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|
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (c) 2010 Yoshihisa Uchida
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "codeclib.h"
#include "codecs/libpcm/support_formats.h"
CODEC_HEADER
/*
* SMAF (Synthetic music Mobile Application Format)
*
* References
* [1] YAMAHA Corporation, Synthetic music Mobile Application Format Ver.3.05, 2002
*/
enum {
SMAF_TRACK_CHUNK_SCORE = 0, /* Score Track */
SMAF_TRACK_CHUNK_AUDIO, /* PCM Audio Track */
};
/* SMAF supported codec formats */
enum {
SMAF_FORMAT_UNSUPPORT = 0, /* unsupported format */
SMAF_FORMAT_SIGNED_PCM, /* 2's complement PCM */
SMAF_FORMAT_UNSIGNED_PCM, /* Offset Binary PCM */
SMAF_FORMAT_ADPCM, /* YAMAHA ADPCM */
};
static int support_formats[2][3] = {
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM },
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT },
};
static const struct pcm_entry pcm_codecs[] = {
{ SMAF_FORMAT_SIGNED_PCM, get_linear_pcm_codec },
{ SMAF_FORMAT_UNSIGNED_PCM, get_linear_pcm_codec },
{ SMAF_FORMAT_ADPCM, get_yamaha_adpcm_codec },
};
#define NUM_FORMATS 3
static int basebits[4] = { 4, 8, 12, 16 };
#define PCM_SAMPLE_SIZE (2048*2)
static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
static const struct pcm_codec *get_codec(uint32_t formattag)
{
int i;
for (i = 0; i < NUM_FORMATS; i++)
{
if (pcm_codecs[i].format_tag == formattag)
{
if (pcm_codecs[i].get_codec)
return pcm_codecs[i].get_codec();
return 0;
}
}
return 0;
}
static unsigned int get_be32(uint8_t *buf)
{
return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
}
static int convert_smaf_audio_format(int track_chunk, unsigned int audio_format)
{
if (audio_format > 3)
return SMAF_FORMAT_UNSUPPORT;
return support_formats[track_chunk][audio_format];
}
static int convert_smaf_audio_basebit(unsigned int basebit)
{
if (basebit > 4)
return 0;
return basebits[basebit];
}
static bool parse_audio_track(struct pcm_format *fmt,
unsigned char **stbuf, unsigned char *endbuf)
{
unsigned char *buf = *stbuf;
int chunksize;
buf += 8;
fmt->channels = ((buf[2] & 0x80) >> 7) + 1;
fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_AUDIO,
(buf[2] >> 4) & 0x07);
if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
{
DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n", (buf[2] >> 4) & 0x07);
return false;
}
fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
if (fmt->bitspersample == 0)
{
DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n", buf[3] >> 4);
return false;
}
buf += 6;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "Awa", 3) == 0)
{
fmt->numbytes = get_be32(buf + 4);
buf += 8;
return true;
}
buf += chunksize;
}
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
static bool parse_score_track(struct pcm_format *fmt,
unsigned char **stbuf, unsigned char *endbuf)
{
unsigned char *buf = *stbuf;
int chunksize;
if (buf[9] != 0x00)
{
DEBUGF("CODEC_ERROR: score track chunk unsupport sequence type %d\n", buf[9]);
return false;
}
/*
* skip to the next chunk.
* MA-2/MA-3/MA-5: padding 16 bytes
* MA-7: padding 32 bytes
*/
if (buf[3] < 7)
buf += 28;
else
buf += 44;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "Mtsp", 4) == 0)
{
buf += 8;
if (memcmp(buf, "Mwa", 3) != 0)
{
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
fmt->numbytes = get_be32(buf + 4) - 3;
fmt->channels = ((buf[8] & 0x80) >> 7) + 1;
fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_SCORE,
(buf[8] >> 4) & 0x07);
if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
{
DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n",
(buf[8] >> 4) & 0x07);
return false;
}
fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0x0f);
if (fmt->bitspersample == 0)
{
DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n",
buf[8] & 0x0f);
return false;
}
buf += 11;
return true;
}
buf += chunksize;
}
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
static bool parse_header(struct pcm_format *fmt, size_t *pos)
{
unsigned char *buf, *stbuf, *endbuf;
size_t chunksize;
ci->memset(fmt, 0, sizeof(struct pcm_format));
/* assume the SMAF pcm data position is less than 1024 bytes */
stbuf = ci->request_buffer(&chunksize, 1024);
if (chunksize < 1024)
return false;
buf = stbuf;
endbuf = stbuf + chunksize;
if (memcmp(buf, "MMMD", 4) != 0)
{
DEBUGF("CODEC_ERROR: does not smaf format %c%c%c%c\n",
buf[0], buf[1], buf[2], buf[3]);
return false;
}
buf += 8;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "ATR", 3) == 0)
{
if (!parse_audio_track(fmt, &buf, endbuf))
return false;
break;
}
if (memcmp(buf, "MTR", 3) == 0)
{
if (!parse_score_track(fmt, &buf, endbuf))
return false;
break;
}
buf += chunksize;
}
if (buf >= endbuf)
{
DEBUGF("CODEC_ERROR: unsupported smaf format\n");
return false;
}
/* blockalign */
if (fmt->formattag == SMAF_FORMAT_SIGNED_PCM ||
fmt->formattag == SMAF_FORMAT_UNSIGNED_PCM)
fmt->blockalign = fmt->channels * fmt->bitspersample >> 3;
/* data signess (default signed) */
fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM);
fmt->is_little_endian = false;
/* sets pcm data position */
*pos = buf - stbuf;
return true;
}
static struct pcm_format format;
static uint32_t bytesdone;
static uint8_t *read_buffer(size_t *realsize)
{
uint8_t *buffer = (uint8_t *)ci->request_buffer(realsize, format.chunksize);
if (bytesdone + (*realsize) > format.numbytes)
*realsize = format.numbytes - bytesdone;
bytesdone += *realsize;
ci->advance_buffer(*realsize);
return buffer;
}
enum codec_status codec_main(void)
{
int status = CODEC_OK;
uint32_t decodedsamples;
uint32_t i = CODEC_OK;
size_t n;
int bufcount;
int endofstream;
uint8_t *smafbuf;
off_t firstblockposn; /* position of the first block in file */
const struct pcm_codec *codec;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
if (codec_init()) {
i = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
ci->memset(&format, 0, sizeof(struct pcm_format));
format.is_signed = true;
format.is_little_endian = false;
decodedsamples = 0;
codec = 0;
if (!parse_header(&format, &n))
{
i = CODEC_ERROR;
goto done;
}
codec = get_codec(format.formattag);
if (codec == 0)
{
DEBUGF("CODEC_ERROR: unsupport audio format: 0x%x\n", (int)format.formattag);
i = CODEC_ERROR;
goto done;
}
if (!codec->set_format(&format))
{
i = CODEC_ERROR;
goto done;
}
/* common format check */
if (format.channels == 0) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
status = CODEC_ERROR;
goto done;
}
if (format.samplesperblock == 0) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n");
status = CODEC_ERROR;
goto done;
}
if (format.blockalign == 0)
{
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n");
i = CODEC_ERROR;
goto done;
}
if (format.numbytes == 0) {
DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
status = CODEC_ERROR;
goto done;
}
/* check chunksize */
if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels
> PCM_SAMPLE_SIZE)
format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign;
if (format.chunksize == 0)
{
DEBUGF("CODEC_ERROR: chunksize is 0\n");
i = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (format.channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (format.channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
} else {
DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
i = CODEC_ERROR;
goto done;
}
firstblockposn = 1024 - n;
ci->advance_buffer(firstblockposn);
/* The main decoder loop */
bytesdone = 0;
ci->set_elapsed(0);
endofstream = 0;
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, &read_buffer);
decodedsamples = newpos->samples;
if (newpos->pos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
}
ci->seek_complete();
}
smafbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > format.numbytes) {
n = format.numbytes - bytesdone;
endofstream = 1;
}
status = codec->decode(smafbuf, n, samples, &bufcount);
if (status == CODEC_ERROR)
{
DEBUGF("codec error\n");
goto done;
}
ci->pcmbuf_insert(samples, NULL, bufcount);
ci->advance_buffer(n);
bytesdone += n;
decodedsamples += bufcount;
if (bytesdone >= format.numbytes)
endofstream = 1;
ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
}
i = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return i;
}
|