reduce firmware and sun audio codec.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25140 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 27 insertions, 46 deletions

diff --git a/apps/metadata/au.c b/apps/metadata/au.c
index 0639bd1..94e7453 100644
--- a/apps/metadata/au.c
+++ b/apps/metadata/au.c
@@ -20,8 +20,6 @@
  ****************************************************************************/
 #include <stdio.h>
 #include <string.h>
-#include <stdlib.h>
-#include <ctype.h>
 #include <inttypes.h>
 
 #include "system.h"
@@ -30,62 +28,42 @@
 #include "metadata_parsers.h"
 #include "logf.h"
 
-/* table of bits per sample / 8 */
-static const unsigned char bitspersamples[28] = {
-    0,
-    1,  /* G.711 MULAW */
-    1,  /* 8bit  */
-    2,  /* 16bit */
-    3,  /* 24bit */
-    4,  /* 32bit */
-    4,  /* 32bit */
-    8,  /* 64bit */
-    0,  /* Fragmented sample data */
-    0,  /* DSP program */
-    0,  /* 8bit fixed point */
-    0,  /* 16bit fixed point */
-    0,  /* 24bit fixed point */
-    0,  /* 32bit fixed point */
-    0,
-    0,
-    0,
-    0,
-    0,  /* 16bit linear with emphasis */
-    0,  /* 16bit linear compressed */
-    0,  /* 16bit linear with emphasis and compression */
-    0,  /* Music kit DSP commands */
-    0,
-    0,  /* G.721 MULAW */
-    0,  /* G.722 */
-    0,  /* G.723 3bit */
-    0,  /* G.723 5bit */
-    1,  /* G.711 ALAW */
+static const unsigned char bitspersamples[9] = {
+    0,  /* encoding */
+    8,  /* 1:  G.711 MULAW      */
+    8,  /* 2:  Linear PCM 8bit  */
+    16, /* 3:  Linear PCM 16bit */
+    24, /* 4:  Linear PCM 24bit */
+    32, /* 5:  Linear PCM 32bit */
+    32, /* 6:  IEEE float 32bit */
+    64, /* 7:  IEEE float 64bit */
+        /* encoding 8 - 26 unsupported. */
+    8,  /* 27: G.711 ALAW       */
 };
 
 static inline unsigned char get_au_bitspersample(unsigned int encoding)
 {
-    if (encoding > 27)
-        return 0;
-    return bitspersamples[encoding];
+    if (encoding < 8)
+        return bitspersamples[encoding];
+    else if (encoding == 27)
+        return bitspersamples[8];
+
+    return 0;
 }
 
 bool get_au_metadata(int fd, struct mp3entry* id3)
 {
-    /* Use the trackname part of the id3 structure as a temporary buffer */
+    /* temporary buffer */
     unsigned char* buf = (unsigned char *)id3->path;
     unsigned long numbytes = 0;
-    int read_bytes;
     int offset;
-    unsigned char bits_ch; /* bitspersample * channels */
 
     id3->vbr      = false;   /* All Sun audio files are CBR */
     id3->filesize = filesize(fd);
     id3->length   = 0;
 
-    if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 24)) < 0))
-        return false;
-
-    if (read_bytes < 24 || (memcmp(buf, ".snd", 4) != 0))
+    lseek(fd, 0, SEEK_SET);
+    if ((read(fd, buf, 24) < 24) || (memcmp(buf, ".snd", 4) != 0))
     {
         /*
          * no header
@@ -96,10 +74,12 @@ bool get_au_metadata(int fd, struct mp3entry* id3)
          */
         numbytes = id3->filesize;
         id3->frequency = 8000;
-        bits_ch = 1;
+        id3->bitrate   = 8;
     }
     else
     {
+        /* parse header */
+
         /* data offset */
         offset = get_long_be(buf + 4);
         if (offset < 24)
@@ -112,13 +92,14 @@ bool get_au_metadata(int fd, struct mp3entry* id3)
         if (numbytes == (uint32_t)0xffffffff)
             numbytes = id3->filesize - offset;
 
-        bits_ch = get_au_bitspersample(get_long_be(buf + 12)) * get_long_be(buf + 20);
         id3->frequency = get_long_be(buf + 16);
+        id3->bitrate = get_au_bitspersample(get_long_be(buf + 12)) * get_long_be(buf + 20)
+                                                                   * id3->frequency / 1000;
     }
 
     /* Calculate track length [ms] */
-    if (bits_ch)
-        id3->length = ((int64_t)numbytes * 1000LL) / (bits_ch * id3->frequency);
+    if (id3->bitrate)
+        id3->length = (numbytes << 3) / id3->bitrate;
 
     return true;
 }