rockbox/apps/codecs/aiff.c
Daniel Stenberg 2acc0ac542 Updated our source code header to explicitly mention that we are GPL v2 or
later. We still need to hunt down snippets used that are not. 1324 modified
files...
http://www.rockbox.org/mail/archive/rockbox-dev-archive-2008-06/0060.shtml


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@17847 a1c6a512-1295-4272-9138-f99709370657
2008-06-28 18:10:04 +00:00

275 lines
8.8 KiB
C

/***************************************************************************
* __________ __ ___.
* 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);
ci->configure(CODEC_SET_FILEBUF_WATERMARK, 1024*512);
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;
}