Simplify interleaving code.

[ffmpeg] / libavcodec / acelp_vectors.h

diff --git a/libavcodec/acelp_vectors.h b/libavcodec/acelp_vectors.h
index 2cc5f3664925204219a838977ba50d34e3683296..3e6652f377a9b2d93b91142c72e67f0473696ed3 100644 (file)
--- a/libavcodec/acelp_vectors.h
+++ b/libavcodec/acelp_vectors.h
@@ -212,7 +212,7 @@ void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
                              int length);
 
 /**
- * Adaptative gain control (as used in AMR postfiltering)
+ * Adaptive gain control (as used in AMR postfiltering)
  *
  * @param buf_out the input speech buffer
  * @param speech_energ input energy
@@ -220,8 +220,8 @@ void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
  * @param alpha exponential filter factor
  * @param gain_mem a pointer to the filter memory (single float of size)
  */
-void ff_adaptative_gain_control(float *buf_out, float speech_energ,
-                                int size, float alpha, float *gain_mem);
+void ff_adaptive_gain_control(float *buf_out, float speech_energ,
+                              int size, float alpha, float *gain_mem);
 
 /**
  * Set the sum of squares of a signal by scaling