예제 #1
0
/** Performs echo cancellation on a frame */
void speex_echo_cancel(SpeexEchoState *st, short *ref, short *echo, short *out, float *Yout)
{
    int i,j,m;
    int N,M;
    float scale;
    float ESR;
    float SER;
    float Sry=0,Srr=0,Syy=0,Sey=0,See=0,Sxx=0;
    float leak_estimate;

    leak_estimate = .1+(.9/(1+2*st->sum_adapt));

    N = st->window_size;
    M = st->M;
    scale = 1.0f/N;
    st->cancel_count++;

    /* Copy input data to buffer */
    for (i=0; i<st->frame_size; i++)
    {
        st->x[i] = st->x[i+st->frame_size];
        st->x[i+st->frame_size] = echo[i];

        st->d[i] = st->d[i+st->frame_size];
        st->d[i+st->frame_size] = ref[i];
    }

    /* Shift memory: this could be optimized eventually*/
    for (i=0; i<N*(M-1); i++)
        st->X[i]=st->X[i+N];

    /* Copy new echo frame */
    for (i=0; i<N; i++)
        st->X[(M-1)*N+i]=st->x[i];

    /* Convert x (echo input) to frequency domain */
    spx_drft_forward(st->fft_lookup, &st->X[(M-1)*N]);

    /* Compute filter response Y */
    for (i=0; i<N; i++)
        st->Y[i] = 0;
    for (j=0; j<M; j++)
        spectral_mul_accum(&st->X[j*N], &st->W[j*N], st->Y, N);

    /* Convert Y (filter response) to time domain */
    for (i=0; i<N; i++)
        st->y[i] = st->Y[i];
    spx_drft_backward(st->fft_lookup, st->y);
    for (i=0; i<N; i++)
        st->y[i] *= scale;

    /* Transform d (reference signal) to frequency domain */
    for (i=0; i<N; i++)
        st->D[i]=st->d[i];
    spx_drft_forward(st->fft_lookup, st->D);

    /* Compute error signal (signal with echo removed) */
    for (i=0; i<st->frame_size; i++)
    {
        float tmp_out;
        tmp_out = (float)ref[i] - st->y[i+st->frame_size];

        st->E[i] = 0;
        st->E[i+st->frame_size] = tmp_out;

        /* Saturation */
        if (tmp_out>32767)
            tmp_out = 32767;
        else if (tmp_out<-32768)
            tmp_out = -32768;
        out[i] = tmp_out;
    }

    /* This bit of code provides faster adaptation by doing a projection of the previous gradient on the
       "MMSE surface" */
    if (1)
    {
        float Sge, Sgg, Syy;
        float gain;
        Syy = inner_prod(st->y+st->frame_size, st->y+st->frame_size, st->frame_size);
        for (i=0; i<N; i++)
            st->Y2[i] = 0;
        for (j=0; j<M; j++)
            spectral_mul_accum(&st->X[j*N], &st->PHI[j*N], st->Y2, N);
        for (i=0; i<N; i++)
            st->y2[i] = st->Y2[i];
        spx_drft_backward(st->fft_lookup, st->y2);
        for (i=0; i<N; i++)
            st->y2[i] *= scale;
        Sge = inner_prod(st->y2+st->frame_size, st->E+st->frame_size, st->frame_size);
        Sgg = inner_prod(st->y2+st->frame_size, st->y2+st->frame_size, st->frame_size);
        /* Compute projection gain */
        gain = Sge/(N+.03*Syy+Sgg);
        if (gain>2)
            gain = 2;
        if (gain < -2)
            gain = -2;

        /* Apply gain to weights, echo estimates, output */
        for (i=0; i<N; i++)
            st->Y[i] += gain*st->Y2[i];
        for (i=0; i<st->frame_size; i++)
        {
            st->y[i+st->frame_size] += gain*st->y2[i+st->frame_size];
            st->E[i+st->frame_size] -= gain*st->y2[i+st->frame_size];
        }
        for (i=0; i<M*N; i++)
            st->W[i] += gain*st->PHI[i];
    }

    /* Compute power spectrum of output (D-Y) and filter response (Y) */
    for (i=0; i<N; i++)
        st->D[i] -= st->Y[i];
    power_spectrum(st->D, st->Rf, N);
    power_spectrum(st->Y, st->Yf, N);

    /* Compute frequency-domain adaptation mask */
    for (j=0; j<=st->frame_size; j++)
    {
        float r;
        r = leak_estimate*st->Yf[j] / (1+st->Rf[j]);
        if (r>1)
            r = 1;
        st->fratio[j] = r;
    }

    /* Compute a bunch of correlations */
    Sry = inner_prod(st->y+st->frame_size, st->d+st->frame_size, st->frame_size);
    Sey = inner_prod(st->y+st->frame_size, st->E+st->frame_size, st->frame_size);
    See = inner_prod(st->E+st->frame_size, st->E+st->frame_size, st->frame_size);
    Syy = inner_prod(st->y+st->frame_size, st->y+st->frame_size, st->frame_size);
    Srr = inner_prod(st->d+st->frame_size, st->d+st->frame_size, st->frame_size);
    Sxx = inner_prod(st->x+st->frame_size, st->x+st->frame_size, st->frame_size);

    /* Compute smoothed cross-correlation and energy */
    st->Sey = .98*st->Sey + .02*Sey;
    st->Syy = .98*st->Syy + .02*Syy;
    st->See = .98*st->See + .02*See;

    /* Check if filter is completely mis-adapted (if so, reset filter) */
    if (st->Sey/(1+st->Syy + .01*st->See) < -1)
    {
        /*fprintf (stderr, "reset at %d\n", st->cancel_count);*/
        speex_echo_state_reset(st);
        return;
    }

    SER = Srr / (1+Sxx);
    ESR = leak_estimate*Syy / (1+See);
    if (ESR>1)
        ESR = 1;
#if 1
    /* If over-cancellation (creating echo with 180 phase) damp filter */
    if (st->Sey/(1+st->Syy) < -.1 && (ESR > .3))
    {
        for (i=0; i<M*N; i++)
            st->W[i] *= .95;
        st->Sey *= .5;
        /*fprintf (stderr, "corrected down\n");*/
    }
#endif
#if 1
    /* If under-cancellation (leaving echo with 0 phase) scale filter up */
    if (st->Sey/(1+st->Syy) > .1 && (ESR > .1 || SER < 10))
    {
        for (i=0; i<M*N; i++)
            st->W[i] *= 1.05;
        st->Sey *= .5;
        /*fprintf (stderr, "corrected up %d\n", st->cancel_count);*/
    }
#endif

    /* We consider that the filter is adapted if the following is true*/
    if (ESR>.6 && st->sum_adapt > 1)
    {
        /*if (!st->adapted)
           fprintf(stderr, "Adapted at %d %f\n", st->cancel_count, st->sum_adapt);*/
        st->adapted = 1;
    }

    /* Update frequency-dependent energy ratio with the total energy ratio */
    for (i=0; i<=st->frame_size; i++)
    {
        st->fratio[i]  = (.2*ESR+.8*min(.005+ESR,st->fratio[i]));
    }

    if (st->adapted)
    {
        st->adapt_rate = .95f/(2+M);
    } else {
        /* Temporary adaption rate if filter is not adapted correctly */
        if (SER<.1)
            st->adapt_rate =.8/(2+M);
        else if (SER<1)
            st->adapt_rate =.4/(2+M);
        else if (SER<10)
            st->adapt_rate =.2/(2+M);
        else if (SER<30)
            st->adapt_rate =.08/(2+M);
        else
            st->adapt_rate = 0;
    }

    /* How much have we adapted so far? */
    st->sum_adapt += st->adapt_rate;

    /* Compute echo power in each frequency bin */
    {
        float ss = 1.0f/st->cancel_count;
        if (ss < .3/M)
            ss=.3/M;
        power_spectrum(&st->X[(M-1)*N], st->Xf, N);
        /* Smooth echo energy estimate over time */
        for (j=0; j<=st->frame_size; j++)
            st->power[j] = (1-ss)*st->power[j] + ss*st->Xf[j];


        /* Combine adaptation rate to the the inverse energy estimate */
        if (st->adapted)
        {
            /* If filter is adapted, include the frequency-dependent ratio too */
            for (i=0; i<=st->frame_size; i++)
                st->power_1[i] = st->adapt_rate*st->fratio[i] /(1.f+st->power[i]);
        } else {
            for (i=0; i<=st->frame_size; i++)
                st->power_1[i] = st->adapt_rate/(1.f+st->power[i]);
        }
    }


    /* Convert error to frequency domain */
    spx_drft_forward(st->fft_lookup, st->E);

    /* Do some regularization (prevents problems when system is ill-conditoned) */
    for (m=0; m<M; m++)
        for (i=0; i<N; i++)
            st->W[m*N+i] *= 1-st->regul[i]*ESR;

    /* Compute weight gradient */
    for (j=0; j<M; j++)
    {
        weighted_spectral_mul_conj(st->power_1, &st->X[j*N], st->E, st->PHI+N*j, N);
    }

    /* Gradient descent */
    for (i=0; i<M*N; i++)
        st->W[i] += st->PHI[i];

    /* AUMDF weight constraint */
    for (j=0; j<M; j++)
    {
        /* Remove the "if" to make this an MDF filter */
        if (st->cancel_count%M == j)
        {
            spx_drft_backward(st->fft_lookup, &st->W[j*N]);
            for (i=0; i<N; i++)
                st->W[j*N+i]*=scale;
            for (i=st->frame_size; i<N; i++)
            {
                st->W[j*N+i]=0;
            }
            spx_drft_forward(st->fft_lookup, &st->W[j*N]);
        }
    }

    /* Compute spectrum of estimated echo for use in an echo post-filter (if necessary)*/
    if (Yout)
    {
        if (st->adapted)
        {
            /* If the filter is adapted, take the filtered echo */
            for (i=0; i<st->frame_size; i++)
                st->last_y[i] = st->last_y[st->frame_size+i];
            for (i=0; i<st->frame_size; i++)
                st->last_y[st->frame_size+i] = st->y[st->frame_size+i];
        } else {
            /* If filter isn't adapted yet, all we can do is take the echo signal directly */
            for (i=0; i<N; i++)
                st->last_y[i] = st->x[i];
        }

        /* Apply hanning window (should pre-compute it)*/
        for (i=0; i<N; i++)
            st->Yps[i] = (.5-.5*cos(2*M_PI*i/N))*st->last_y[i];

        /* Compute power spectrum of the echo */
        spx_drft_forward(st->fft_lookup, st->Yps);
        power_spectrum(st->Yps, st->Yps, N);

        /* Estimate residual echo */
        for (i=0; i<=st->frame_size; i++)
            Yout[i] = 2*leak_estimate*st->Yps[i];
    }

}
예제 #2
0
파일: mdf.c 프로젝트: FreshLeaf8865/mumble
/** 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];*/
   }

}