*
* Copyright (c) 2008 Vladimir Voroshilov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
*
* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
*
- * \note fc_in and fc_out should not overlap!
+ * @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.
- * @param out [out] pointer to output buffer
+ * @param[out] out pointer to output buffer
* @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
* @param in input signal
* @param buffer_length amount of data to process
*
* @return 1 if overflow occurred, 0 - otherwise
*
- * @note Output buffer must contain 10 samples of past
+ * @note Output buffer must contain filter_length samples of past
* speech data before pointer.
*
* 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 out [out] pointer to output buffer
+ * @param[out] out pointer to output buffer
* - the array out[-filter_length, -1] must
* contain the previous result of this 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 10 samples of past
+ * @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 */