C++ VARDECL函数代码示例

void celt_fir_c(
         const opus_val16 *_x,
         const opus_val16 *num,
         opus_val16 *_y,
         int N,
         int ord,
         opus_val16 *mem,
         int arch)
{
   int i,j;
   VARDECL(opus_val16, rnum);
   VARDECL(opus_val16, x);
   SAVE_STACK;

   ALLOC(rnum, ord, opus_val16);
   ALLOC(x, N+ord, opus_val16);
   for(i=0;i<ord;i++)
      rnum[i] = num[ord-i-1];
   for(i=0;i<ord;i++)
      x[i] = mem[ord-i-1];
   for (i=0;i<N;i++)
      x[i+ord]=_x[i];
   for(i=0;i<ord;i++)
      mem[i] = _x[N-i-1];
#ifdef SMALL_FOOTPRINT
   (void)arch;
   for (i=0;i<N;i++)
   {
      opus_val32 sum = SHL32(EXTEND32(_x[i]), SIG_SHIFT);
      for (j=0;j<ord;j++)
      {
         sum = MAC16_16(sum,rnum[j],x[i+j]);
      }
      _y[i] = SATURATE16(PSHR32(sum, SIG_SHIFT));
   }
#else
   for (i=0;i<N-3;i+=4)
   {
      opus_val32 sum[4]={0,0,0,0};
      xcorr_kernel(rnum, x+i, sum, ord, arch);
      _y[i  ] = SATURATE16(ADD32(EXTEND32(_x[i  ]), PSHR32(sum[0], SIG_SHIFT)));
      _y[i+1] = SATURATE16(ADD32(EXTEND32(_x[i+1]), PSHR32(sum[1], SIG_SHIFT)));
      _y[i+2] = SATURATE16(ADD32(EXTEND32(_x[i+2]), PSHR32(sum[2], SIG_SHIFT)));
      _y[i+3] = SATURATE16(ADD32(EXTEND32(_x[i+3]), PSHR32(sum[3], SIG_SHIFT)));
   }
   for (;i<N;i++)
   {
      opus_val32 sum = 0;
      for (j=0;j<ord;j++)
         sum = MAC16_16(sum,rnum[j],x[i+j]);
      _y[i] = SATURATE16(ADD32(EXTEND32(_x[i]), PSHR32(sum, SIG_SHIFT)));
   }
#endif
   RESTORE_STACK;
}

static opus_val32 downmix_and_resample(downmix_func downmix, const void *_x, opus_val32 *y, opus_val32 S[3], int subframe, int offset, int c1, int c2, int C, int Fs)
{
   VARDECL(opus_val32, tmp);
   opus_val32 scale;
   int j;
   opus_val32 ret = 0;
   SAVE_STACK;

   if (subframe==0) return 0;
   if (Fs == 48000)
   {
      subframe *= 2;
      offset *= 2;
   } else if (Fs == 16000) {
      subframe = subframe*2/3;
      offset = offset*2/3;
   }
   ALLOC(tmp, subframe, opus_val32);

   downmix(_x, tmp, subframe, offset, c1, c2, C);
#ifdef FIXED_POINT
   scale = (1<<SIG_SHIFT);
#else
   scale = 1.f/32768;
#endif
   if (c2==-2)
      scale /= C;
   else if (c2>-1)
      scale /= 2;
   for (j=0;j<subframe;j++)
      tmp[j] *= scale;
   if (Fs == 48000)
   {
      ret = silk_resampler_down2_hp(S, y, tmp, subframe);
   } else if (Fs == 24000) {
      OPUS_COPY(y, tmp, subframe);
   } else if (Fs == 16000) {
      VARDECL(opus_val32, tmp3x);
      ALLOC(tmp3x, 3*subframe, opus_val32);
      /* Don't do this at home! This resampler is horrible and it's only (barely)
         usable for the purpose of the analysis because we don't care about all
         the aliasing between 8 kHz and 12 kHz. */
      for (j=0;j<subframe;j++)
      {
         tmp3x[3*j] = tmp[j];
         tmp3x[3*j+1] = tmp[j];
         tmp3x[3*j+2] = tmp[j];
      }
      silk_resampler_down2_hp(S, y, tmp3x, 3*subframe);
   }
   RESTORE_STACK;
   return ret;
}

int opus_decode_float(OpusDecoder *st, const unsigned char *data,
                      opus_int32 len, float *pcm, int frame_size, int decode_fec)
{
    VARDECL(opus_int16, out);
    int ret, i;
    int nb_samples;
    ALLOC_STACK;

    if(frame_size<=0)
    {
        RESTORE_STACK;
        return OPUS_BAD_ARG;
    }
    if (data != NULL && len > 0 && !decode_fec)
    {
        nb_samples = opus_decoder_get_nb_samples(st, data, len);
        if (nb_samples>0)
            frame_size = IMIN(frame_size, nb_samples);
        else
            return OPUS_INVALID_PACKET;
    }
    ALLOC(out, frame_size*st->channels, opus_int16);

    ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 0);
    if (ret > 0)
    {
        for (i=0; i<ret*st->channels; i++)
            pcm[i] = (1.f/32768.f)*(out[i]);
    }
    RESTORE_STACK;
    return ret;
}

static void compute_weighted_codebook(const signed char *shape_cb, const spx_word16_t *r, spx_word16_t *resp, spx_word16_t *resp2, spx_word32_t *E, int shape_cb_size, int subvect_size, char *stack)
{
   int i, j, k;
   VARDECL(spx_word16_t *shape);
   ALLOC(shape, subvect_size, spx_word16_t);
   for (i=0;i<shape_cb_size;i++)
   {
      spx_word16_t *res;
      
      res = resp+i*subvect_size;
      for (k=0;k<subvect_size;k++)
         shape[k] = (spx_word16_t)shape_cb[i*subvect_size+k];
      E[i]=0;

      /* Compute codeword response using convolution with impulse response */
      for(j=0;j<subvect_size;j++)
      {
         spx_word32_t resj=0;
         spx_word16_t res16;
         for (k=0;k<=j;k++)
            resj = MAC16_16(resj,shape[k],r[j-k]);
#ifdef FIXED_POINT
         res16 = EXTRACT16(SHR32(resj, 13));
#else
         res16 = 0.03125f*resj;
#endif
         /* Compute codeword energy */
         E[i]=MAC16_16(E[i],res16,res16);
         res[j] = res16;
         /*printf ("%d\n", (int)res[j]);*/
      }
   }

}

void noise_codebook_quant(
spx_word16_t target[],			/* target vector */
spx_coef_t ak[],			/* LPCs for this subframe */
spx_coef_t awk1[],			/* Weighted LPCs for this subframe */
spx_coef_t awk2[],			/* Weighted LPCs for this subframe */
const void *par,                      /* Codebook/search parameters*/
int   p,                        /* number of LPC coeffs */
int   nsf,                      /* number of samples in subframe */
spx_sig_t *exc,
spx_word16_t *r,
SpeexBits *bits,
char *stack,
int   complexity,
int   update_target
)
{
   int i;
   VARDECL(spx_word16_t *tmp);
   ALLOC(tmp, nsf, spx_word16_t);
   residue_percep_zero16(target, ak, awk1, awk2, tmp, nsf, p, stack);

   for (i=0;i<nsf;i++)
      exc[i]+=SHL32(EXTEND32(tmp[i]),8);
   SPEEX_MEMSET(target, 0, nsf);
}

static inline spx_word32_t cheb_poly_eva(spx_word32_t *coef,spx_word16_t x,int m,char *stack)
/*  float coef[]  	coefficients of the polynomial to be evaluated 	*/
/*  float x   		the point where polynomial is to be evaluated 	*/
/*  int m 		order of the polynomial 			*/
{
    int i;
    VARDECL(spx_word16_t *T);
    spx_word32_t sum;
    int m2=m>>1;
    VARDECL(spx_word16_t *coefn);

    /*Prevents overflows*/
    if (x>16383)
       x = 16383;
    if (x<-16383)
       x = -16383;

    /* Allocate memory for Chebyshev series formulation */
    ALLOC(T, m2+1, spx_word16_t);
    ALLOC(coefn, m2+1, spx_word16_t);

    for (i=0;i<m2+1;i++)
    {
       coefn[i] = coef[i];
       /*printf ("%f ", coef[i]);*/
    }
    /*printf ("\n");*/

    /* Initialise values */
    T[0]=16384;
    T[1]=x;

    /* Evaluate Chebyshev series formulation using iterative approach  */
    /* Evaluate polynomial and return value also free memory space */
    sum = ADD32(coefn[m2], MULT16_16_P14(coefn[m2-1],x));
    /*x *= 2;*/
    for(i=2;i<=m2;i++)
    {
       T[i] = SUB16(MULT16_16_Q13(x,T[i-1]), T[i-2]);
       sum = ADD32(sum, MULT16_16_P14(coefn[m2-i],T[i]));
       /*printf ("%f ", sum);*/
    }
    
    /*printf ("\n");*/
    return sum;
}

/* Resample with a 2nd order AR filter followed by FIR interpolation */
void silk_resampler_private_down_FIR(
    void                            *SS,            /* I/O  Resampler state             */
    opus_int16                      out[],          /* O    Output signal               */
    const opus_int16                in[],           /* I    Input signal                */
    opus_int32                      inLen           /* I    Number of input samples     */
)
{
    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
    opus_int32 nSamplesIn;
    opus_int32 max_index_Q16, index_increment_Q16;
    VARDECL( opus_int32, buf );
    const opus_int16 *FIR_Coefs;
    SAVE_STACK;

    ALLOC( buf, S->batchSize + S->FIR_Order, opus_int32 );

    /* Copy buffered samples to start of buffer */
    silk_memcpy( buf, S->sFIR.i32, S->FIR_Order * sizeof( opus_int32 ) );

    FIR_Coefs = &S->Coefs[ 2 ];

    /* Iterate over blocks of frameSizeIn input samples */
    index_increment_Q16 = S->invRatio_Q16;
    while( 1 ) {
        nSamplesIn = silk_min( inLen, S->batchSize );

        /* Second-order AR filter (output in Q8) */
        silk_resampler_private_AR2( S->sIIR, &buf[ S->FIR_Order ], in, S->Coefs, nSamplesIn );

        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 );

        /* Interpolate filtered signal */
        out = silk_resampler_private_down_FIR_INTERPOL( out, buf, FIR_Coefs, S->FIR_Order,
            S->FIR_Fracs, max_index_Q16, index_increment_Q16 );

        in += nSamplesIn;
        inLen -= nSamplesIn;

        if( inLen > 1 ) {
            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
            silk_memcpy( buf, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
        } else {
            break;
        }
    }

    /* Copy last part of filtered signal to the state for the next call */
    silk_memcpy( S->sFIR.i32, &buf[ nSamplesIn ], S->FIR_Order * sizeof( opus_int32 ) );
    RESTORE_STACK;
}

int opus_projection_decoder_init(OpusProjectionDecoder *st, opus_int32 Fs,
  int channels, int streams, int coupled_streams,
  unsigned char *demixing_matrix, opus_int32 demixing_matrix_size)
{
  int nb_input_streams;
  opus_int32 expected_matrix_size;
  int i, ret;
  unsigned char mapping[255];
  VARDECL(opus_int16, buf);
  ALLOC_STACK;

  /* Verify supplied matrix size. */
  nb_input_streams = streams + coupled_streams;
  expected_matrix_size = nb_input_streams * channels * sizeof(opus_int16);
  if (expected_matrix_size != demixing_matrix_size)
  {
    RESTORE_STACK;
    return OPUS_BAD_ARG;
  }

  /* Convert demixing matrix input into internal format. */
  ALLOC(buf, nb_input_streams * channels, opus_int16);
  for (i = 0; i < nb_input_streams * channels; i++)
  {
    int s = demixing_matrix[2*i + 1] << 8 | demixing_matrix[2*i];
    s = ((s & 0xFFFF) ^ 0x8000) - 0x8000;
    buf[i] = (opus_int16)s;
  }

  /* Assign demixing matrix. */
  st->demixing_matrix_size_in_bytes =
    mapping_matrix_get_size(channels, nb_input_streams);
  if (!st->demixing_matrix_size_in_bytes)
  {
    RESTORE_STACK;
    return OPUS_BAD_ARG;
  }

  mapping_matrix_init(get_demixing_matrix(st), channels, nb_input_streams, 0,
    buf, demixing_matrix_size);

  /* Set trivial mapping so each input channel pairs with a matrix column. */
  for (i = 0; i < channels; i++)
    mapping[i] = i;

  ret = opus_multistream_decoder_init(
    get_multistream_decoder(st), Fs, channels, streams, coupled_streams, mapping);
  RESTORE_STACK;
  return ret;
}

void _celt_autocorr(
                   const opus_val16 *x,   /*  in: [0...n-1] samples x   */
                   opus_val32       *ac,  /* out: [0...lag-1] ac values */
                   const opus_val16       *window,
                   int          overlap,
                   int          lag,
                   int          n
                  )
{
   opus_val32 d;
   int i;
   VARDECL(opus_val16, xx);
   SAVE_STACK;
   ALLOC(xx, n, opus_val16);
   celt_assert(n>0);
   celt_assert(overlap>=0);
   for (i=0;i<n;i++)
      xx[i] = x[i];
   for (i=0;i<overlap;i++)
   {
      xx[i] = MULT16_16_Q15(x[i],window[i]);
      xx[n-i-1] = MULT16_16_Q15(x[n-i-1],window[i]);
   }
#ifdef FIXED_POINT
   {
      opus_val32 ac0=0;
      int shift;
      for(i=0;i<n;i++)
         ac0 += SHR32(MULT16_16(xx[i],xx[i]),9);
      ac0 += 1+n;

      shift = celt_ilog2(ac0)-30+10;
      shift = (shift+1)/2;
      for(i=0;i<n;i++)
         xx[i] = VSHR32(xx[i], shift);
   }
#endif
   while (lag>=0)
   {
      for (i = lag, d = 0; i < n; i++)
         d += xx[i] * xx[i-lag];
      ac[lag] = d;
      /*printf ("%f ", ac[lag]);*/
      lag--;
   }
   /*printf ("\n");*/
   ac[0] += 10;

   RESTORE_STACK;
}

static int celt_plc_pitch_search(celt_sig *decode_mem[2], int C, int arch)
{
   int pitch_index;
   VARDECL( opus_val16, lp_pitch_buf );
   SAVE_STACK;
   ALLOC( lp_pitch_buf, DECODE_BUFFER_SIZE>>1, opus_val16 );
   pitch_downsample(decode_mem, lp_pitch_buf,
         DECODE_BUFFER_SIZE, C, arch);
   pitch_search(lp_pitch_buf+(PLC_PITCH_LAG_MAX>>1), lp_pitch_buf,
         DECODE_BUFFER_SIZE-PLC_PITCH_LAG_MAX,
         PLC_PITCH_LAG_MAX-PLC_PITCH_LAG_MIN, &pitch_index, arch);
   pitch_index = PLC_PITCH_LAG_MAX-pitch_index;
   RESTORE_STACK;
   return pitch_index;
}

/** Forced pitch delay and gain */
int forced_pitch_quant(
spx_word16_t target[],                 /* Target vector */
spx_word16_t *sw,
spx_coef_t ak[],                     /* LPCs for this subframe */
spx_coef_t awk1[],                   /* Weighted LPCs #1 for this subframe */
spx_coef_t awk2[],                   /* Weighted LPCs #2 for this subframe */
spx_sig_t exc[],                    /* Excitation */
const void *par,
int   start,                    /* Smallest pitch value allowed */
int   end,                      /* Largest pitch value allowed */
spx_word16_t pitch_coef,               /* Voicing (pitch) coefficient */
int   p,                        /* Number of LPC coeffs */
int   nsf,                      /* Number of samples in subframe */
SpeexBits *bits,
char *stack,
spx_word16_t *exc2,
spx_word16_t *r,
int complexity,
int cdbk_offset,
int plc_tuning,
spx_word32_t *cumul_gain
)
{
   int i;
   VARDECL(spx_word16_t *res);
   ALLOC(res, nsf, spx_word16_t);
#ifdef FIXED_POINT
   if (pitch_coef>63)
      pitch_coef=63;
#else
   if (pitch_coef>.99)
      pitch_coef=.99;
#endif
   for (i=0;i<nsf&&i<start;i++)
   {
      exc[i]=MULT16_16(SHL16(pitch_coef, 7),exc2[i-start]);
   }
   for (;i<nsf;i++)
   {
      exc[i]=MULT16_32_Q15(SHL16(pitch_coef, 9),exc[i-start]);
   }
   for (i=0;i<nsf;i++)
      res[i] = EXTRACT16(PSHR32(exc[i], SIG_SHIFT-1));
   syn_percep_zero16(res, ak, awk1, awk2, res, nsf, p, stack);
   for (i=0;i<nsf;i++)
      target[i]=EXTRACT16(SATURATE(SUB32(EXTEND32(target[i]),EXTEND32(res[i])),32700));
   return start;
}

void silk_noise_shape_analysis_FIX(
    silk_encoder_state_FIX          *psEnc,                                 /* I/O  Encoder state FIX                                                           */
    silk_encoder_control_FIX        *psEncCtrl,                             /* I/O  Encoder control FIX                                                         */
    const opus_int16                *pitch_res,                             /* I    LPC residual from pitch analysis                                            */
    const opus_int16                *x,                                     /* I    Input signal [ frame_length + la_shape ]                                    */
    int                              arch                                   /* I    Run-time architecture                                                       */
)
{
    silk_shape_state_FIX *psShapeSt = &psEnc->sShape;
    opus_int     k, i, nSamples, Qnrg, b_Q14, warping_Q16, scale = 0;
    opus_int32   SNR_adj_dB_Q7, HarmBoost_Q16, HarmShapeGain_Q16, Tilt_Q16, tmp32;
    opus_int32   nrg, pre_nrg_Q30, log_energy_Q7, log_energy_prev_Q7, energy_variation_Q7;
    opus_int32   delta_Q16, BWExp1_Q16, BWExp2_Q16, gain_mult_Q16, gain_add_Q16, strength_Q16, b_Q8;
    opus_int32   auto_corr[     MAX_SHAPE_LPC_ORDER + 1 ];
    opus_int32   refl_coef_Q16[ MAX_SHAPE_LPC_ORDER ];
    opus_int32   AR1_Q24[       MAX_SHAPE_LPC_ORDER ];
    opus_int32   AR2_Q24[       MAX_SHAPE_LPC_ORDER ];
    VARDECL( opus_int16, x_windowed );
    const opus_int16 *x_ptr, *pitch_res_ptr;
    SAVE_STACK;

    /* Point to start of first LPC analysis block */
    x_ptr = x - psEnc->sCmn.la_shape;

    /****************/
    /* GAIN CONTROL */
    /****************/
    SNR_adj_dB_Q7 = psEnc->sCmn.SNR_dB_Q7;

    /* Input quality is the average of the quality in the lowest two VAD bands */
    psEncCtrl->input_quality_Q14 = ( opus_int )silk_RSHIFT( (opus_int32)psEnc->sCmn.input_quality_bands_Q15[ 0 ]
        + psEnc->sCmn.input_quality_bands_Q15[ 1 ], 2 );

    /* Coding quality level, between 0.0_Q0 and 1.0_Q0, but in Q14 */
    psEncCtrl->coding_quality_Q14 = silk_RSHIFT( silk_sigm_Q15( silk_RSHIFT_ROUND( SNR_adj_dB_Q7 -
        SILK_FIX_CONST( 20.0, 7 ), 4 ) ), 1 );

    /* Reduce coding SNR during low speech activity */
    if( psEnc->sCmn.useCBR == 0 ) {
        b_Q8 = SILK_FIX_CONST( 1.0, 8 ) - psEnc->sCmn.speech_activity_Q8;
        b_Q8 = silk_SMULWB( silk_LSHIFT( b_Q8, 8 ), b_Q8 );
        SNR_adj_dB_Q7 = silk_SMLAWB( SNR_adj_dB_Q7,
            silk_SMULBB( SILK_FIX_CONST( -BG_SNR_DECR_dB, 7 ) >> ( 4 + 1 ), b_Q8 ),                                       /* Q11*/
            silk_SMULWB( SILK_FIX_CONST( 1.0, 14 ) + psEncCtrl->input_quality_Q14, psEncCtrl->coding_quality_Q14 ) );     /* Q12*/
    }

static int transient_analysis(celt_word32_t *in, int len, int C, int *transient_time, int *transient_shift)
{
   int c, i, n;
   celt_word32_t ratio;
   VARDECL(celt_word32_t, begin);
   SAVE_STACK;
   ALLOC(begin, len, celt_word32_t);
   for (i=0;i<len;i++)
      begin[i] = ABS32(SHR32(in[C*i],SIG_SHIFT));
   for (c=1;c<C;c++)
   {
      for (i=0;i<len;i++)
         begin[i] = MAX32(begin[i], ABS32(SHR32(in[C*i+c],SIG_SHIFT)));
   }
   for (i=1;i<len;i++)
      begin[i] = MAX32(begin[i-1],begin[i]);
   n = -1;
   for (i=8;i<len-8;i++)
   {
      if (begin[i] < MULT16_32_Q15(QCONST16(.2f,15),begin[len-1]))
         n=i;
   }
   if (n<32)
   {
      n = -1;
      ratio = 0;
   } else {
      ratio = DIV32(begin[len-1],1+begin[n-16]);
   }
   if (ratio < 0)
      ratio = 0;
   if (ratio > 1000)
      ratio = 1000;
   ratio *= ratio;
   
   if (ratio > 2048)
      *transient_shift = 3;
   else
      *transient_shift = 0;
   
   *transient_time = n;
   
   RESTORE_STACK;
   return ratio > 20;
}

int opus_decode_float(OpusDecoder *st, const unsigned char *data,
      opus_int32 len, float *pcm, int frame_size, int decode_fec)
{
   VARDECL(opus_int16, out);
   int ret, i;
   ALLOC_STACK;

   ALLOC(out, frame_size*st->channels, opus_int16);

   ret = opus_decode_native(st, data, len, out, frame_size, decode_fec, 0, NULL, 0);
   if (ret > 0)
   {
      for (i=0;i<ret*st->channels;i++)
         pcm[i] = (1.f/32768.f)*(out[i]);
   }
   RESTORE_STACK;
   return ret;
}

/* Upsample using a combination of allpass-based 2x upsampling and FIR interpolation */
void silk_resampler_private_IIR_FIR(
    void                            *SS,            /* I/O  Resampler state             */
    opus_int16                      out[],          /* O    Output signal               */
    const opus_int16                in[],           /* I    Input signal                */
    opus_int32                      inLen           /* I    Number of input samples     */
)
{
    silk_resampler_state_struct *S = (silk_resampler_state_struct *)SS;
    opus_int32 nSamplesIn;
    opus_int32 max_index_Q16, index_increment_Q16;
    VARDECL( opus_int16, buf );
    SAVE_STACK;

    ALLOC( buf, 2 * S->batchSize + RESAMPLER_ORDER_FIR_12, opus_int16 );

    /* Copy buffered samples to start of buffer */
    silk_memcpy( buf, S->sFIR.i16, RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );

    /* Iterate over blocks of frameSizeIn input samples */
    index_increment_Q16 = S->invRatio_Q16;
    while( 1 ) {
        nSamplesIn = silk_min( inLen, S->batchSize );

        /* Upsample 2x */
        silk_resampler_private_up2_HQ( S->sIIR, &buf[ RESAMPLER_ORDER_FIR_12 ], in, nSamplesIn );

        max_index_Q16 = silk_LSHIFT32( nSamplesIn, 16 + 1 );         /* + 1 because 2x upsampling */
        out = silk_resampler_private_IIR_FIR_INTERPOL( out, buf, max_index_Q16, index_increment_Q16 );
        in += nSamplesIn;
        inLen -= nSamplesIn;

        if( inLen > 0 ) {
            /* More iterations to do; copy last part of filtered signal to beginning of buffer */
            silk_memcpy( buf, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
        } else {
            break;
        }
    }

    /* Copy last part of filtered signal to the state for the next call */
    silk_memcpy( S->sFIR.i16, &buf[ nSamplesIn << 1 ], RESAMPLER_ORDER_FIR_12 * sizeof( opus_int16 ) );
    RESTORE_STACK;
}