+void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
+ const int16_t* in, int length);
+
+/**
+ * Apply an order 2 rational transfer function in-place.
+ *
+ * @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 *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);
+