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[ffmpeg] / libavcodec / celp_filters.h

diff --git a/libavcodec/celp_filters.h b/libavcodec/celp_filters.h
index d9db95d45426d2cc46bbe89de13c2d690f1045f9..145e3d33465b19fce065a64c6787b5daa6b84279 100644 (file)
--- a/libavcodec/celp_filters.h
+++ b/libavcodec/celp_filters.h
@@ -36,10 +36,8 @@
  *
  * \note fc_in and fc_out should not overlap!
  */
-void ff_celp_convolve_circ(int16_t* fc_out,
-                           const int16_t* fc_in,
-                           const int16_t* filter,
-                           int len);
+void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
+                           const int16_t *filter, int len);
 
 /**
  * Add an array to a rotated array.
@@ -58,7 +56,7 @@ void ff_celp_circ_addf(float *out, const float *in,
 
 /**
  * LP synthesis filter.
- * @param out [out] pointer to output buffer
+ * @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
@@ -74,38 +72,34 @@ void ff_celp_circ_addf(float *out, const float *in,
  *
  * Routine applies 1/A(z) filter to given speech data.
  */
-int ff_celp_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 ff_celp_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);
 
 /**
  * LP synthesis filter.
- * @param out [out] pointer to output buffer
+ * @param[out] out pointer to output buffer
  *        - the array out[-filter_length, -1] must
  *        contain the previous result of this filter
  * @param filter_coeffs filter coefficients.
  * @param in input signal
  * @param buffer_length amount of data to process
- * @param filter_length filter length (10 for 10th order LP filter)
+ * @param filter_length filter length (10 for 10th order LP filter). Must be
+ *                      greater than 4 and even.
  *
  * @note Output buffer must contain filter_length samples of past
  *       speech data before pointer.
  *
  * Routine applies 1/A(z) filter to given speech data.
  */
-void ff_celp_lp_synthesis_filterf(float *out,
-                                  const float* filter_coeffs,
-                                  const float* in,
-                                  int buffer_length,
+void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
+                                  const float *in, int buffer_length,
                                   int filter_length);
 
 /**
  * LP zero synthesis filter.
- * @param out [out] pointer to output buffer
+ * @param[out] out pointer to output buffer
  * @param filter_coeffs filter coefficients.
  * @param in input signal
  *        - the array in[-filter_length, -1] must
@@ -118,10 +112,8 @@ void ff_celp_lp_synthesis_filterf(float *out,
  *
  * Routine applies A(z) filter to given speech data.
  */
-void ff_celp_lp_zero_synthesis_filterf(float *out,
-                                       const float* filter_coeffs,
-                                       const float* in,
-                                       int buffer_length,
+void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
+                                       const float *in, int buffer_length,
                                        int filter_length);
 
 #endif /* AVCODEC_CELP_FILTERS_H */