/* *=================================================================== * 3GPP AMR Wideband Floating-point Speech Codec *=================================================================== */ #include #include #include "typedef.h" #include "dec_main.h" #include "dec_lpc.h" #define MAX_16 (Word16)0x7FFF #define MIN_16 (Word16)0x8000 #define L_SUBFR 64 /* Subframe size */ #define L_SUBFR16k 80 /* Subframe size at 16kHz */ #define M16k 20 /* Order of LP filter */ #define PREEMPH_FAC 22282 /* preemphasis factor (0.68 in Q15) */ #define FAC4 4 #define FAC5 5 #define UP_FAC 20480 /* 5/4 in Q14 */ #define INV_FAC5 6554 /* 1/5 in Q15 */ #define NB_COEF_UP 12 #define L_FIR 31 #define MODE_7k 0 #define MODE_24k 8 extern const Word16 D_ROM_pow2[]; extern const Word16 D_ROM_isqrt[]; extern const Word16 D_ROM_log2[]; extern const Word16 D_ROM_fir_up[]; extern const Word16 D_ROM_fir_6k_7k[]; extern const Word16 D_ROM_fir_7k[]; extern const Word16 D_ROM_hp_gain[]; #ifdef WIN32 #pragma warning( disable : 4310) #endif /* * D_UTIL_random * * Parameters: * seed I/O: seed for random number * * Function: * Signed 16 bits random generator. * * Returns: * random number */ Word16 D_UTIL_random(Word16 *seed) { /*static Word16 seed = 21845;*/ *seed = (Word16)(*seed * 31821L + 13849L); return(*seed); } /* * D_UTIL_pow2 * * Parameters: * exponant I: (Q0) Integer part. (range: 0 <= val <= 30) * fraction I: (Q15) Fractionnal part. (range: 0.0 <= val < 1.0) * * Function: * L_x = pow(2.0, exponant.fraction) (exponant = interger part) * = pow(2.0, 0.fraction) << exponant * * Algorithm: * * The function Pow2(L_x) is approximated by a table and linear * interpolation. * * 1 - i = bit10 - b15 of fraction, 0 <= i <= 31 * 2 - a = bit0 - b9 of fraction * 3 - L_x = table[i] << 16 - (table[i] - table[i + 1]) * a * 2 * 4 - L_x = L_x >> (30-exponant) (with rounding) * * Returns: * range 0 <= val <= 0x7fffffff */ Word32 D_UTIL_pow2(Word16 exponant, Word16 fraction) { Word32 L_x, tmp, i, exp; Word16 a; L_x = fraction * 32; /* L_x = fraction<<6 */ i = L_x >> 15; /* Extract b10-b16 of fraction */ a = (Word16)(L_x); /* Extract b0-b9 of fraction */ a = (Word16)(a & (Word16)0x7fff); L_x = D_ROM_pow2[i] << 16; /* table[i] << 16 */ tmp = D_ROM_pow2[i] - D_ROM_pow2[i + 1]; /* table[i] - table[i+1] */ tmp = L_x - ((tmp * a) << 1); /* L_x -= tmp*a*2 */ exp = 30 - exponant; if (exp <= 31) { L_x = tmp >> exp; if ((1 << (exp - 1)) & tmp) { L_x++; } } else { L_x = 0; } return(L_x); } /* * D_UTIL_norm_l * * Parameters: * L_var1 I: 32 bit Word32 signed integer (Word32) whose value * falls in the range 0x8000 0000 <= var1 <= 0x7fff ffff. * * Function: * Produces the number of left shifts needed to normalize the 32 bit * variable L_var1 for positive values on the interval with minimum of * 1073741824 and maximum of 2147483647, and for negative values on * the interval with minimum of -2147483648 and maximum of -1073741824; * in order to normalize the result, the following operation must be done : * norm_L_var1 = L_shl(L_var1,norm_l(L_var1)). * * Returns: * 16 bit Word16 signed integer (Word16) whose value falls in the range * 0x0000 0000 <= var_out <= 0x0000 001f. */ Word16 D_UTIL_norm_l(Word32 L_var1) { Word16 var_out; if(L_var1 == 0) { var_out = 0; } else { if(L_var1 == (Word32)0xffffffffL) { var_out = 31; } else { if(L_var1 < 0) { L_var1 = ~L_var1; } for(var_out = 0; L_var1 < (Word32)0x40000000L; var_out++) { L_var1 <<= 1; } } } return(var_out); } /* * D_UTIL_norm_s * * Parameters: * L_var1 I: 32 bit Word32 signed integer (Word32) whose value * falls in the range 0xffff 8000 <= var1 <= 0x0000 7fff. * * Function: * Produces the number of left shift needed to normalize the 16 bit * variable var1 for positive values on the interval with minimum * of 16384 and maximum of 32767, and for negative values on * the interval with minimum of -32768 and maximum of -16384. * * Returns: * 16 bit Word16 signed integer (Word16) whose value falls in the range * 0x0000 0000 <= var_out <= 0x0000 000f. */ Word16 D_UTIL_norm_s(Word16 var1) { Word16 var_out; if(var1 == 0) { var_out = 0; } else { if(var1 == -1) { var_out = 15; } else { if(var1 < 0) { var1 = (Word16)~var1; } for(var_out = 0; var1 < 0x4000; var_out++) { var1 <<= 1; } } } return(var_out); } /* * D_UTIL_dot_product12 * * Parameters: * x I: 12bit x vector * y I: 12bit y vector * lg I: vector length * exp O: exponent of result (0..+30) * * Function: * Compute scalar product of using accumulator. * The result is normalized (in Q31) with exponent (0..30). * * Returns: * Q31 normalised result (1 < val <= -1) */ Word32 D_UTIL_dot_product12(Word16 x[], Word16 y[], Word16 lg, Word16 *exp) { Word32 sum, i, sft; sum = 0L; for(i = 0; i < lg; i++) { sum += x[i] * y[i]; } sum = (sum << 1) + 1; /* Normalize acc in Q31 */ sft = D_UTIL_norm_l(sum); sum = sum << sft; *exp = (Word16)(30 - sft); /* exponent = 0..30 */ return(sum); } /* * D_UTIL_normalised_inverse_sqrt * * Parameters: * frac I/O: (Q31) normalized value (1.0 < frac <= 0.5) * exp I/O: exponent (value = frac x 2^exponent) * * Function: * Compute 1/sqrt(value). * If value is negative or zero, result is 1 (frac=7fffffff, exp=0). * * The function 1/sqrt(value) is approximated by a table and linear * interpolation. * 1. If exponant is odd then shift fraction right once. * 2. exponant = -((exponant - 1) >> 1) * 3. i = bit25 - b30 of fraction, 16 <= i <= 63 ->because of normalization. * 4. a = bit10 - b24 * 5. i -= 16 * 6. fraction = table[i]<<16 - (table[i] - table[i+1]) * a * 2 * * Returns: * void */ void D_UTIL_normalised_inverse_sqrt(Word32 *frac, Word16 *exp) { Word32 i, tmp; Word16 a; if(*frac <= (Word32)0) { *exp = 0; *frac = 0x7fffffffL; return; } if((*exp & 0x1) == 1) /* If exponant odd -> shift right */ { *frac = *frac >> 1; } *exp = (Word16)(-((*exp - 1) >> 1)); *frac = *frac >> 9; i = *frac >>16; /* Extract b25-b31 */ *frac = *frac >> 1; a = (Word16)(*frac); /* Extract b10-b24 */ a = (Word16)(a & (Word16)0x7fff); i = i - 16; *frac = D_ROM_isqrt[i] << 16; /* table[i] << 16 */ tmp = D_ROM_isqrt[i] - D_ROM_isqrt[i + 1]; /* table[i] - table[i+1]) */ *frac = *frac - ((tmp * a) << 1); /* frac -= tmp*a*2 */ return; } /* * D_UTIL_inverse_sqrt * * Parameters: * L_x I/O: (Q0) input value (range: 0<=val<=7fffffff) * * Function: * Compute 1/sqrt(L_x). * If value is negative or zero, result is 1 (7fffffff). * * The function 1/sqrt(value) is approximated by a table and linear * interpolation. * 1. Normalization of L_x * 2. call Normalised_Inverse_sqrt(L_x, exponant) * 3. L_y = L_x << exponant * * Returns: * (Q31) output value (range: 0 <= val < 1) */ Word32 D_UTIL_inverse_sqrt(Word32 L_x) { Word32 L_y; Word16 exp; exp = D_UTIL_norm_l(L_x); L_x = (L_x << exp); /* L_x is normalized */ exp = (Word16)(31 - exp); D_UTIL_normalised_inverse_sqrt(&L_x, &exp); if(exp < 0) { L_y = (L_x >> -exp); /* denormalization */ } else { L_y = (L_x << exp); /* denormalization */ } return(L_y); } /* * D_UTIL_normalised_log2 * * Parameters: * L_x I: input value (normalized) * exp I: norm_l (L_x) * exponent O: Integer part of Log2. (range: 0<=val<=30) * fraction O: Fractional part of Log2. (range: 0<=val<1) * * Function: * Computes log2(L_x, exp), where L_x is positive and * normalized, and exp is the normalisation exponent * If L_x is negative or zero, the result is 0. * * The function Log2(L_x) is approximated by a table and linear * interpolation. The following steps are used to compute Log2(L_x) * * 1. exponent = 30 - norm_exponent * 2. i = bit25 - b31 of L_x; 32 <= i <= 63 (because of normalization). * 3. a = bit10 - b24 * 4. i -= 32 * 5. fraction = table[i] << 16 - (table[i] - table[i + 1]) * a * 2 * * * Returns: * void */ static void D_UTIL_normalised_log2(Word32 L_x, Word16 exp, Word16 *exponent, Word16 *fraction) { Word32 i, a, tmp; Word32 L_y; if (L_x <= 0) { *exponent = 0; *fraction = 0; return; } *exponent = (Word16)(30 - exp); L_x = L_x >> 10; i = L_x >> 15; /* Extract b25-b31 */ a = L_x; /* Extract b10-b24 of fraction */ a = a & 0x00007fff; i = i - 32; L_y = D_ROM_log2[i] << 16; /* table[i] << 16 */ tmp = D_ROM_log2[i] - D_ROM_log2[i + 1]; /* table[i] - table[i+1] */ L_y = L_y - ((tmp * a) << 1); /* L_y -= tmp*a*2 */ *fraction = (Word16)(L_y >> 16); return; } /* * D_UTIL_log2 * * Parameters: * L_x I: input value * exponent O: Integer part of Log2. (range: 0<=val<=30) * fraction O: Fractional part of Log2. (range: 0<=val<1) * * Function: * Computes log2(L_x), where L_x is positive. * If L_x is negative or zero, the result is 0. * * Returns: * void */ void D_UTIL_log2(Word32 L_x, Word16 *exponent, Word16 *fraction) { Word16 exp; exp = D_UTIL_norm_l(L_x); D_UTIL_normalised_log2((L_x <>1 * * Function: * Extract from a 32 bit integer two 16 bit DPF. * * Returns: * void */ void D_UTIL_l_extract(Word32 L_32, Word16 *hi, Word16 *lo) { *hi = (Word16)(L_32 >> 16); *lo = (Word16)((L_32 >> 1) - (*hi * 32768)); return; } /* * D_UTIL_mpy_32_16 * * Parameters: * hi I: hi part of 32 bit number * lo I: lo part of 32 bit number * n I: 16 bit number * * Function: * Multiply a 16 bit integer by a 32 bit (DPF). The result is divided * by 2^15. * * L_32 = (hi1*lo2)<<1 + ((lo1*lo2)>>15)<<1 * * Returns: * 32 bit result */ Word32 D_UTIL_mpy_32_16(Word16 hi, Word16 lo, Word16 n) { Word32 L_32; L_32 = hi * n; L_32 += (lo * n) >> 15; return(L_32 << 1); } /* * D_UTIL_mpy_32 * * Parameters: * hi1 I: hi part of first number * lo1 I: lo part of first number * hi2 I: hi part of second number * lo2 I: lo part of second number * * Function: * Multiply two 32 bit integers (DPF). The result is divided by 2^31 * * L_32 = (hi1*lo2)<<1 + ((lo1*lo2)>>15)<<1 * * Returns: * 32 bit result */ Word32 D_UTIL_mpy_32(Word16 hi1, Word16 lo1, Word16 hi2, Word16 lo2) { Word32 L_32; L_32 = hi1 * hi2; L_32 += (hi1 * lo2) >> 15; L_32 += (lo1 * hi2) >> 15; return(L_32 << 1); } /* * D_UTIL_saturate * * Parameters: * inp I: 32-bit number * * Function: * Saturation to 16-bit number * * Returns: * 16-bit number */ Word16 D_UTIL_saturate(Word32 inp) { Word16 out; if ((inp < MAX_16) & (inp > MIN_16)) { out = (Word16)inp; } else { if (inp > 0) { out = MAX_16; } else { out = MIN_16; } } return(out); } /* * D_UTIL_signal_up_scale * * Parameters: * x I/O: signal to scale * lg I: size of x[] * exp I: exponent: x = round(x << exp) * * Function: * Scale signal up to get maximum of dynamic. * * Returns: * 32 bit result */ void D_UTIL_signal_up_scale(Word16 x[], Word16 lg, Word16 exp) { Word32 i, tmp; for (i = 0; i < lg; i++) { tmp = x[i] << exp; x[i] = D_UTIL_saturate(tmp); } return; } /* * D_UTIL_signal_down_scale * * Parameters: * x I/O: signal to scale * lg I: size of x[] * exp I: exponent: x = round(x << exp) * * Function: * Scale signal up to get maximum of dynamic. * * Returns: * 32 bit result */ void D_UTIL_signal_down_scale(Word16 x[], Word16 lg, Word16 exp) { Word32 i, tmp; for(i = 0; i < lg; i++) { tmp = x[i] << 16; tmp = tmp >> exp; x[i] = (Word16)((tmp + 0x8000) >> 16); } return; } /* * D_UTIL_deemph_32 * * Parameters: * x_hi I: input signal (bit31..16) * x_lo I: input signal (bit15..4) * y O: output signal (x16) * mu I: (Q15) deemphasis factor * L I: vector size * mem I/O: memory (y[-1]) * * Function: * Filtering through 1/(1-mu z^-1) * * Returns: * void */ static void D_UTIL_deemph_32(Word16 x_hi[], Word16 x_lo[], Word16 y[], Word16 mu, Word16 L, Word16 *mem) { Word32 i, fac; Word32 tmp; fac = mu >> 1; /* Q15 --> Q14 */ /* L_tmp = hi<<16 + lo<<4 */ tmp = (x_hi[0] << 12) + x_lo[0]; tmp = (tmp << 6) + (*mem * fac); tmp = (tmp + 0x2000) >> 14; y[0] = D_UTIL_saturate(tmp); for(i = 1; i < L; i++) { tmp = (x_hi[i] << 12) + x_lo[i]; tmp = (tmp << 6) + (y[i - 1] * fac); tmp = (tmp + 0x2000) >> 14; y[i] = D_UTIL_saturate(tmp); } *mem = y[L - 1]; return; } /* * D_UTIL_synthesis_32 * * Parameters: * a I: LP filter coefficients * m I: order of LP filter * exc I: excitation * Qnew I: exc scaling = 0(min) to 8(max) * sig_hi O: synthesis high * sig_lo O: synthesis low * lg I: size of filtering * * Function: * Perform the synthesis filtering 1/A(z). * * Returns: * void */ static void D_UTIL_synthesis_32(Word16 a[], Word16 m, Word16 exc[], Word16 Qnew, Word16 sig_hi[], Word16 sig_lo[], Word16 lg) { Word32 i, j, a0, s; Word32 tmp, tmp2; /* See if a[0] is scaled */ s = D_UTIL_norm_s((Word16)a[0]) - 2; a0 = a[0] >> (4 + Qnew); /* input / 16 and >>Qnew */ /* Do the filtering. */ for(i = 0; i < lg; i++) { tmp = 0; for(j = 1; j <= m; j++) { tmp -= sig_lo[i - j] * a[j]; } tmp = tmp >> (15 - 4); /* -4 : sig_lo[i] << 4 */ tmp2 = exc[i] * a0; for(j = 1; j <= m; j++) { tmp2 -= sig_hi[i - j] * a[j]; } tmp += tmp2 << 1; tmp <<= s; /* sig_hi = bit16 to bit31 of synthesis */ sig_hi[i] = (Word16)(tmp >> 13); /* sig_lo = bit4 to bit15 of synthesis */ sig_lo[i] = (Word16)((tmp >> 1) - (sig_hi[i] * 4096)); } return; } /* * D_UTIL_hp50_12k8 * * Parameters: * signal I/O: signal * lg I: lenght of signal * mem I/O: filter memory [6] * * Function: * 2nd order high pass filter with cut off frequency at 50 Hz. * * Algorithm: * * y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2] * + a[1]*y[i-1] + a[2]*y[i-2]; * * b[3] = {0.989501953f, -1.979003906f, 0.989501953f}; * a[3] = {1.000000000F, 1.978881836f,-0.966308594f}; * * * Returns: * void */ static void D_UTIL_hp50_12k8(Word16 signal[], Word16 lg, Word16 mem[]) { Word32 i, L_tmp; Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1, x2; y2_hi = mem[0]; y2_lo = mem[1]; y1_hi = mem[2]; y1_lo = mem[3]; x0 = mem[4]; x1 = mem[5]; for(i = 0; i < lg; i++) { x2 = x1; x1 = x0; x0 = signal[i]; /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */ /* + a[1]*y[i-1] + a[2] * y[i-2]; */ L_tmp = 8192L; /* rounding to maximise precision */ L_tmp = L_tmp + (y1_lo * 16211); L_tmp = L_tmp + (y2_lo * (-8021)); L_tmp = L_tmp >> 14; L_tmp = L_tmp + (y1_hi * 32422); L_tmp = L_tmp + (y2_hi * (-16042)); L_tmp = L_tmp + (x0 * 8106); L_tmp = L_tmp + (x1 * (-16212)); L_tmp = L_tmp + (x2 * 8106); L_tmp = L_tmp << 2; /* coeff Q11 --> Q14 */ y2_hi = y1_hi; y2_lo = y1_lo; D_UTIL_l_extract(L_tmp, &y1_hi, &y1_lo); L_tmp = (L_tmp + 0x4000) >> 15; /* coeff Q14 --> Q15 with saturation */ signal[i] = D_UTIL_saturate(L_tmp); } mem[0] = y2_hi; mem[1] = y2_lo; mem[2] = y1_hi; mem[3] = y1_lo; mem[4] = x0; mem[5] = x1; return; } /* * D_UTIL_interpol * * Parameters: * x I: input vector * fir I: filter coefficient * frac I: fraction (0..resol) * up_samp I: resolution * nb_coef I: number of coefficients * * Function: * Fractional interpolation of signal at position (frac/up_samp) * * Returns: * result of interpolation */ Word16 D_UTIL_interpol(Word16 *x, Word16 const *fir, Word16 frac, Word16 resol, Word16 nb_coef) { Word32 i, k; Word32 sum; x = x - nb_coef + 1; sum = 0L; for(i = 0, k = ((resol - 1) - frac); i < 2 * nb_coef; i++, k = (Word16)(k + resol)) { sum = sum + (x[i] * fir[k]); } if((sum < 536846336) & (sum > -536879104)) { sum = (sum + 0x2000) >> 14; } else if(sum > 536846336) { sum = 32767; } else { sum = -32768; } return((Word16)sum); /* saturation can occur here */ } /* * D_UTIL_up_samp * * Parameters: * res_d I: signal to upsampling * res_u O: upsampled output * L_frame I: length of output * * Function: * Upsampling * * Returns: * void */ static void D_UTIL_up_samp(Word16 *sig_d, Word16 *sig_u, Word16 L_frame) { Word32 pos, i, j; Word16 frac; pos = 0; /* position with 1/5 resolution */ for(j = 0; j < L_frame; j++) { i = (pos * INV_FAC5) >> 15; /* integer part = pos * 1/5 */ frac = (Word16)(pos - ((i << 2) + i)); /* frac = pos - (pos/5)*5 */ sig_u[j] = D_UTIL_interpol(&sig_d[i], D_ROM_fir_up, frac, FAC5, NB_COEF_UP); pos = pos + FAC4; /* position + 4/5 */ } return; } /* * D_UTIL_oversamp_16k * * Parameters: * sig12k8 I: signal to oversampling * lg I: length of input * sig16k O: oversampled signal * mem I/O: memory (2*12) * * Function: * Oversampling from 12.8kHz to 16kHz * * Returns: * void */ static void D_UTIL_oversamp_16k(Word16 sig12k8[], Word16 lg, Word16 sig16k[], Word16 mem[]) { Word16 lg_up; Word16 signal[L_SUBFR + (2 * NB_COEF_UP)]; memcpy(signal, mem, (2 * NB_COEF_UP) * sizeof(Word16)); memcpy(signal + (2 * NB_COEF_UP), sig12k8, lg * sizeof(Word16)); lg_up = (Word16)(((lg * UP_FAC) >> 15) << 1); D_UTIL_up_samp(signal + NB_COEF_UP, sig16k, lg_up); memcpy(mem, signal + lg, (2 * NB_COEF_UP) * sizeof(Word16)); return; } /* * D_UTIL_hp400_12k8 * * Parameters: * signal I/O: signal * lg I: lenght of signal * mem I/O: filter memory [6] * * Function: * 2nd order high pass filter with cut off frequency at 400 Hz. * * Algorithm: * * y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2] * + a[1]*y[i-1] + a[2]*y[i-2]; * * b[3] = {0.893554687, -1.787109375, 0.893554687}; * a[3] = {1.000000000, 1.787109375, -0.864257812}; * * * Returns: * void */ void D_UTIL_hp400_12k8(Word16 signal[], Word16 lg, Word16 mem[]) { Word32 i, L_tmp; Word16 y2_hi, y2_lo, y1_hi, y1_lo, x0, x1, x2; y2_hi = mem[0]; y2_lo = mem[1]; y1_hi = mem[2]; y1_lo = mem[3]; x0 = mem[4]; x1 = mem[5]; for(i = 0; i < lg; i++) { x2 = x1; x1 = x0; x0 = signal[i]; /* y[i] = b[0]*x[i] + b[1]*x[i-1] + b140[2]*x[i-2] */ /* + a[1]*y[i-1] + a[2] * y[i-2]; */ L_tmp = 8192L + (y1_lo * 29280); L_tmp = L_tmp + (y2_lo * (-14160)); L_tmp = (L_tmp >> 14); L_tmp = L_tmp + (y1_hi * 58560); L_tmp = L_tmp + (y2_hi * (-28320)); L_tmp = L_tmp + (x0 * 1830); L_tmp = L_tmp + (x1 * (-3660)); L_tmp = L_tmp + (x2 * 1830); L_tmp = (L_tmp << 1); /* coeff Q12 --> Q13 */ y2_hi = y1_hi; y2_lo = y1_lo; D_UTIL_l_extract(L_tmp, &y1_hi, &y1_lo); /* signal is divided by 16 to avoid overflow in energy computation */ signal[i] = (Word16)((L_tmp + 0x8000) >> 16); } mem[0] = y2_hi; mem[1] = y2_lo; mem[2] = y1_hi; mem[3] = y1_lo; mem[4] = x0; mem[5] = x1; return; } /* * D_UTIL_synthesis * * Parameters: * a I: LP filter coefficients * m I: order of LP filter * x I: input signal * y O: output signal * lg I: size of filtering * mem I/O: initial filter states * update_m I: update memory flag * * Function: * Perform the synthesis filtering 1/A(z). * * Returns: * void */ static void D_UTIL_synthesis(Word16 a[], Word16 m, Word16 x[], Word16 y[], Word16 lg, Word16 mem[], Word16 update) { Word32 i, j, tmp, s; Word16 y_buf[L_SUBFR16k + M16k], a0; Word16 *yy; yy = &y_buf[m]; /* See if a[0] is scaled */ s = D_UTIL_norm_s(a[0]) - 2; /* copy initial filter states into synthesis buffer */ memcpy(y_buf, mem, m * sizeof(Word16)); a0 = (Word16)(a[0] >> 1); /* input / 2 */ /* Do the filtering. */ for(i = 0; i < lg; i++) { tmp = x[i] * a0; for(j = 1; j <= m; j++) { tmp -= a[j] * yy[i - j]; } tmp <<= s; y[i] = yy[i] = (Word16)((tmp + 0x800) >> 12); } /* Update memory if required */ if(update) { memcpy(mem, &yy[lg - m], m * sizeof(Word16)); } return; } /* * D_UTIL_bp_6k_7k * * Parameters: * signal I/O: signal * lg I: lenght of signal * mem I/O: filter memory [4] * * Function: * 15th order band pass 6kHz to 7kHz FIR filter. * * Returns: * void */ void D_UTIL_bp_6k_7k(Word16 signal[], Word16 lg, Word16 mem[]) { Word32 x[L_SUBFR16k + (L_FIR - 1)]; Word32 i, j, tmp; for(i = 0; i < (L_FIR - 1); i++) { x[i] = (Word16)mem[i]; /* gain of filter = 4 */ } for(i = 0; i < lg; i++) { x[i + L_FIR - 1] = signal[i] >> 2; /* gain of filter = 4 */ } for(i = 0; i < lg; i++) { tmp = 0; for(j = 0; j < L_FIR; j++) { tmp += x[i + j] * D_ROM_fir_6k_7k[j]; } signal[i] = (Word16)((tmp + 0x4000) >> 15); } for(i = 0; i < (L_FIR - 1); i++) { mem[i] = (Word16)x[lg + i]; /* gain of filter = 4 */ } return; } /* * D_UTIL_hp_7k * * Parameters: * signal I/O: ISF vector * lg I: length of signal * mem I/O: memory (30) * * Function: * 15th order high pass 7kHz FIR filter * * Returns: * void */ static void D_UTIL_hp_7k(Word16 signal[], Word16 lg, Word16 mem[]) { Word32 i, j, tmp; Word16 x[L_SUBFR16k + (L_FIR - 1)]; memcpy(x, mem, (L_FIR - 1) * sizeof(Word16)); memcpy(&x[L_FIR - 1], signal, lg * sizeof(Word16)); for(i = 0; i < lg; i++) { tmp = 0; for(j = 0; j < L_FIR; j++) { tmp += x[i + j] * D_ROM_fir_7k[j]; } signal[i] = (Word16)((tmp + 0x4000) >> 15); } memcpy(mem, x + lg, (L_FIR - 1) * sizeof(Word16)); return; } /* * D_UTIL_Dec_synthesis * * Parameters: * Aq I: quantized Az * exc I: excitation at 12kHz * Q_new I: scaling performed on exc * synth16k O: 16kHz synthesis signal * prms I: parameters * HfIsf I/O: High frequency ISF:s * mode I: codec mode * newDTXState I: dtx state * bfi I: bad frame indicator * st I/O: State structure * * Function: * Synthesis of signal at 16kHz with HF extension. * * Returns: * void */ void D_UTIL_dec_synthesis(Word16 Aq[], Word16 exc[], Word16 Q_new, Word16 synth16k[], Word16 prms, Word16 HfIsf[], Word16 mode, Word16 newDTXState, Word16 bfi, Decoder_State *st) { Word32 tmp, i; Word16 exp; Word16 ener, exp_ener; Word32 fac; Word16 synth_hi[M + L_SUBFR], synth_lo[M + L_SUBFR]; Word16 synth[L_SUBFR]; Word16 HF[L_SUBFR16k]; /* High Frequency vector */ Word16 Ap[M16k + 1]; Word16 HfA[M16k + 1]; Word16 HF_corr_gain; Word16 HF_gain_ind; Word32 gain1, gain2; Word16 weight1, weight2; /* * Speech synthesis * * - Find synthesis speech corresponding to exc2[]. * - Perform fixed deemphasis and hp 50hz filtering. * - Oversampling from 12.8kHz to 16kHz. */ memcpy(synth_hi, st->mem_syn_hi, M * sizeof(Word16)); memcpy(synth_lo, st->mem_syn_lo, M * sizeof(Word16)); D_UTIL_synthesis_32(Aq, M, exc, Q_new, synth_hi + M, synth_lo + M, L_SUBFR); memcpy(st->mem_syn_hi, synth_hi + L_SUBFR, M * sizeof(Word16)); memcpy(st->mem_syn_lo, synth_lo + L_SUBFR, M * sizeof(Word16)); D_UTIL_deemph_32(synth_hi + M, synth_lo + M, synth, PREEMPH_FAC, L_SUBFR, &(st->mem_deemph)); D_UTIL_hp50_12k8(synth, L_SUBFR, st->mem_sig_out); D_UTIL_oversamp_16k(synth, L_SUBFR, synth16k, st->mem_oversamp); /* * HF noise synthesis * * - Generate HF noise between 5.5 and 7.5 kHz. * - Set energy of noise according to synthesis tilt. * tilt > 0.8 ==> - 14 dB (voiced) * tilt 0.5 ==> - 6 dB (voiced or noise) * tilt < 0.0 ==> 0 dB (noise) */ /* generate white noise vector */ for(i = 0; i < L_SUBFR16k; i++) { HF[i] = (Word16)(D_UTIL_random(&(st->mem_seed2)) >> 3); } /* energy of excitation */ D_UTIL_signal_down_scale(exc, L_SUBFR, 3); Q_new = (Word16)(Q_new - 3); ener = (Word16)(D_UTIL_dot_product12(exc, exc, L_SUBFR, &exp_ener) >> 16); exp_ener = (Word16)(exp_ener - (Q_new << 1)); /* set energy of white noise to energy of excitation */ tmp = (Word16)(D_UTIL_dot_product12(HF, HF, L_SUBFR16k, &exp) >> 16); if(tmp > ener) { tmp = tmp >> 1; /* Be sure tmp < ener */ exp = (Word16)(exp + 1); } tmp = (tmp << 15) / ener; if(tmp > 32767) { tmp = 32767; } tmp = tmp << 16; /* result is normalized */ exp = (Word16)(exp - exp_ener); D_UTIL_normalised_inverse_sqrt(&tmp, &exp); /* L_tmp x 2, L_tmp in Q31 */ /* tmp = 2 x sqrt(ener_exc/ener_hf) */ if(exp >= 0) { tmp = tmp >> (15 - exp); } else { tmp = tmp >> (-exp); tmp = tmp >> 15; } /* saturation */ if(tmp > 0x7FFF) { tmp = 0x7FFF; } for(i = 0; i < L_SUBFR16k; i++) { HF[i] = (Word16)((HF[i] * tmp) >> 15); } /* find tilt of synthesis speech (tilt: 1=voiced, -1=unvoiced) */ D_UTIL_hp400_12k8(synth, L_SUBFR, st->mem_hp400); tmp = 0L; for(i = 0; i < L_SUBFR; i++) { tmp = tmp + (synth[i] * synth[i]); } tmp = (tmp << 1) + 1; exp = D_UTIL_norm_l(tmp); ener = (Word16)((tmp << exp) >> 16); /* ener = r[0] */ tmp = 0L; for(i = 1; i < L_SUBFR; i++) { tmp = tmp + (synth[i] * synth[i - 1]); } tmp = (tmp << 1) + 1; tmp = (tmp << exp) >> 16; /* tmp = r[1] */ if(tmp > 0) { fac = ((tmp << 15) / ener); if(fac > 32767) { fac = 32767; } } else { fac = 0; } /* modify energy of white noise according to synthesis tilt */ gain1 = (32767 - fac); gain2 = ((32767 - fac) * 20480) >> 15; gain2 = (gain2 << 1); if(gain2 > 32767) gain2 = 32767; if(st->mem_vad_hist > 0) { weight1 = 0; weight2 = 32767; } else { weight1 = 32767; weight2 = 0; } tmp = (weight1 * gain1) >> 15; tmp = tmp + ((weight2 * gain2) >> 15); if(tmp != 0) { tmp = tmp + 1; } if(tmp < 3277) { tmp = 3277; /* 0.1 in Q15 */ } if((mode == MODE_24k) & (bfi == 0)) { /* HF correction gain */ HF_gain_ind = prms; HF_corr_gain = D_ROM_hp_gain[HF_gain_ind]; /* HF gain */ for(i = 0; i < L_SUBFR16k; i++) { HF[i] = (Word16)(((HF[i] * HF_corr_gain) >> 15) << 1); } } else { for(i = 0; i < L_SUBFR16k; i++) { HF[i] = (Word16)((HF[i] * tmp) >> 15); } } if((mode <= MODE_7k) & (newDTXState == SPEECH)) { D_LPC_isf_extrapolation(HfIsf); D_LPC_isp_a_conversion(HfIsf, HfA, 0, M16k); D_LPC_a_weight(HfA, Ap, 29491, M16k); /* fac=0.9 */ D_UTIL_synthesis(Ap, M16k, HF, HF, L_SUBFR16k, st->mem_syn_hf, 1); } else { /* synthesis of noise: 4.8kHz..5.6kHz --> 6kHz..7kHz */ D_LPC_a_weight(Aq, Ap, 19661, M); /* fac=0.6 */ D_UTIL_synthesis(Ap, M, HF, HF, L_SUBFR16k, st->mem_syn_hf + (M16k - M), 1); } /* noise High Pass filtering (1ms of delay) */ D_UTIL_bp_6k_7k(HF, L_SUBFR16k, st->mem_hf); if(mode == MODE_24k) { /* Low Pass filtering (7 kHz) */ D_UTIL_hp_7k(HF, L_SUBFR16k, st->mem_hf3); } /* add filtered HF noise to speech synthesis */ for(i = 0; i < L_SUBFR16k; i++) { tmp = (synth16k[i] + HF[i]); synth16k[i] = D_UTIL_saturate(tmp); } return; } /* * D_UTIL_preemph * * Parameters: * x I/O: signal * mu I: preemphasis factor * lg I: vector size * mem I/O: memory (x[-1]) * * Function: * Filtering through 1 - mu z^-1 * * * Returns: * void */ void D_UTIL_preemph(Word16 x[], Word16 mu, Word16 lg, Word16 *mem) { Word32 i, L_tmp; Word16 temp; temp = x[lg - 1]; for(i = lg - 1; i > 0; i--) { L_tmp = x[i] << 15; L_tmp = L_tmp - (x[i - 1] * mu); x[i] = (Word16)((L_tmp + 0x4000) >> 15); } L_tmp = x[0] << 15; L_tmp = L_tmp - (*mem * mu); x[0] = (Word16)((L_tmp + 0x4000) >> 15); *mem = temp; return; }