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rockbox/apps/codecs/smaf.c

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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_AUDIO_TRACK_CHUNK = 0, /* PCM Audio Track */
SMAF_SCORE_TRACK_CHUNK, /* Score 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 const int support_formats[2][3] = {
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT },
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM },
};
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 const 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(const uint8_t *buf)
{
return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
}
static int convert_smaf_channels(unsigned int ch)
{
return (ch >> 7) + 1;
}
static int convert_smaf_audio_format(unsigned int chunk, unsigned int audio_format)
{
int idx = (audio_format & 0x70) >> 4;
if (idx < 3)
return support_formats[chunk][idx];
DEBUGF("CODEC_ERROR: unsupport audio format: %d\n", audio_format);
return SMAF_FORMAT_UNSUPPORT;
}
static int convert_smaf_audio_basebit(unsigned int basebit)
{
if (basebit < 4)
return basebits[basebit];
DEBUGF("CODEC_ERROR: illegal basebit: %d\n", basebit);
return 0;
}
static unsigned int search_chunk(const unsigned char *name, int nlen, off_t *pos)
{
const unsigned char *buf;
unsigned int chunksize;
size_t size;
while (true)
{
buf = ci->request_buffer(&size, 8);
if (size < 8)
break;
chunksize = get_be32(buf + 4);
ci->advance_buffer(8);
*pos += 8;
if (memcmp(buf, name, nlen) == 0)
return chunksize;
ci->advance_buffer(chunksize);
*pos += chunksize;
}
DEBUGF("CODEC_ERROR: missing '%s' chunk\n", name);
return 0;
}
static bool parse_audio_track(struct pcm_format *fmt, unsigned int chunksize, off_t *pos)
{
const unsigned char *buf;
size_t size;
/* search PCM Audio Track Chunk */
ci->advance_buffer(chunksize);
*pos += chunksize;
if (search_chunk("ATR", 3, pos) == 0)
{
DEBUGF("CODEC_ERROR: missing PCM Audio Track Chunk\n");
return false;
}
/*
* get format
* buf
* +0: Format Type
* +1: Sequence Type
* +2: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: frequency
* +3: bit 4-7: base bit
* +4: TimeBase_D
* +5: TimeBase_G
*
* Note: If PCM Audio Track does not include Sequence Data Chunk,
* tmp+6 is the start position of Wave Data Chunk.
*/
buf = ci->request_buffer(&size, 6);
if (size < 6)
{
DEBUGF("CODEC_ERROR: smaf is too small\n");
return false;
}
fmt->formattag = convert_smaf_audio_format(SMAF_AUDIO_TRACK_CHUNK, buf[2]);
fmt->channels = convert_smaf_channels(buf[2]);
fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
/* search Wave Data Chunk */
ci->advance_buffer(6);
*pos += 6;
fmt->numbytes = search_chunk("Awa", 3, pos);
if (fmt->numbytes == 0)
{
DEBUGF("CODEC_ERROR: missing Wave Data Chunk\n");
return false;
}
return true;
}
static bool parse_score_track(struct pcm_format *fmt, off_t *pos)
{
const unsigned char *buf;
unsigned int chunksize;
size_t size;
/* parse Optional Data Chunk */
buf = ci->request_buffer(&size, 13);
if (size < 13)
{
DEBUGF("CODEC_ERROR: smaf is too small\n");
return false;
}
if (memcmp(buf + 5, "OPDA", 4) != 0)
{
DEBUGF("CODEC_ERROR: missing Optional Data Chunk\n");
return false;
}
/* Optional Data Chunk size */
chunksize = get_be32(buf + 9);
/* search Score Track Chunk */
ci->advance_buffer(13 + chunksize);
*pos += (13 + chunksize);
if (search_chunk("MTR", 3, pos) == 0)
{
DEBUGF("CODEC_ERROR: missing Score Track Chunk\n");
return false;
}
/*
* search next chunk
* usually, next chunk ('M***') found within 40 bytes.
*/
buf = ci->request_buffer(&size, 40);
if (size < 40)
{
DEBUGF("CODEC_ERROR: smaf is too small\n");
return false;
}
size = 0;
while (size < 40 && buf[size] != 'M')
size++;
if (size >= 40)
{
DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk");
return false;
}
/* search Score Track Stream PCM Data Chunk */
ci->advance_buffer(size);
*pos += size;
if (search_chunk("Mtsp", 4, pos) == 0)
{
DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk\n");
return false;
}
/*
* parse Score Track Stream Wave Data Chunk
* buf
* +4-7: chunk size (WaveType(3bytes) + wave data count)
* +8: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: base bit
* +9: frequency (MSB)
* +10: frequency (LSB)
*/
buf = ci->request_buffer(&size, 9);
if (size < 9)
{
DEBUGF("CODEC_ERROR: smaf is too small\n");
return false;
}
if (memcmp(buf, "Mwa", 3) != 0)
{
DEBUGF("CODEC_ERROR: missing Score Track Stream Wave Data Chunk\n");
return false;
}
fmt->formattag = convert_smaf_audio_format(SMAF_SCORE_TRACK_CHUNK, buf[8]);
fmt->channels = convert_smaf_channels(buf[8]);
fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0xf);
fmt->numbytes = get_be32(buf + 4) - 3;
*pos += 11;
return true;
}
static bool parse_header(struct pcm_format *fmt, off_t *pos)
{
const unsigned char *buf;
unsigned int chunksize;
size_t size;
ci->memset(fmt, 0, sizeof(struct pcm_format));
/* check File Chunk and Contents Info Chunk */
buf = ci->request_buffer(&size, 16);
if (size < 16)
{
DEBUGF("CODEC_ERROR: smaf is too small\n");
return false;
}
if ((memcmp(buf, "MMMD", 4) != 0) || (memcmp(buf + 8, "CNTI", 4) != 0))
{
DEBUGF("CODEC_ERROR: does not smaf format\n");
return false;
}
chunksize = get_be32(buf + 12);
ci->advance_buffer(16);
*pos = 16;
if (chunksize > 5)
{
if (!parse_audio_track(fmt, chunksize, pos))
return false;
}
else if (!parse_score_track(fmt, pos))
return false;
/* data signess (default signed) */
fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM);
/* data is always big endian */
fmt->is_little_endian = false;
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;
}
/* this is the codec entry point */
enum codec_status codec_main(enum codec_entry_call_reason reason)
{
if (reason == CODEC_LOAD) {
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
}
return CODEC_OK;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
uint32_t decodedsamples;
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;
intptr_t param;
if (codec_init())
return CODEC_ERROR;
codec_set_replaygain(ci->id3);
/* Need to save offset for later use (cleared indirectly by advance_buffer) */
bytesdone = ci->id3->offset;
decodedsamples = 0;
codec = 0;
ci->seek_buffer(0);
if (!parse_header(&format, &firstblockposn))
{
return CODEC_ERROR;
}
codec = get_codec(format.formattag);
if (codec == 0)
{
DEBUGF("CODEC_ERROR: unsupport audio format: 0x%x\n", (int)format.formattag);
return CODEC_ERROR;
}
if (!codec->set_format(&format))
{
return CODEC_ERROR;
}
/* 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");
return CODEC_ERROR;
}
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");
return CODEC_ERROR;
}
ci->seek_buffer(firstblockposn);
/* make sure we're at the correct offset */
if (bytesdone > (uint32_t) firstblockposn)
{
/* Round down to previous block */
struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
PCM_SEEK_POS, &read_buffer);
if (newpos->pos > format.numbytes)
return CODEC_OK;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
}
else
{
/* already where we need to be */
bytesdone = 0;
}
/* The main decoder loop */
endofstream = 0;
while (!endofstream) {
enum codec_command_action action = ci->get_command(&param);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
&read_buffer);
if (newpos->pos > format.numbytes)
{
ci->set_elapsed(ci->id3->length);
ci->seek_complete();
break;
}
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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;
}
if (codec->decode(smafbuf, n, samples, &bufcount) == CODEC_ERROR)
{
DEBUGF("codec error\n");
return CODEC_ERROR;
}
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);
}
return CODEC_OK;
}
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