*
* \note fc_in and fc_out should not overlap!
*/
-void ff_celp_convolve_circ(
- int16_t* fc_out,
- const int16_t* fc_in,
- const int16_t* filter,
- int len);
+void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
+ const int16_t *filter, int len);
+
+/**
+ * Add an array to a rotated array.
+ *
+ * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
+ *
+ * @param out result vector
+ * @param in samples to be added unfiltered
+ * @param lagged samples to be rotated, multiplied and added
+ * @param lag lagged vector delay in the range [0, n]
+ * @param fac scalefactor for lagged samples
+ * @param n number of samples
+ */
+void ff_celp_circ_addf(float *out, const float *in,
+ const float *lagged, int lag, float fac, int n);
/**
* LP synthesis filter.
*
* Routine applies 1/A(z) filter to given speech data.
*/
-int ff_celp_lp_synthesis_filter(
- int16_t *out,
- const int16_t* filter_coeffs,
- const int16_t* in,
- int buffer_length,
- int filter_length,
- int stop_on_overflow,
- int rounder);
+int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
+ const int16_t *in, int buffer_length,
+ int filter_length, int stop_on_overflow,
+ int rounder);
/**
* LP synthesis filter.
* @param filter_coeffs filter coefficients.
* @param in input signal
* @param buffer_length amount of data to process
- * @param filter_length filter length (10 for 10th order LP filter)
+ * @param filter_length filter length (10 for 10th order LP filter). Must be
+ * greater than 4 and even.
*
* @note Output buffer must contain filter_length samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
-void ff_celp_lp_synthesis_filterf(
- float *out,
- const float* filter_coeffs,
- const float* in,
- int buffer_length,
- int filter_length);
+void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
+ const float *in, int buffer_length,
+ int filter_length);
+
+/**
+ * LP zero synthesis filter.
+ * @param out [out] pointer to output buffer
+ * @param filter_coeffs filter coefficients.
+ * @param in input signal
+ * - the array in[-filter_length, -1] must
+ * contain the previous input of this filter
+ * @param buffer_length amount of data to process
+ * @param filter_length filter length (10 for 10th order LP filter)
+ *
+ * @note Output buffer must contain filter_length samples of past
+ * speech data before pointer.
+ *
+ * Routine applies A(z) filter to given speech data.
+ */
+void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
+ const float *in, int buffer_length,
+ int filter_length);
#endif /* AVCODEC_CELP_FILTERS_H */