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reduce firmware and sun audio codec.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25140 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Yoshihisa Uchida 2010-03-13 05:19:40 +00:00
parent 131bb698ad
commit 4446d1bc85
5 changed files with 81 additions and 98 deletions
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63
apps/codecs/au.c
View file

@ -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)

10
apps/codecs/libpcm/ieee_float.c
View file

@ -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 */

17
apps/codecs/libpcm/itut_g711.c
View file

@ -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;
fmt->samplesperblock = fmt->blockalign / (fmt->bytespersample * fmt->channels);
if (fmt->blockalign == 0)
fmt->blockalign = fmt->channels;
fmt->samplesperblock = fmt->blockalign / fmt->channels;
/* chunksize = about 1/50[sec] data */
fmt->chunksize = (ci->id3->frequency / (50 * fmt->samplesperblock))

16
apps/codecs/libpcm/linear_pcm.c
View file

@ -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);

73
apps/metadata/au.c
View file

@ -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;
}