int n;
int x[10];
float y[10];
+ int no_repeat_mask;
int pitch_lag;
float pitch_fac;
} AMRFixed;
*/
extern const uint8_t ff_fc_2pulses_9bits_track2_gray[32];
+/**
+ * b60 hamming windowed sinc function coefficients
+ */
+extern const float ff_b60_sinc[61];
+
+/**
+ * Table of pow(0.7,n)
+ */
+extern const float ff_pow_0_7[10];
+
+/**
+ * Table of pow(0.75,n)
+ */
+extern const float ff_pow_0_75[10];
+
+/**
+ * Table of pow(0.55,n)
+ */
+extern const float ff_pow_0_55[10];
+
/**
* Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
* @param fc_v [out] decoded fixed codebook vector (2.13)
*
* Used in G.729 @8k, G.729 @4.4k, G.729 @6.4k, AMR @7.95k, AMR @7.40k
*/
-void ff_acelp_fc_pulse_per_track(
- int16_t* fc_v,
- const uint8_t *tab1,
- const uint8_t *tab2,
- int pulse_indexes,
- int pulse_signs,
- int pulse_count,
- int bits);
+void ff_acelp_fc_pulse_per_track(int16_t* fc_v,
+ const uint8_t *tab1,
+ const uint8_t *tab2,
+ int pulse_indexes,
+ int pulse_signs,
+ int pulse_count,
+ int bits);
/**
* Decode the algebraic codebook index to pulse positions and signs and
*
* out[i] = (in_a[i]*weight_a + in_b[i]*weight_b + rounder) >> shift
*/
-void ff_acelp_weighted_vector_sum(
- int16_t* out,
- const int16_t *in_a,
- const int16_t *in_b,
- int16_t weight_coeff_a,
- int16_t weight_coeff_b,
- int16_t rounder,
- int shift,
- int length);
+void ff_acelp_weighted_vector_sum(int16_t* out,
+ const int16_t *in_a,
+ const int16_t *in_b,
+ int16_t weight_coeff_a,
+ int16_t weight_coeff_b,
+ int16_t rounder,
+ int shift,
+ int length);
/**
* float implementation of weighted sum of two vectors.
* @note It is safe to pass the same buffer for out and in_a or in_b.
*/
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
- float weight_coeff_a, float weight_coeff_b, int length);
+ float weight_coeff_a, float weight_coeff_b,
+ int length);
/**
- * Adaptative gain control (as used in AMR postfiltering)
+ * Adaptive gain control (as used in AMR postfiltering)
*
* @param buf_out the input speech buffer
* @param speech_energ input energy
* @param alpha exponential filter factor
* @param gain_mem a pointer to the filter memory (single float of size)
*/
-void ff_adaptative_gain_control(float *buf_out, float speech_energ,
- int size, float alpha, float *gain_mem);
+void ff_adaptive_gain_control(float *buf_out, float speech_energ,
+ int size, float alpha, float *gain_mem);
/**
* Set the sum of squares of a signal by scaling