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authorVitor Sessak <vitor1001@gmail.com>2009-10-27 23:53:18 +0000
committerVitor Sessak <vitor1001@gmail.com>2009-10-27 23:53:18 +0000
commit504eee37debbf7ce6ec3b79ae8825727258c3fd7 (patch)
tree094306ce1889f1e9ddf817f67e6d86c681ed6aa4 /libavcodec/acelp_filters.h
parent2be414c8dee911744268341b5bd94b9e6a96f67c (diff)
Commit some functions that are used by both SIPR and AMR.
Based on AMR SoC code by Robert Swain and Colin McQuillan. Originally committed as revision 20392 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/acelp_filters.h')
-rw-r--r--libavcodec/acelp_filters.h19
1 files changed, 19 insertions, 0 deletions
diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h
index 2cbe9bb17d..c5be5a6cb1 100644
--- a/libavcodec/acelp_filters.h
+++ b/libavcodec/acelp_filters.h
@@ -56,6 +56,14 @@ void ff_acelp_interpolate(int16_t* out, const int16_t* in,
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)
@@ -97,4 +105,15 @@ void ff_acelp_apply_order_2_transfer_function(float *samples,
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 */