X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Facelp_filters.h;h=c5be5a6cb1a5869ba75e08f9549927a1e9684ffb;hb=86decad65240ea9e1bea177f1091f80d08b62402;hp=324be3b63d8b75dfd09a23d455e0f1d0c35617b3;hpb=5038ecb1f17099ad9dc1d1e79516d9f6550e1ebd;p=ffmpeg

diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index 324be3b63d8..c5be5a6cb1a 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -20,8 +20,8 @@
  * 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>
 
@@ -51,74 +51,17 @@ extern const int16_t ff_acelp_interp_filter[61];
  * 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);
+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);
 
 /**
- * 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
- *
- *  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!
+ * Floating point version of ff_acelp_interpolate()
  */
-void ff_acelp_convolve_circ(
-        int16_t* fc_out,
-        const int16_t* fc_in,
-        const int16_t* filter,
-        int len);
+void ff_acelp_interpolatef(float *out, const float *in,
+                           const float *filter_coeffs, int precision,
+                           int frac_pos, int filter_length, int length);
 
-/**
- * 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
- *
- * @note Output buffer must contain 10 samples of past
- *       speech data before pointer.
- *
- * Routine applies 1/A(z) filter to given speech data.
- */
-int ff_acelp_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);
-
-/**
- * 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);
 
 /**
  * high-pass filtering and upscaling (4.2.5 of G.729).
@@ -131,7 +74,7 @@ void ff_acelp_weighted_filter(
  * 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
  *       tail of the previous subframe.
@@ -143,10 +86,34 @@ void ff_acelp_weighted_filter(
  *         fixed-point all coefficients are the same as in G.729. Thus this
  *         routine can be used for the fixed-point AMR decoder, too.
  */
-void ff_acelp_high_pass_filter(
-        int16_t* out,
-        int hpf_f[2],
-        const int16_t* in,
-        int length);
+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 samples [in/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,
+                                              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 /* FFMPEG_ACELP_FILTERS_H */
+#endif /* AVCODEC_ACELP_FILTERS_H */