* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef FFMPEG_ACELP_FILTERS_H
-#define FFMPEG_ACELP_FILTERS_H
+#ifndef AVCODEC_ACELP_FILTERS_H
+#define AVCODEC_ACELP_FILTERS_H
+
+#include <stdint.h>
+
+/**
+ * low-pass Finite Impulse Response filter coefficients.
+ *
+ * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
+ * the coefficients are scaled by 2^15.
+ * This array only contains the right half of the filter.
+ * This filter is likely identical to the one used in G.729, though this
+ * could not be determined from the original comments with certainity.
+ */
+extern const int16_t ff_acelp_interp_filter[61];
+
+/**
+ * Generic FIR interpolation routine.
+ * @param out [out] buffer for interpolated data
+ * @param in input data
+ * @param filter_coeffs interpolation filter coefficients (0.15)
+ * @param precision sub sample factor, that is the precision of the position
+ * @param frac_pos fractional part of position [0..precision-1]
+ * @param filter_length filter length
+ * @param length length of output
+ *
+ * filter_coeffs contains coefficients of the right half of the symmetric
+ * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
+ * See ff_acelp_interp_filter for an example.
+ *
+ */
+void ff_acelp_interpolate(
+ int16_t* out,
+ const int16_t* in,
+ const int16_t* filter_coeffs,
+ int precision,
+ int frac_pos,
+ int filter_length,
+ int length);
/**
- * \brief Circularly convolve fixed vector with a phase dispersion impulse
+ * Circularly convolve fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
- * \param fc_out vector with filter applied
- * \param fc_in source vector
- * \param filter phase filter coefficients
+ * @param fc_out vector with filter applied
+ * @param fc_in source vector
+ * @param filter phase filter coefficients
*
* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
*
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
- int subframe_size);
+ int len);
/**
- * \brief LP synthesis filter
- * \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
- * \param filter_length filter length (11 for 10th order LP filter)
- * \param stop_on_overflow 1 - return immediately if overflow occurs
+ * LP synthesis filter.
+ * @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
+ * @param filter_length filter length (10 for 10th order LP filter)
+ * @param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
+ * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
*
- * \return 1 if overflow occurred, 0 - otherwise
+ * @return 1 if overflow occurred, 0 - otherwise
*
- * \note Output buffer must contain 10 samples of past
+ * @note Output buffer must contain 10 samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
const int16_t* in,
int buffer_length,
int filter_length,
- int stop_on_overflow);
+ int stop_on_overflow,
+ int rounder);
-/**
- * \brief Calculates coefficients of weighted A(z/weight) filter.
- * \param out [out] weighted A(z/weight) result
- * filter (-0x8000 <= (3.12) < 0x8000)
- * \param in source filter (-0x8000 <= (3.12) < 0x8000)
- * \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
- * \param filter_length filter length (11 for 10th order LP filter)
- *
- * out[i]=weight_pow[i]*in[i] , i=0..9
- */
-void ff_acelp_weighted_filter(
- int16_t *out,
- const int16_t* in,
- const int16_t *weight_pow,
- int filter_length);
/**
- * \brief high-pass filtering and upscaling (4.2.5 of G.729)
- * \param out [out] output buffer for filtered speech data
- * \param hpf_f [in/out] past filtered data from previous (2 items long)
+ * high-pass filtering and upscaling (4.2.5 of G.729).
+ * @param out [out] output buffer for filtered speech data
+ * @param hpf_f [in/out] past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
- * \param in speech data to process
- * \param length input data size
+ * @param in speech data to process
+ * @param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
- * The filter has a cut-off frequency of 100Hz
+ * The filter has a cut-off frequency of 1/80 of the sampling freq
*
- * \note Two items before the top of the out buffer must contain two items from the
+ * @note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
*
- * \remark It is safe to pass the same array in in and out parameters
+ * @remark It is safe to pass the same array in in and out parameters.
*
- * \remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
+ * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
const int16_t* in,
int length);
-#endif // FFMPEG_ACELP_FILTERS_H
+#endif /* AVCODEC_ACELP_FILTERS_H */
projects / ffmpeg / blobdiff