out[n] -= filter_coeffs[i-1] * out[n-i];
}
}
+
+void ff_celp_lp_zero_synthesis_filterf(
+ float *out,
+ const float* filter_coeffs,
+ const float* in,
+ int buffer_length,
+ int filter_length)
+{
+ int i,n;
+
+ // Avoids a +1 in the inner loop.
+ filter_length++;
+
+ for(n=0; n<buffer_length; n++)
+ {
+ out[n] = in[n];
+ for(i=1; i<filter_length; i++)
+ out[n] -= filter_coeffs[i-1] * in[n-i];
+ }
+}
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 */