Windows Televison (WTV) demuxer

[ffmpeg] / libavcodec / acelp_filters.h

diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index e1e3d685ae9d294542482fdd05c171cae5b15b53..0b1ccf4e71095092543ab169fe41521098897743 100644 (file)
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -38,7 +38,7 @@ 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
@@ -51,20 +51,22 @@ 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);
+
+/**
+ * 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 +86,35 @@ 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 */