frsh: Export information about the last RTP contract and VRES

[frescor/ffmpeg.git] / libavcodec / acelp_filters.h

diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index 324be3b63d8b75dfd09a23d455e0f1d0c35617b3..e1e3d685ae9d294542482fdd05c171cae5b15b53 100644 (file)
--- 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>
 
@@ -60,65 +60,6 @@ void ff_acelp_interpolate(
         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!
- */
-void ff_acelp_convolve_circ(
-        int16_t* fc_out,
-        const int16_t* fc_in,
-        const int16_t* filter,
-        int len);
-
-/**
- * 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 +72,7 @@ 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.
@@ -149,4 +90,4 @@ void ff_acelp_high_pass_filter(
         const int16_t* in,
         int length);
 
-#endif /* FFMPEG_ACELP_FILTERS_H */
+#endif /* AVCODEC_ACELP_FILTERS_H */