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rockbox/apps/codecs/wav64.c
Yoshihisa Uchida 7a3822c8b0 fix bug: WAV file playback does not resume (FS#11077) 
Not only WAV but also Sun audio, SMAF, vox and WAV64 can resume.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@25289 a1c6a512-1295-4272-9138-f99709370657
2010-03-22 10:02:05 +00:00

454 lines
15 KiB
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
/* Wave64 codec
*
* References
* [1] VCS Aktiengesellschaft, Sony Wave64, Informations_about_Sony_Wave64.pdf
*/
#define PCM_SAMPLE_SIZE (4096*2)
static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
/* Wave64 GUIDs */
#define WAVE64_GUID_RIFF "riff\x2e\x91\xcf\x11\xa5\xd6\x28\xdb\x04\xc1\x00\x00"
#define WAVE64_GUID_WAVE "wave\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
#define WAVE64_GUID_FMT "fmt \xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
#define WAVE64_GUID_FACT "fact\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
#define WAVE64_GUID_DATA "data\xf3\xac\xd3\x11\x8c\xd1\x00\xc0\x4f\x8e\xdb\x8a"
/* This codec support WAVE files with the following formats: */
enum
{
WAVE_FORMAT_UNKNOWN = 0x0000, /* Microsoft Unknown Wave Format */
WAVE_FORMAT_PCM = 0x0001, /* Microsoft PCM Format */
WAVE_FORMAT_ADPCM = 0x0002, /* Microsoft ADPCM Format */
WAVE_FORMAT_IEEE_FLOAT = 0x0003, /* IEEE Float */
WAVE_FORMAT_ALAW = 0x0006, /* Microsoft ALAW */
WAVE_FORMAT_MULAW = 0x0007, /* Microsoft MULAW */
WAVE_FORMAT_DVI_ADPCM = 0x0011, /* Intel's DVI ADPCM */
WAVE_FORMAT_DIALOGIC_OKI_ADPCM = 0x0017, /* Dialogic OKI ADPCM */
WAVE_FORMAT_YAMAHA_ADPCM = 0x0020, /* Yamaha ADPCM */
WAVE_FORMAT_XBOX_ADPCM = 0x0069, /* XBOX ADPCM */
IBM_FORMAT_MULAW = 0x0101, /* same as WAVE_FORMAT_MULAW */
IBM_FORMAT_ALAW = 0x0102, /* same as WAVE_FORMAT_ALAW */
WAVE_FORMAT_SWF_ADPCM = 0x5346, /* Adobe SWF ADPCM */
WAVE_FORMAT_EXTENSIBLE = 0xFFFE
};
const struct pcm_entry wave_codecs[] = {
{ WAVE_FORMAT_UNKNOWN, 0 },
{ WAVE_FORMAT_PCM, get_linear_pcm_codec },
{ WAVE_FORMAT_ADPCM, get_ms_adpcm_codec },
{ WAVE_FORMAT_IEEE_FLOAT, get_ieee_float_codec },
{ WAVE_FORMAT_ALAW, get_itut_g711_alaw_codec },
{ WAVE_FORMAT_MULAW, get_itut_g711_mulaw_codec },
{ WAVE_FORMAT_DVI_ADPCM, get_dvi_adpcm_codec },
{ WAVE_FORMAT_DIALOGIC_OKI_ADPCM, get_dialogic_oki_adpcm_codec },
{ WAVE_FORMAT_YAMAHA_ADPCM, get_yamaha_adpcm_codec },
{ WAVE_FORMAT_XBOX_ADPCM, get_dvi_adpcm_codec },
{ IBM_FORMAT_MULAW, get_itut_g711_mulaw_codec },
{ IBM_FORMAT_ALAW, get_itut_g711_alaw_codec },
{ WAVE_FORMAT_SWF_ADPCM, get_swf_adpcm_codec },
};
#define NUM_FORMATS 13
static const struct pcm_codec *get_wave_codec(uint32_t formattag)
{
int i;
for (i = 0; i < NUM_FORMATS; i++)
{
if (wave_codecs[i].format_tag == formattag)
{
if (wave_codecs[i].get_codec)
return wave_codecs[i].get_codec();
return 0;
}
}
return 0;
}
static struct pcm_format format;
static uint32_t bytesdone;
/* Read an unaligned 64-bit little endian unsigned integer from buffer. */
static uint64_t get_uint64_le(void* buf)
{
unsigned char* p = (unsigned char*) buf;
return p[0] | (p[1] << 8) | (p[2] << 16) | (p[3] << 24) | ((uint64_t)p[4] << 32) |
((uint64_t)p[5] << 40) | ((uint64_t)p[6] << 48) | ((uint64_t)p[7] << 56);
}
static bool set_msadpcm_coeffs(unsigned char *buf)
{
int i;
int num;
uint64_t size;
buf += 16; /* skip 'fmt ' GUID */
size = get_uint64_le(buf);
if (size < 50)
{
DEBUGF("CODEC_ERROR: microsoft adpcm 'fmt ' chunk size=%d < 50\n", (int)size);
return false;
}
/* get nNumCoef */
buf += 28;
num = buf[0] | (buf[1] << 8);
/*
* In many case, nNumCoef is 7.
* Depending upon the encoder, as for this value there is a possibility of
* increasing more.
* If you found the file where this value exceeds 7, please report.
*/
if (num != MSADPCM_NUM_COEFF)
{
DEBUGF("CODEC_ERROR: microsoft adpcm nNumCoef=%d != 7\n", num);
return false;
}
/* get aCoeffs */
buf += 2;
for (i = 0; i < MSADPCM_NUM_COEFF; i++)
{
format.coeffs[i][0] = buf[0] | (SE(buf[1]) << 8);
format.coeffs[i][1] = buf[2] | (SE(buf[3]) << 8);
buf += 4;
}
return true;
}
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(void)
{
int status = CODEC_OK;
uint32_t decodedsamples;
size_t n;
int bufcount;
int endofstream;
unsigned char *buf;
uint8_t *wavbuf;
off_t firstblockposn; /* position of the first block in file */
const struct pcm_codec *codec;
uint64_t size;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
next_track:
if (codec_init()) {
DEBUGF("codec_init() error\n");
status = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
/* Need to save offset for later use (cleared indirectly by advance_buffer) */
bytesdone = ci->id3->offset;
/* get RIFF chunk header */
buf = ci->request_buffer(&n, 40);
if (n < 40) {
DEBUGF("request_buffer error\n");
status = CODEC_ERROR;
goto done;
}
if ((memcmp(buf , WAVE64_GUID_RIFF, 16) != 0) ||
(memcmp(buf+24, WAVE64_GUID_WAVE, 16) != 0))
{
status = CODEC_ERROR;
goto done;
}
/* advance to first WAVE chunk */
ci->advance_buffer(40);
firstblockposn = 40;
ci->memset(&format, 0, sizeof(struct pcm_format));
format.is_signed = true;
format.is_little_endian = true;
decodedsamples = 0;
codec = 0;
/* iterate over WAVE chunks until the 'data' chunk, which should be after the 'fmt ' chunk */
while (true) {
/* get WAVE chunk header */
buf = ci->request_buffer(&n, 1024);
if (n < 8) {
DEBUGF("data chunk request_buffer error\n");
/* no more chunks, 'data' chunk must not have been found */
status = CODEC_ERROR;
goto done;
}
/* chunkSize */
size = get_uint64_le(buf+16) - 24;
if (memcmp(buf, WAVE64_GUID_FMT, 16) == 0) {
if (size < 16) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%d < 16\n", (int)size);
status = CODEC_ERROR;
goto done;
}
/* wFormatTag */
format.formattag=buf[24]|(buf[25]<<8);
/* wChannels */
format.channels=buf[26]|(buf[27]<<8);
/* skipping dwSamplesPerSec */
/* skipping dwAvgBytesPerSec */
/* wBlockAlign */
format.blockalign=buf[36]|(buf[37]<<8);
/* wBitsPerSample */
format.bitspersample=buf[38]|(buf[39]<<8);
if (format.formattag != WAVE_FORMAT_PCM) {
if (size < 18) {
/* this is not a fatal error with some formats,
* we'll see later if we can't decode it */
DEBUGF("CODEC_WARNING: non-PCM WAVE (formattag=0x%x) "
"doesn't have ext. fmt descr (chunksize=%d<18).\n",
(unsigned int)format.formattag, (int)size);
}
else
{
if (format.formattag != WAVE_FORMAT_EXTENSIBLE)
format.samplesperblock = buf[42]|(buf[43]<<8);
else {
format.size = buf[40]|(buf[41]<<8);
if (format.size < 22) {
DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
"missing extension\n");
status = CODEC_ERROR;
goto done;
}
/* wValidBitsPerSample */
format.bitspersample = buf[42]|(buf[43]<<8);
/* skipping dwChannelMask (4bytes) */
/* SubFormat (only get the first two bytes) */
format.formattag = buf[48]|(buf[49]<<8);
}
}
}
/* msadpcm specific */
if (format.formattag == WAVE_FORMAT_ADPCM)
{
if (!set_msadpcm_coeffs(buf))
{
status = CODEC_ERROR;
goto done;
}
}
/* get codec */
codec = get_wave_codec(format.formattag);
if (!codec)
{
DEBUGF("CODEC_ERROR: unsupported wave format 0x%x\n",
(unsigned int) format.formattag);
status = CODEC_ERROR;
goto done;
}
/* riff 8bit linear pcm is unsigned */
if (format.formattag == WAVE_FORMAT_PCM && format.bitspersample == 8)
format.is_signed = false;
/* check format, and calculate chunk size */
if (!codec->set_format(&format))
{
status = CODEC_ERROR;
goto done;
}
} else if (memcmp(buf, WAVE64_GUID_DATA, 16) == 0) {
format.numbytes = size;
/* advance to start of data */
ci->advance_buffer(24);
firstblockposn += 24;
break;
} else if (memcmp(buf, WAVE64_GUID_FACT, 16) == 0) {
/* skip 'fact' chunk */
} else {
DEBUGF("unknown Wave64 chunk: "
"'%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x%02x'\n",
buf[0], buf[1], buf[ 2], buf[ 3], buf[ 4], buf[ 5], buf[ 6], buf[ 7],
buf[8], buf[9], buf[10], buf[11], buf[12], buf[13], buf[14], buf[15]);
}
/* go to next chunk (8byte bound) */
size += 24 + ((1 + ~size) & 0x07);
ci->advance_buffer(size);
firstblockposn += size;
}
if (!codec)
{
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found\n");
status = 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");
status = 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");
status = 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\n");
status = CODEC_ERROR;
goto done;
}
/* 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)
goto done;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
ci->seek_complete();
} else {
/* already where we need to be */
bytesdone = 0;
}
/* The main decoder loop */
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, PCM_SEEK_TIME,
&read_buffer);
if (newpos->pos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
ci->seek_complete();
}
wavbuf = (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(wavbuf, 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);
}
status = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return status;
}
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