X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Facelp_filters.h;h=2be4c24395b90408535d34b760c5ff7e415f7d23;hb=83678dbbae64ad8c501e0c732c1117e642c25dae;hp=e1e3d685ae9d294542482fdd05c171cae5b15b53;hpb=4599d22c0cd7fa952bc02375d2899d8c9a31b9ae;p=ffmpeg

diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index e1e3d685ae9..2be4c24395b 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -3,20 +3,20 @@
  *
  * 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
  */
 
@@ -32,13 +32,13 @@
  * 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.
+ * could not be determined from the original comments with certainty.
  */
 extern const int16_t ff_acelp_interp_filter[61];
 
 /**
  * Generic FIR interpolation routine.
- * @param out [out] buffer for interpolated data
+ * @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
@@ -49,22 +49,23 @@ extern const int16_t ff_acelp_interp_filter[61];
  * 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);
+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);
+
+/**
+ * Floating point version of ff_acelp_interpolate()
+ */
+void ff_acelp_interpolatef(float *out, const float *in,
+                           const float *filter_coeffs, int precision,
+                           int frac_pos, int filter_length, int length);
 
 
 /**
  * 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)
+ * @param[out]     out   output buffer for filtered speech data
+ * @param[in,out]  hpf_f past filtered data from previous (2 items long)
  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
  * @param in speech data to process
  * @param length input data size
@@ -84,10 +85,35 @@ void ff_acelp_interpolate(
  *         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 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);
+
 
 #endif /* AVCODEC_ACELP_FILTERS_H */