[patch] experimental support for resampling of 24-bit sound by lavcresample

mplayer-float-resample.diff (text/plain), 6.40 KB, created by wbrana on 2008-12-09 20:20:07 UTC

diff --git a/libaf/af_lavcresample.c b/libaf/af_lavcresample.c
index 2dbdf78..c06104b 100644
--- a/libaf/af_lavcresample.c
+++ b/libaf/af_lavcresample.c
@@ -28,10 +28,12 @@
 #include "libavcodec/avcodec.h"
 #include "libavutil/rational.h"
 
+#define FORMAT float
+
 // Data for specific instances of this filter
 typedef struct af_resample_s{
     struct AVResampleContext *avrctx;
-    int16_t *in[AF_NCH];
+    FORMAT *in[AF_NCH];
     int in_alloc;
     int index;
     
@@ -56,8 +58,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
 
     af->data->nch    = data->nch;
     if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
+    af->data->format = AF_FORMAT_FLOAT_NE;
+    af->data->bps    = sizeof(FORMAT);
     af->mul = (double)af->data->rate / data->rate;
     af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
 
@@ -104,29 +106,29 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data)
 {    
   af_resample_t *s = af->setup;
   int i, j, consumed, ret;
-  int16_t *in = (int16_t*)data->audio;
-  int16_t *out;
+  FORMAT *in = (FORMAT*)data->audio;
+  FORMAT *out;
   int chans   = data->nch;
-  int in_len  = data->len/(2*chans);
+  int in_len  = data->len/(sizeof(FORMAT)*chans);
   int out_len = in_len * af->mul + 10;
-  int16_t tmp[AF_NCH][out_len];
+  FORMAT tmp[AF_NCH][out_len];
     
   if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
       return NULL;
   
-  out= (int16_t*)af->data->audio;
+  out= (FORMAT*)af->data->audio;
   
-  out_len= min(out_len, af->data->len/(2*chans));
+  out_len= min(out_len, af->data->len/(sizeof(FORMAT)*chans));
   
   if(s->in_alloc < in_len + s->index){
       s->in_alloc= in_len + s->index;
       for(i=0; i<chans; i++){
-          s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
+          s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(FORMAT));
       }
   }
 
   if(chans==1){
-      memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
+      memcpy(&s->in[0][s->index], in, in_len * sizeof(FORMAT));
   }else if(chans==2){
       for(j=0; j<in_len; j++){
           s->in[0][j + s->index]= *(in++);
@@ -148,11 +150,11 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data)
   
   s->index= in_len - consumed;
   for(i=0; i<chans; i++){
-      memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
+      memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(FORMAT));
   }
 
   if(chans==1){
-      memcpy(out, tmp[0], out_len*sizeof(int16_t));
+      memcpy(out, tmp[0], out_len*sizeof(FORMAT));
   }else if(chans==2){
       for(j=0; j<out_len; j++){
           *(out++)= tmp[0][j];
@@ -167,7 +169,7 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data)
   }
 
   data->audio = af->data->audio;
-  data->len   = out_len*chans*2;
+  data->len   = out_len*chans*sizeof(FORMAT);
   data->rate  = af->data->rate;
   return data;
 }
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 53ded30..13bdc0e 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -2407,7 +2407,7 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
 void audio_resample_close(ReSampleContext *s);
 
 struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
-int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
+int av_resample(struct AVResampleContext *c, float *dst, float *src, int *consumed, int src_size, int dst_size, int update_ctx);
 void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
 void av_resample_close(struct AVResampleContext *c);
 
diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
index 4397d2a..710eb75 100644
--- a/libavcodec/resample2.c
+++ b/libavcodec/resample2.c
@@ -28,33 +28,14 @@
 #include "avcodec.h"
 #include "dsputil.h"
 
-#ifndef CONFIG_RESAMPLE_HP
-#define FILTER_SHIFT 15
-
-#define FELEM int16_t
-#define FELEM2 int32_t
-#define FELEML int64_t
-#define FELEM_MAX INT16_MAX
-#define FELEM_MIN INT16_MIN
-#define WINDOW_TYPE 9
-#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
-#define FILTER_SHIFT 30
-
-#define FELEM int32_t
-#define FELEM2 int64_t
-#define FELEML int64_t
-#define FELEM_MAX INT32_MAX
-#define FELEM_MIN INT32_MIN
-#define WINDOW_TYPE 12
-#else
 #define FILTER_SHIFT 0
 
 #define FELEM double
 #define FELEM2 double
 #define FELEML double
 #define WINDOW_TYPE 24
-#endif
 
+#define FORMAT float
 
 typedef struct AVResampleContext{
     FELEM *filter_bank;
@@ -130,11 +111,13 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
 
         /* normalize so that an uniform color remains the same */
         for(i=0;i<tap_count;i++) {
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
+//#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
             filter[ph * tap_count + i] = tab[i] / norm;
+/*
 #else
             filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
 #endif
+*/
         }
     }
 #if 0
@@ -233,7 +216,7 @@ void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensa
  * @param update_ctx If this is 0 then the context will not 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 occurred
  */
-int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
+int av_resample(AVResampleContext *c, float *dst, float *src, int *consumed, int src_size, int dst_size, int update_ctx){
     int dst_index, i;
     int index= c->index;
     int frac= c->frac;
@@ -277,13 +260,21 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
                 val += src[sample_index + i] * (FELEM2)filter[i];
             }
         }
-
+        if (val>1){
+        	val=1;
+        } if (val<-1){
+        	val=-1;
+        };
+		dst[dst_index] = val;
+/*
 #ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
-        dst[dst_index] = av_clip_int16(lrintf(val));
+      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;
 #endif
+*/
 
         frac += dst_incr_frac;
         index += dst_incr;