const int16_t* filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
+/**
+ * Floating point version of ff_acelp_interpolate()
+ */
+void ff_acelp_interpolatef(float *out, const float *in,
+ const float *filter_coeffs, int precision,
+ int frac_pos, int filter_length, int length);
+
+
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
* @param out [out] output buffer for filtered speech data
/**
* Apply an order 2 rational transfer function in-place.
*
- * @param samples [in/out]
+ * @param out output buffer for filtered speech samples
+ * @param in input buffer containing speech data (may be the same as out)
* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
* @param gain scale factor for final output
* @param mem intermediate values used by filter (should be 0 initially)
* @param n number of samples
*/
-void ff_acelp_apply_order_2_transfer_function(float *samples,
+void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain,
float mem[2], int n);
+/**
+ * Apply tilt compensation filter, 1 - tilt * z-1.
+ *
+ * @param mem pointer to the filter's state (one single float)
+ * @param tilt tilt factor
+ * @param samples array where the filter is applied
+ * @param size the size of the samples array
+ */
+void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
+
+
#endif /* AVCODEC_ACELP_FILTERS_H */