rockbox/apps/codecs/aiff.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (c) 2005 Jvo Studer
*
* 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 <inttypes.h>
CODEC_HEADER
/* Macro that sign extends an unsigned byte */
#define SE(x) ((int32_t)((int8_t)(x)))
/* This codec supports AIFF files with the following formats:
* - PCM, 8, 16 and 24 bits, mono or stereo
*/
enum
{
AIFF_FORMAT_PCM = 0x0001, /* AIFF PCM Format (big endian) */
IEEE_FORMAT_FLOAT = 0x0003, /* IEEE Float */
AIFF_FORMAT_ALAW = 0x0004, /* AIFC ALaw compressed */
AIFF_FORMAT_ULAW = 0x0005 /* AIFC uLaw compressed */
};
/* Maximum number of bytes to process in one iteration */
/* for 44.1kHz stereo 16bits, this represents 0.023s ~= 1/50s */
#define AIF_CHUNK_SIZE (1024*2)
static int32_t samples[AIF_CHUNK_SIZE] IBSS_ATTR;
enum codec_status codec_main(void)
{
uint32_t numbytes, bytesdone;
uint16_t num_channels = 0;
uint32_t num_sample_frames = 0;
uint16_t sample_size = 0;
uint32_t sample_rate = 0;
uint32_t i;
size_t n;
int bufcount;
int endofstream;
unsigned char *buf;
uint8_t *aifbuf;
long chunksize;
uint32_t offset2snd = 0;
uint16_t block_size = 0;
uint32_t avgbytespersec = 0;
off_t firstblockposn; /* position of the first block in file */
/* 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);
/* assume the AIFF header is less than 1024 bytes */
buf = ci->request_buffer(&n, 1024);
if (n < 54) {
i = CODEC_ERROR;
goto done;
}
if ((memcmp(buf, "FORM", 4) != 0) || (memcmp(&buf[8], "AIFF", 4) != 0)) {
i = CODEC_ERROR;
goto done;
}
buf += 12;
n -= 12;
numbytes = 0;
/* read until 'SSND' chunk, which typically is last */
while (numbytes == 0 && n >= 8) {
/* chunkSize */
i = ((buf[4]<<24)|(buf[5]<<16)|(buf[6]<<8)|buf[7]);
if (memcmp(buf, "COMM", 4) == 0) {
if (i < 18) {
DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu < 18\n",
(unsigned long)i);
i = CODEC_ERROR;
goto done;
}
/* num_channels */
num_channels = ((buf[8]<<8)|buf[9]);
/* num_sample_frames */
num_sample_frames = ((buf[10]<<24)|(buf[11]<<16)|(buf[12]<<8)
|buf[13]);
/* sample_size */
sample_size = ((buf[14]<<8)|buf[15]);
/* sample_rate (don't use last 4 bytes, only integer fs) */
if (buf[16] != 0x40) {
DEBUGF("CODEC_ERROR: weird sampling rate (no @)\n");
i = CODEC_ERROR;
goto done;
}
sample_rate = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
sample_rate = sample_rate >> (16 + 14 - buf[17]);
/* calc average bytes per second */
avgbytespersec = sample_rate*num_channels*sample_size/8;
} else if (memcmp(buf, "SSND", 4)==0) {
if (sample_size == 0) {
DEBUGF("CODEC_ERROR: unsupported chunk order\n");
i = CODEC_ERROR;
goto done;
}
/* offset2snd */
offset2snd = (buf[8]<<24)|(buf[9]<<16)|(buf[10]<<8)|buf[11];
/* block_size */
block_size = (buf[12]<<24)|(buf[13]<<16)|(buf[14]<<8)|buf[15];
if (block_size == 0)
block_size = num_channels*sample_size;
numbytes = i - 8 - offset2snd;
i = 8 + offset2snd; /* advance to the beginning of data */
} else {
DEBUGF("unsupported AIFF chunk: '%c%c%c%c', size=%lu\n",
buf[0], buf[1], buf[2], buf[3], (unsigned long)i);
}
if (i & 0x01) /* odd chunk sizes must be padded */
i++;
buf += i + 8;
if (n < (i + 8)) {
DEBUGF("CODEC_ERROR: AIFF header size > 1024\n");
i = CODEC_ERROR;
goto done;
}
n -= i + 8;
} /* while 'SSND' */
if (num_channels == 0) {
DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
i = CODEC_ERROR;
goto done;
}
if (numbytes == 0) {
DEBUGF("CODEC_ERROR: 'SSND' chunk not found or has zero length\n");
i = CODEC_ERROR;
goto done;
}
if (sample_size > 24) {
DEBUGF("CODEC_ERROR: PCM with more than 24 bits per sample "
"is unsupported\n");
i = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (num_channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (num_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;
/* chunksize is computed so that one chunk is about 1/50s.
* this make 4096 for 44.1kHz 16bits stereo.
* It also has to be a multiple of blockalign */
chunksize = (1 + avgbytespersec/(50*block_size))*block_size;
/* check that the output buffer is big enough (convert to samplespersec,
then round to the block_size multiple below) */
if (((uint64_t)chunksize*ci->id3->frequency*num_channels*2)
/(uint64_t)avgbytespersec >= AIF_CHUNK_SIZE) {
chunksize = ((uint64_t)AIF_CHUNK_SIZE*avgbytespersec
/((uint64_t)ci->id3->frequency*num_channels*2
*block_size))*block_size;
}
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
uint32_t newpos;
/* use avgbytespersec to round to the closest blockalign multiple,
add firstblockposn. 64-bit casts to avoid overflows. */
newpos = (((uint64_t)avgbytespersec*(ci->seek_time - 1))
/(1000LL*block_size))*block_size;
if (newpos > numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos))
bytesdone = newpos;
ci->seek_complete();
}
aifbuf = (uint8_t *)ci->request_buffer(&n, chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > numbytes) {
n = numbytes - bytesdone;
endofstream = 1;
}
if (sample_size > 24) {
for (i = 0; i < n; i += 4) {
samples[i/4] = (SE(aifbuf[i])<<21)|(aifbuf[i + 1]<<13)
|(aifbuf[i + 2]<<5)|(aifbuf[i + 3]>>3);
}
bufcount = n >> 2;
} else if (sample_size > 16) {
for (i = 0; i < n; i += 3) {
samples[i/3] = (SE(aifbuf[i])<<21)|(aifbuf[i + 1]<<13)
|(aifbuf[i + 2]<<5);
}
bufcount = n/3;
} else if (sample_size > 8) {
for (i = 0; i < n; i += 2)
samples[i/2] = (SE(aifbuf[i])<<21)|(aifbuf[i + 1]<<13);
bufcount = n >> 1;
} else {
for (i = 0; i < n; i++)
samples[i] = SE(aifbuf[i]) << 21;
bufcount = n;
}
if (num_channels == 2)
bufcount >>= 1;
ci->pcmbuf_insert(samples, NULL, bufcount);
ci->advance_buffer(n);
bytesdone += n;
if (bytesdone >= numbytes)
endofstream = 1;
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
}
i = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return i;
}