/*FIXME: Implement this for multiple channels */
|
|
*((spx_int32_t *)ptr) = st->M * st->frame_size;
|
|
break;
|
|
case SPEEX_ECHO_GET_IMPULSE_RESPONSE:
|
|
{
|
|
int M = st->M, N = st->window_size, n = st->frame_size, i, j;
|
|
spx_int32_t *filt = (spx_int32_t *) ptr;
|
|
for(j=0;j<M;j++)
|
|
{
|
|
/*FIXME: Implement this for multiple channels */
|
|
#ifdef FIXED_POINT
|
|
for (i=0;i<N;i++)
|
|
st->wtmp2[i] = EXTRACT16(PSHR32(st->W[j*N+i],16+NORMALIZE_SCALEDOWN));
|
|
spx_ifft(st->fft_table, st->wtmp2, st->wtmp);
|
|
#else
|
|
spx_ifft(st->fft_table, &st->W[j*N], st->wtmp);
|
|
#endif
|
|
for(i=0;i<n;i++)
|
|
filt[j*n+i] = PSHR32(MULT16_16(32767,st->wtmp[i]), WEIGHT_SHIFT-NORMALIZE_SCALEDOWN);
|
|
}
|
|
}
|
|
break;
|
|
default:
|
|
speex_warning_int("Unknown speex_echo_ctl request: ", request);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|