Fix misspelled parameter names in Doxygen documentation.

[ffmpeg] / libavcodec / resample2.c

diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
index 0cbe9b7a665aa32f82831cd3c646a60c4869ffd1..45f41a144c2a9a2e20d63bb4641292f493f63085 100644 (file)
--- a/libavcodec/resample2.c
+++ b/libavcodec/resample2.c
@@ -17,17 +17,15 @@
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
  */
 
 /**
- * @file resample2.c
+ * @file
  * audio resampling
  * @author Michael Niedermayer <michaelni@gmx.at>
  */
 
 #include "avcodec.h"
-#include "common.h"
 #include "dsputil.h"
 
 #ifndef CONFIG_RESAMPLE_HP
@@ -40,7 +38,7 @@
 #define FELEM_MIN INT16_MIN
 #define WINDOW_TYPE 9
 #elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
-#define FILTER_SHIFT 22
+#define FILTER_SHIFT 30
 
 #define FELEM int32_t
 #define FELEM2 int64_t
@@ -51,14 +49,15 @@
 #else
 #define FILTER_SHIFT 0
 
-#define FELEM long double
-#define FELEM2 long double
-#define FELEML long double
+#define FELEM double
+#define FELEM2 double
+#define FELEML double
 #define WINDOW_TYPE 24
 #endif
 
 
 typedef struct AVResampleContext{
+    const AVClass *av_class;
     FELEM *filter_bank;
     int filter_length;
     int ideal_dst_incr;
@@ -77,11 +76,13 @@ typedef struct AVResampleContext{
  */
 static double bessel(double x){
     double v=1;
+    double lastv=0;
     double t=1;
     int i;
 
     x= x*x/4;
-    for(i=1; i<50; i++){
+    for(i=1; v != lastv; i++){
+        lastv=v;
         t *= x/(i*i);
         v += t;
     }
@@ -94,8 +95,8 @@ static double bessel(double x){
  * @param scale wanted sum of coefficients for each filter
  * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
  */
-void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
-    int ph, i, v;
+static void build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
+    int ph, i;
     double x, y, w, tab[tap_count];
     const int center= (tap_count-1)/2;
 
@@ -177,10 +178,6 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
 #endif
 }
 
-/**
- * initalizes a audio resampler.
- * note, if either rate is not a integer then simply scale both rates up so they are
- */
 AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
     AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
     double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
@@ -192,7 +189,7 @@ AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size,
 
     c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
     c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
-    av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
+    build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
     memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
     c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
 
@@ -208,33 +205,12 @@ void av_resample_close(AVResampleContext *c){
     av_freep(&c);
 }
 
-/**
- * Compensates samplerate/timestamp drift. The compensation is done by changing
- * the resampler parameters, so no audible clicks or similar distortions ocur
- * @param compensation_distance distance in output samples over which the compensation should be performed
- * @param sample_delta number of output samples which should be output less
- *
- * example: av_resample_compensate(c, 10, 500)
- * here instead of 510 samples only 500 samples would be output
- *
- * note, due to rounding the actual compensation might be slightly different,
- * especially if the compensation_distance is large and the in_rate used during init is small
- */
 void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
 //    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
     c->compensation_distance= compensation_distance;
     c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
 }
 
-/**
- * resamples.
- * @param src an array of unconsumed samples
- * @param consumed the number of samples of src which have been consumed are returned here
- * @param src_size the number of unconsumed samples available
- * @param dst_size the amount of space in samples available in dst
- * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
- * @return the number of samples written in dst or -1 if an error occured
- */
 int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
     int dst_index, i;
     int index= c->index;
@@ -281,7 +257,7 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
         }
 
 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
-        dst[dst_index] = av_clip(lrintf(val), -32768, 32767);
+        dst[dst_index] = av_clip_int16(lrintf(val));
 #else
         val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
         dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;