more horizontal refactoring

[swftools.git] / lib / modules / swfsound.c

diff --git a/lib/modules/swfsound.c b/lib/modules/swfsound.c
index 935814c..d76baec 100644 (file)
--- a/lib/modules/swfsound.c
+++ b/lib/modules/swfsound.c
@@ -60,7 +60,7 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first)
     char*buf;
     int len = 0;
 
-    buf = malloc(init->bufferSize);
+    buf = rfx_alloc(init->bufferSize);
     if(!buf)
        return;
     
@@ -73,10 +73,17 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int numsamples, char first)
        swf_SetU16(tag, 0); // seek
     }
     swf_SetBlock(tag, buf, len);
-    free(buf);
+    rfx_free(buf);
 }
 #endif
 
+void swf_SetSoundDefineRaw(TAG*tag, S16*samples, int numsamples)
+{
+    swf_SetU8(tag,(/*compression*/0<<4)|(/*rate*/3<<2)|(/*size*/1<<1)|/*mono*/0);
+    swf_SetU32(tag, numsamples); // 44100 -> 11025
+    swf_SetBlock(tag, (U8*)samples, numsamples*2);
+}
+
 /* TODO: find a better way to set these from the outside */
 
 int swf_mp3_in_samplerate = 44100;
@@ -88,7 +95,7 @@ int swf_mp3_bitrate = 32;
 #define HAVE_SOUND
 
 #include <stdarg.h>
-#include "../lame/lame.h"
+#include <lame.h>
 
 static lame_global_flags*lame_flags;
 
@@ -164,7 +171,7 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
     int numsamples = (int)(((swf_mp3_out_samplerate > 22050) ? 1152 : 576) * ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate));
     int fs = 0;
 
-    buf = malloc(bufsize);
+    buf = rfx_alloc(bufsize);
     if(!buf)
        return;
 
@@ -184,7 +191,7 @@ void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
        fprintf(stderr, "ok: mp3 nonempty block, %d samples, first:%d, framesize:%d\n",
                numsamples, first, fs);
     }*/
-    free(buf);
+    rfx_free(buf);
 }
 
 void swf_SetSoundStreamEnd(TAG*tag)
@@ -192,12 +199,6 @@ void swf_SetSoundStreamEnd(TAG*tag)
     lame_close (lame_flags);
 }
 
-void swf_SetSoundDefineRaw(TAG*tag, S16*samples, int num, int samplerate)
-{
-    //swf_SetU8(tag,(/*compression*/0<<4)|(/*rate*/3<<2)|(/*size*/1<<1)|/*mono*/0);
-    //swf_SetU32(tag, numsamples); // 44100 -> 11025
-    //swf_SetBlock(tag, wav2.data, numsamples*2);
-}
 void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
 {
     char*buf;
@@ -226,7 +227,7 @@ void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
            ((double)swf_mp3_in_samplerate/swf_mp3_out_samplerate)) // account for resampling
            );
 
-    buf = malloc(bufsize);
+    buf = rfx_alloc(bufsize);
     if(!buf)
        return;
 
@@ -243,7 +244,7 @@ void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
        len = 0;
     }
 
-    free(buf);
+    rfx_free(buf);
 }
 
 #endif
@@ -256,23 +257,19 @@ void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
 
 void swf_SetSoundStreamHead(TAG*tag, int avgnumsamples)
 {
-    fprintf(stderr, "Error: no sound support compiled in.\n");exit(1);
+    fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
 }
 void swf_SetSoundStreamBlock(TAG*tag, S16*samples, int seek, char first)
 {
-    fprintf(stderr, "Error: no sound support compiled in.\n");exit(1);
+    fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
 }
 void swf_SetSoundStreamEnd(TAG*tag)
 {
-    fprintf(stderr, "Error: no sound support compiled in.\n");exit(1);
-}
-void swf_SetSoundDefineRaw(TAG*tag, S16*samples, int num, int samplerate)
-{
-    fprintf(stderr, "Error: no sound support compiled in.\n");exit(1);
+    fprintf(stderr, "Error: no mp3 soundstream support compiled in.\n");exit(1);
 }
 void swf_SetSoundDefine(TAG*tag, S16*samples, int num)
 {
-    fprintf(stderr, "Error: no sound support compiled in.\n");exit(1);
+    swf_SetSoundDefineRaw(tag, samples,num);
 }
 
 #endif
@@ -312,3 +309,25 @@ void swf_SetSoundInfo(TAG*tag, SOUNDINFO*info)
 }
 
 
+void swf_SetSoundDefineMP3(TAG*tag, U8* data, unsigned length,
+                           unsigned SampRate,
+                           unsigned Channels,
+                           unsigned NumFrames)
+{
+    U8 compression = 2; // 0 = raw, 1 = ADPCM, 2 = mp3, 3 = raw le, 6 = nellymoser
+    U8 rate;     // 0 = 5.5 Khz, 1 = 11 Khz, 2 = 22 Khz, 3 = 44 Khz
+    U8 size = 1; // 0 = 8 bit, 1 = 16 bit
+    U8 type = Channels==2; // 0=mono, 1=stereo
+    
+    rate = (SampRate >= 40000) ? 3
+         : (SampRate >= 19000) ? 2
+         : (SampRate >= 8000) ? 1
+         : 0;
+
+    swf_SetU8(tag,(compression<<4)|(rate<<2)|(size<<1)|type);
+
+    swf_SetU32(tag, NumFrames * 576);
+
+    swf_SetU16(tag, 0); //delayseek
+    swf_SetBlock(tag, data, length);
+}