reduce firmware and sun audio codec.

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

5 files changed, 81 insertions, 98 deletions

diff --git a/apps/codecs/au.c b/apps/codecs/au.c
index cf2a799be6..19348bc299 100644
--- a/apps/codecs/au.c
+++ b/apps/codecs/au.c
@@ -45,35 +45,17 @@ enum
     AU_FORMAT_ALAW,          /* G.711 ALAW */
 };
 
-static int support_formats[28][2] = {
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_MULAW,      8  }, /* G.711 MULAW */
-  { AU_FORMAT_PCM,        8  }, /* Linear PCM 8bit (signed) */
-  { AU_FORMAT_PCM,        16 }, /* Linear PCM 16bit (signed, big endian) */
-  { AU_FORMAT_PCM,        24 }, /* Linear PCM 24bit (signed, big endian) */
-  { AU_FORMAT_PCM,        32 }, /* Linear PCM 32bit (signed, big endian) */
-  { AU_FORMAT_IEEE_FLOAT, 32 }, /* Linear PCM float 32bit (signed, big endian) */
-  { AU_FORMAT_IEEE_FLOAT, 64 }, /* Linear PCM float 64bit (signed, big endian) */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* Fragmented sample data */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* DSP program */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 8bit fixed point */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit fixed point */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 24bit fixed point */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 32bit fixed point */
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear with emphasis */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear compressed */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear with emphasis and compression */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* Music kit DSP commands */
-  { AU_FORMAT_UNSUPPORT,  0  },
-  { AU_FORMAT_UNSUPPORT,  0  }, /* G.721 MULAW */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* G.722 */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* G.723 3bit */
-  { AU_FORMAT_UNSUPPORT,  0  }, /* G.723 5bit */
-  { AU_FORMAT_ALAW,       8  }, /* G.711 ALAW */
+static const char support_formats[9][2] = {
+  { AU_FORMAT_UNSUPPORT,  0  }, /* encoding */
+  { AU_FORMAT_MULAW,      8  }, /* 1:  G.711 MULAW */
+  { AU_FORMAT_PCM,        8  }, /* 2:  Linear PCM 8bit (signed) */
+  { AU_FORMAT_PCM,        16 }, /* 3:  Linear PCM 16bit (signed, big endian) */
+  { AU_FORMAT_PCM,        24 }, /* 4:  Linear PCM 24bit (signed, big endian) */
+  { AU_FORMAT_PCM,        32 }, /* 5:  Linear PCM 32bit (signed, big endian) */
+  { AU_FORMAT_IEEE_FLOAT, 32 }, /* 6:  Linear PCM float 32bit (signed, big endian) */
+  { AU_FORMAT_IEEE_FLOAT, 64 }, /* 7:  Linear PCM float 64bit (signed, big endian) */
+                                /* encoding 8 - 26 unsupported. */
+  { AU_FORMAT_ALAW,       8  }, /* 27: G.711 ALAW */
 };
 
 const struct pcm_entry au_codecs[] = {
@@ -108,16 +90,17 @@ static unsigned int get_be32(uint8_t *buf)
 
 static int convert_au_format(unsigned int encoding, struct pcm_format *fmt)
 {
-    if (encoding > 27)
-    {
-        fmt->formattag = AU_FORMAT_UNSUPPORT;
-        fmt->bitspersample = 0;
-    }
-    else
+    fmt->formattag = AU_FORMAT_UNSUPPORT;
+    if (encoding < 8)
     {
         fmt->formattag = support_formats[encoding][0];
         fmt->bitspersample = support_formats[encoding][1];
     }
+    else if (encoding == 27)
+    {
+        fmt->formattag = support_formats[8][0];
+        fmt->bitspersample = support_formats[8][1];
+    }
 
     return fmt->formattag;
 }
@@ -138,7 +121,7 @@ enum codec_status codec_main(void)
     int offset = 0;
 
     /* Generic codec initialisation */
-    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH);
+    ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
   
 next_track:
     if (codec_init()) {
@@ -199,11 +182,6 @@ next_track:
         }
         /* skip sample rate */
         format.channels = get_be32(buf + 20);
-        if (format.channels == 0) {
-            DEBUGF("CODEC_ERROR: sun audio 0-channels file\n");
-            status = CODEC_ERROR;
-            goto done;
-        }
     }
 
     /* advance to first WAVE chunk */
@@ -215,9 +193,6 @@ next_track:
     codec = 0;
     bytesdone = 0;
 
-    /* blockalign = 1 sample */
-    format.blockalign = format.bitspersample * format.channels >> 3;
-
     /* get codec */
     codec = get_au_codec(format.formattag);
     if (!codec)
diff --git a/apps/codecs/libpcm/ieee_float.c b/apps/codecs/libpcm/ieee_float.c
index 0530993f31..7e3498edcb 100644
--- a/apps/codecs/libpcm/ieee_float.c
+++ b/apps/codecs/libpcm/ieee_float.c
@@ -32,6 +32,12 @@ static bool set_format(struct pcm_format *format)
 {
     fmt = format;
 
+    if (fmt->channels == 0)
+    {
+        DEBUGF("CODEC_ERROR: channels is 0\n");
+        return false;
+    }
+
     if (fmt->bitspersample != 32 && fmt->bitspersample != 64)
     {
         DEBUGF("CODEC_ERROR: ieee float must be 32 or 64 bitspersample: %d\n",
@@ -40,6 +46,10 @@ static bool set_format(struct pcm_format *format)
     }
 
     fmt->bytespersample = fmt->bitspersample >> 3;
+
+    if (fmt->blockalign == 0)
+        fmt->blockalign = fmt->bytespersample * fmt->channels;
+
     fmt->samplesperblock = fmt->blockalign / (fmt->bytespersample * fmt->channels);
 
     /* chunksize = about 1/50[sec] data */
diff --git a/apps/codecs/libpcm/itut_g711.c b/apps/codecs/libpcm/itut_g711.c
index 4644a9c694..097dd5cc25 100644
--- a/apps/codecs/libpcm/itut_g711.c
+++ b/apps/codecs/libpcm/itut_g711.c
@@ -112,6 +112,12 @@ static bool set_format(struct pcm_format *format)
 {
     fmt = format;
 
+    if (fmt->channels == 0)
+    {
+        DEBUGF("CODEC_ERROR: channels is 0\n");
+        return false;
+    }
+
     if (fmt->bitspersample != 8)
     {
         DEBUGF("CODEC_ERROR: alaw and mulaw must have 8 bitspersample: %d\n",
@@ -119,13 +125,12 @@ static bool set_format(struct pcm_format *format)
         return false;
     }
 
-    if (fmt->totalsamples == 0)
-    {
-        fmt->bytespersample = 1;
-        fmt->totalsamples = fmt->numbytes / (fmt->bytespersample * fmt->channels);
-    }
+    fmt->bytespersample = 1;
+
+    if (fmt->blockalign == 0)
+        fmt->blockalign = fmt->channels;
 
-    fmt->samplesperblock = fmt->blockalign / (fmt->bytespersample * fmt->channels);
+    fmt->samplesperblock = fmt->blockalign / fmt->channels;
 
     /* chunksize = about 1/50[sec] data */
     fmt->chunksize = (ci->id3->frequency / (50 * fmt->samplesperblock))
diff --git a/apps/codecs/libpcm/linear_pcm.c b/apps/codecs/libpcm/linear_pcm.c
index 82c70eb3b6..e58856efe8 100644
--- a/apps/codecs/libpcm/linear_pcm.c
+++ b/apps/codecs/libpcm/linear_pcm.c
@@ -38,6 +38,18 @@ static bool set_format(struct pcm_format *format)
 {
     fmt = format;
 
+    if (fmt->channels == 0)
+    {
+        DEBUGF("CODEC_ERROR: channels is 0\n");
+        return false;
+    }
+
+    if (fmt->bitspersample == 0)
+    {
+        DEBUGF("CODEC_ERROR: bitspersample is 0\n");
+        return false;
+    }
+
     if (fmt->bitspersample > 32)
     {
         DEBUGF("CODEC_ERROR: pcm with more than 32 bitspersample "
@@ -47,8 +59,8 @@ static bool set_format(struct pcm_format *format)
 
     fmt->bytespersample = fmt->bitspersample >> 3;
 
-    if (fmt->totalsamples == 0)
-        fmt->totalsamples = fmt->numbytes/fmt->bytespersample;
+    if (fmt->blockalign == 0)
+        fmt->blockalign = fmt->bytespersample * fmt->channels;
 
     fmt->samplesperblock = fmt->blockalign / (fmt->bytespersample * fmt->channels);
 
diff --git a/apps/metadata/au.c b/apps/metadata/au.c
index 0639bd11e6..94e7453644 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;
 }