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Commit FS#10422 by Yoshihisa Uchida. Seperates WAV and AIFF parsing from PCM decoding by introducing libpcm, a library for decoding linear and non-uniform PCM independently of the container format.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24346 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Michael Giacomelli 2010-01-27 17:25:10 +00:00
parent 7a50f6f274
commit c9183bf15e
11 changed files with 1113 additions and 611 deletions
Download patch file Download diff file Expand all files Collapse all files

215
apps/codecs/aiff.c
View file

@ -8,6 +8,7 @@
* $Id$
*
* Copyright (c) 2005 Jvo Studer
* Copyright (c) 2009 Yoshihisa Uchida
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -21,48 +22,62 @@
#include "codeclib.h"
#include <inttypes.h>
#include "codecs/libpcm/support_formats.h"
CODEC_HEADER
/* Macro that sign extends an unsigned byte */
#define SE(x) ((int32_t)((int8_t)(x)))
#define FOURCC(c1, c2, c3, c4) \
((((uint32_t)c1)<<24)|(((uint32_t)c2)<<16)|(((uint32_t)c3)<<8)|((uint32_t)c4))
/* 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 */
/* This codec supports the following AIFC compressionType formats */
enum {
AIFC_FORMAT_PCM = FOURCC('N', 'O', 'N', 'E'), /* AIFC PCM Format (big endian) */
AIFC_FORMAT_ALAW = FOURCC('a', 'l', 'a', 'w'), /* AIFC ALaw compressed */
AIFC_FORMAT_MULAW = FOURCC('u', 'l', 'a', 'w'), /* 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 const struct pcm_entry pcm_codecs[] = {
{ AIFC_FORMAT_PCM, get_linear_pcm_codec },
{ AIFC_FORMAT_ALAW, get_itut_g711_alaw_codec },
{ AIFC_FORMAT_MULAW, get_itut_g711_mulaw_codec },
};
static int32_t samples[AIF_CHUNK_SIZE] IBSS_ATTR;
#define NUM_FORMATS 3
static int32_t samples[PCM_CHUNK_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;
}
enum codec_status codec_main(void)
{
uint32_t numbytes, bytesdone;
uint16_t num_channels = 0;
int status = CODEC_OK;
struct pcm_format format;
uint32_t bytesdone, decodedbytes;
uint32_t num_sample_frames = 0;
uint16_t sample_size = 0;
uint32_t sample_rate = 0;
uint32_t i;
uint32_t i = CODEC_OK;
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 */
bool is_aifc = false;
const struct pcm_codec *codec;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
@ -84,45 +99,80 @@ next_track:
i = CODEC_ERROR;
goto done;
}
if ((memcmp(buf, "FORM", 4) != 0) || (memcmp(&buf[8], "AIFF", 4) != 0)) {
if (memcmp(buf, "FORM", 4) != 0)
{
DEBUGF("CODEC_ERROR: does not aiff format %c%c%c%c\n", buf[0], buf[1], buf[2], buf[3]);
i = CODEC_ERROR;
goto done;
}
if (memcmp(&buf[8], "AIFF", 4) == 0)
is_aifc = false;
else if (memcmp(&buf[8], "AIFC", 4) == 0)
is_aifc = true;
else
{
DEBUGF("CODEC_ERROR: does not aiff format %c%c%c%c\n", buf[8], buf[9], buf[10], buf[11]);
i = CODEC_ERROR;
goto done;
}
buf += 12;
n -= 12;
numbytes = 0;
ci->memset(&format, 0, sizeof(struct pcm_format));
format.is_signed = true;
format.is_little_endian = false;
decodedbytes = 0;
codec = 0;
/* read until 'SSND' chunk, which typically is last */
while (numbytes == 0 && n >= 8) {
while (format.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);
if ((!is_aifc && i < 18) || (is_aifc && i < 22))
{
DEBUGF("CODEC_ERROR: 'COMM' chunk size=%lu < %d\n",
(unsigned long)i, (is_aifc)?22:18);
i = CODEC_ERROR;
goto done;
}
/* num_channels */
num_channels = ((buf[8]<<8)|buf[9]);
format.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]);
format.bitspersample = ((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]);
format.samplespersec = ((buf[18]<<24)|(buf[19]<<16)|(buf[20]<<8)|buf[21])+1;
format.samplespersec >>= (16 + 14 - buf[17]);
/* compressionType (AIFC only) */
if (is_aifc)
{
format.formattag = (buf[26]<<24)|(buf[27]<<16)|(buf[28]<<8)|buf[29];
/*
* aiff's sample_size is uncompressed sound data size.
* But format.bitspersample is compressed sound data size.
*/
if (format.formattag == AIFC_FORMAT_ALAW || format.formattag == AIFC_FORMAT_MULAW)
format.bitspersample = 8;
}
else
format.formattag = AIFC_FORMAT_PCM;
/* calc average bytes per second */
avgbytespersec = sample_rate*num_channels*sample_size/8;
format.avgbytespersec = format.samplespersec*format.channels*format.bitspersample/8;
} else if (memcmp(buf, "SSND", 4)==0) {
if (sample_size == 0) {
if (format.bitspersample == 0) {
DEBUGF("CODEC_ERROR: unsupported chunk order\n");
i = CODEC_ERROR;
goto done;
@ -130,11 +180,14 @@ next_track:
/* 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;
format.blockalign = (buf[12]<<24)|(buf[13]<<16)|(buf[14]<<8)|buf[15];
if (format.blockalign == 0)
format.blockalign = format.channels*format.bitspersample;
format.numbytes = i - 8 - offset2snd;
i = 8 + offset2snd; /* advance to the beginning of data */
} else if (is_aifc && (memcmp(buf, "FVER", 4)==0)) {
/* Format Version Chunk (AIFC only chunk) */
/* skip this chunk */
} else {
DEBUGF("unsupported AIFF chunk: '%c%c%c%c', size=%lu\n",
buf[0], buf[1], buf[2], buf[3], (unsigned long)i);
@ -151,28 +204,36 @@ next_track:
n -= i + 8;
} /* while 'SSND' */
if (num_channels == 0) {
if (format.channels == 0) {
DEBUGF("CODEC_ERROR: 'COMM' chunk not found or 0-channels file\n");
i = CODEC_ERROR;
goto done;
}
if (numbytes == 0) {
if (format.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");
codec = get_codec(format.formattag);
if (codec == 0)
{
DEBUGF("CODEC_ERROR: AIFC does not support compressionType: 0x%x\n", format.formattag);
i = CODEC_ERROR;
goto done;
}
if (!codec->set_format(&format, 0))
{
i = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (num_channels == 2) {
if (format.channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (num_channels == 1) {
} else if (format.channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
} else {
DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
@ -187,18 +248,6 @@ next_track:
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();
@ -206,61 +255,39 @@ next_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)
uint32_t newpos = codec->get_seek_pos(ci->seek_time);
if (newpos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos))
bytesdone = newpos;
ci->seek_complete();
}
aifbuf = (uint8_t *)ci->request_buffer(&n, chunksize);
aifbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > numbytes) {
n = numbytes - bytesdone;
if (bytesdone + n > format.numbytes) {
n = format.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;
status = codec->decode(aifbuf, n, samples, &bufcount);
if (status == CODEC_ERROR)
{
DEBUGF("codec error\n");
goto done;
}
if (num_channels == 2)
bufcount >>= 1;
ci->pcmbuf_insert(samples, NULL, bufcount);
ci->advance_buffer(n);
bytesdone += n;
if (bytesdone >= numbytes)
decodedbytes += bufcount;
if (bytesdone >= format.numbytes)
endofstream = 1;
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
ci->set_elapsed(decodedbytes*1000LL/ci->id3->frequency);
}
i = CODEC_OK;

3
apps/codecs/codecs.make
View file

@ -39,6 +39,7 @@ include $(APPSDIR)/codecs/libwma/libwma.make
include $(APPSDIR)/codecs/libcook/libcook.make
include $(APPSDIR)/codecs/librm/librm.make
include $(APPSDIR)/codecs/libatrac/libatrac.make
include $(APPSDIR)/codecs/libpcm/libpcm.make
# compile flags for codecs
CODECFLAGS = $(filter-out -fno-strict-aliasing,$(CFLAGS)) -fstrict-aliasing \
@ -85,6 +86,8 @@ $(CODECDIR)/cook.codec : $(CODECDIR)/libcook.a $(CODECDIR)/librm.a
$(CODECDIR)/raac.codec : $(CODECDIR)/libfaad.a $(CODECDIR)/librm.a
$(CODECDIR)/a52_rm.codec : $(CODECDIR)/liba52.a $(CODECDIR)/librm.a
$(CODECDIR)/atrac3_rm.codec : $(CODECDIR)/libatrac.a $(CODECDIR)/librm.a
$(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a
$(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
$(CODECS): $(CODECLIB) # this must be last in codec dependency list

3
apps/codecs/libpcm/SOURCES Normal file
View file

@ -0,0 +1,3 @@
linear_pcm.c
itut_g711.c
dvi_adpcm.c

309
apps/codecs/libpcm/dvi_adpcm.c Executable file
View file

@ -0,0 +1,309 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
* Copyright (C) 2009 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 "pcm_common.h"
/*
* Intel DVI ADPCM
*/
static const uint16_t dvi_adpcm_steptab[89] ICONST_ATTR = {
7, 8, 9, 10, 11, 12, 13, 14,
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767 };
static const int dvi_adpcm_indextab4[8] ICONST_ATTR = {
-1, -1, -1, -1, 2, 4, 6, 8 };
static const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
static struct pcm_format *fmt;
static bool set_format(struct pcm_format *format, const unsigned char *fmtpos)
{
fmt = format;
(void)fmtpos;
if (fmt->bitspersample != 4 && fmt->bitspersample != 3)
{
DEBUGF("CODEC_ERROR: dvi_adpcm must have 3 or 4 bitspersample\n");
return false;
}
if (fmt->size < 2) {
DEBUGF("CODEC_ERROR: dvi_adpcm is missing SamplesPerBlock value\n");
return false;
}
/* 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 */
fmt->chunksize = (1 + fmt->avgbytespersec / (50*fmt->blockalign))*fmt->blockalign;
/* check that the output buffer is big enough (convert to samplespersec,
then round to the blockalign multiple below) */
if ((((uint64_t)fmt->chunksize * ci->id3->frequency * fmt->channels * fmt->bitspersample)>>3)
/(uint64_t)fmt->avgbytespersec >= PCM_CHUNK_SIZE)
fmt->chunksize = ((uint64_t)PCM_CHUNK_SIZE * fmt->avgbytespersec
/((uint64_t)ci->id3->frequency * fmt->channels * 2
* fmt->blockalign)) * fmt->blockalign;
return true;
}
static uint32_t get_seek_pos(long 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)fmt->avgbytespersec*(seek_time - 1))
/ (1000LL*fmt->blockalign))*fmt->blockalign;
return newpos;
}
static int decode_dvi_adpcm(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, size_t *outbufcount)
{
size_t nsamples = 0;
int sample[2];
int samplecode[32][2];
int i;
int stepindex[2];
int c;
int diff;
int step;
int codem;
int code;
if (fmt->bitspersample != 4 && fmt->bitspersample != 3) {
DEBUGF("decode_dvi_adpcm: wrong bitspersample\n");
return CODEC_ERROR;
}
/* decode block header */
for (c = 0; c < fmt->channels && inbufsize >= 4; c++) {
/* decode + push first sample */
sample[c] = (short)(inbuf[0]|(inbuf[1]<<8));/* need cast for sign-extend */
outbuf[c] = sample[c] << 13;
nsamples++;
stepindex[c] = inbuf[2];
/* check for step table index overflow */
if (stepindex[c] > 88) {
DEBUGF("decode_dvi_adpcm: stepindex[%d]=%d>88\n",c,stepindex[c]);
return CODEC_ERROR;
}
inbuf += 4;
inbufsize -= 4;
}
if (fmt->bitspersample == 4) {
while (inbufsize >= (size_t)(fmt->channels*4) &&
(nsamples + (fmt->channels*8) <= *outbufcount))
{
for (c = 0; c < fmt->channels; c++)
{
samplecode[0][c] = inbuf[0]&0xf;
samplecode[1][c] = inbuf[0]>>4;
samplecode[2][c] = inbuf[1]&0xf;
samplecode[3][c] = inbuf[1]>>4;
samplecode[4][c] = inbuf[2]&0xf;
samplecode[5][c] = inbuf[2]>>4;
samplecode[6][c] = inbuf[3]&0xf;
samplecode[7][c] = inbuf[3]>>4;
inbuf += 4;
inbufsize -= 4;
}
for (i = 0; i < 8; i++)
{
for (c = 0; c < fmt->channels; c++)
{
step = dvi_adpcm_steptab[stepindex[c]];
codem = samplecode[i][c];
code = codem & 0x07;
/* adjust the step table index */
stepindex[c] += dvi_adpcm_indextab4[code];
/* check for step table index overflow and underflow */
if (stepindex[c] > 88)
stepindex[c] = 88;
else if (stepindex[c] < 0)
stepindex[c] = 0;
/* calculate the difference */
#ifdef STRICT_IMA
diff = 0;
if (code & 4)
diff += step;
step = step >> 1;
if (code & 2)
diff += step;
step = step >> 1;
if (code & 1)
diff += step;
step = step >> 1;
diff += step;
#else
diff = ((code + code + 1) * step) >> 3; /* faster */
#endif
/* check the sign bit */
/* check for overflow and underflow errors */
if (code != codem)
{
sample[c] -= diff;
if (sample[c] < -32768)
sample[c] = -32768;
}
else
{
sample[c] += diff;
if (sample[c] > 32767)
sample[c] = 32767;
}
/* output the new sample */
outbuf[nsamples] = sample[c] << 13;
nsamples++;
}
}
}
} else { /* bitspersample == 3 */
while (inbufsize >= (uint32_t)(fmt->channels*12) &&
(nsamples + 32*fmt->channels) <= *outbufcount) {
for (c = 0; c < fmt->channels; c++) {
uint16_t bitstream = 0;
int bitsread = 0;
for (i = 0; i < 32 && inbufsize > 0; i++) {
if (bitsread < 3) {
/* read 8 more bits */
bitstream |= inbuf[0]<<bitsread;
bitsread += 8;
inbufsize--;
inbuf++;
}
samplecode[i][c] = bitstream & 7;
bitstream = bitstream>>3;
bitsread -= 3;
}
if (bitsread != 0) {
/* 32*3 = 3 words, so we should end with bitsread==0 */
DEBUGF("decode_dvi_adpcm: error in implementation\n");
return CODEC_ERROR;
}
}
for (i = 0; i < 32; i++) {
for (c = 0; c < fmt->channels; c++) {
step = dvi_adpcm_steptab[stepindex[c]];
codem = samplecode[i][c];
code = codem & 0x03;
/* adjust the step table index */
stepindex[c] += dvi_adpcm_indextab3[code];
/* check for step table index overflow and underflow */
if (stepindex[c] > 88)
stepindex[c] = 88;
else if (stepindex[c] < 0)
stepindex[c] = 0;
/* calculate the difference */
#ifdef STRICT_IMA
diff = 0;
if (code & 2)
diff += step;
step = step >> 1;
if (code & 1)
diff += step;
step = step >> 1;
diff += step;
#else
diff = ((code + code + 1) * step) >> 3; /* faster */
#endif
/* check the sign bit */
/* check for overflow and underflow errors */
if (code != codem) {
sample[c] -= diff;
if (sample[c] < -32768)
sample[c] = -32768;
}
else {
sample[c] += diff;
if (sample[c] > 32767)
sample[c] = 32767;
}
/* output the new sample */
outbuf[nsamples] = sample[c] << 13;
nsamples++;
}
}
}
}
if (nsamples > *outbufcount) {
DEBUGF("decode_dvi_adpcm: output buffer overflow!\n");
return CODEC_ERROR;
}
*outbufcount = nsamples;
if (inbufsize != 0) {
DEBUGF("decode_dvi_adpcm: n=%d unprocessed bytes\n", (int)inbufsize);
}
return CODEC_OK;
}
static int decode(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, int *outbufsize)
{
unsigned int i;
unsigned int nblocks = fmt->chunksize / fmt->blockalign;
(void)inbufsize;
for (i = 0; i < nblocks; i++)
{
size_t decodedsize = fmt->samplesperblock * fmt->channels;
if (decode_dvi_adpcm(inbuf + i * fmt->blockalign, fmt->blockalign,
outbuf + i * fmt->samplesperblock * fmt->channels,
&decodedsize) != CODEC_OK) {
return CODEC_ERROR;
}
}
*outbufsize = nblocks * fmt->samplesperblock;
return CODEC_OK;
}
static const struct pcm_codec codec = {
set_format,
get_seek_pos,
decode,
};
const struct pcm_codec *get_dvi_adpcm_codec(void)
{
return &codec;
}

203
apps/codecs/libpcm/itut_g711.c Normal file
View file

@ -0,0 +1,203 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
* Copyright (C) 2009 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 "pcm_common.h"
/*
* ITU-T G.711 A-law mu-law
*/
static const int16_t alaw2linear16[256] ICONST_ATTR = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992,
-4736, -7552, -7296, -8064, -7808, -6528, -6272,
-7040, -6784, -2752, -2624, -3008, -2880, -2240,
-2112, -2496, -2368, -3776, -3648, -4032, -3904,
-3264, -3136, -3520, -3392, -22016, -20992, -24064,
-23040, -17920, -16896, -19968, -18944, -30208, -29184,
-32256, -31232, -26112, -25088, -28160, -27136, -11008,
-10496, -12032, -11520, -8960, -8448, -9984, -9472,
-15104, -14592, -16128, -15616, -13056, -12544, -14080,
-13568, -344, -328, -376, -360, -280, -264,
-312, -296, -472, -456, -504, -488, -408,
-392, -440, -424, -88, -72, -120, -104,
-24, -8, -56, -40, -216, -200, -248,
-232, -152, -136, -184, -168, -1376, -1312,
-1504, -1440, -1120, -1056, -1248, -1184, -1888,
-1824, -2016, -1952, -1632, -1568, -1760, -1696,
-688, -656, -752, -720, -560, -528, -624,
-592, -944, -912, -1008, -976, -816, -784,
-880, -848, 5504, 5248, 6016, 5760, 4480,
4224, 4992, 4736, 7552, 7296, 8064, 7808,
6528, 6272, 7040, 6784, 2752, 2624, 3008,
2880, 2240, 2112, 2496, 2368, 3776, 3648,
4032, 3904, 3264, 3136, 3520, 3392, 22016,
20992, 24064, 23040, 17920, 16896, 19968, 18944,
30208, 29184, 32256, 31232, 26112, 25088, 28160,
27136, 11008, 10496, 12032, 11520, 8960, 8448,
9984, 9472, 15104, 14592, 16128, 15616, 13056,
12544, 14080, 13568, 344, 328, 376, 360,
280, 264, 312, 296, 472, 456, 504,
488, 408, 392, 440, 424, 88, 72,
120, 104, 24, 8, 56, 40, 216,
200, 248, 232, 152, 136, 184, 168,
1376, 1312, 1504, 1440, 1120, 1056, 1248,
1184, 1888, 1824, 2016, 1952, 1632, 1568,
1760, 1696, 688, 656, 752, 720, 560,
528, 624, 592, 944, 912, 1008, 976,
816, 784, 880, 848
};
static const int16_t ulaw2linear16[256] ICONST_ATTR = {
-32124, -31100, -30076, -29052, -28028, -27004, -25980,
-24956, -23932, -22908, -21884, -20860, -19836, -18812,
-17788, -16764, -15996, -15484, -14972, -14460, -13948,
-13436, -12924, -12412, -11900, -11388, -10876, -10364,
-9852, -9340, -8828, -8316, -7932, -7676, -7420,
-7164, -6908, -6652, -6396, -6140, -5884, -5628,
-5372, -5116, -4860, -4604, -4348, -4092, -3900,
-3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108,
-1980, -1884, -1820, -1756, -1692, -1628, -1564,
-1500, -1436, -1372, -1308, -1244, -1180, -1116,
-1052, -988, -924, -876, -844, -812, -780,
-748, -716, -684, -652, -620, -588, -556,
-524, -492, -460, -428, -396, -372, -356,
-340, -324, -308, -292, -276, -260, -244,
-228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72,
-64, -56, -48, -40, -32, -24, -16,
-8, 0, 32124, 31100, 30076, 29052, 28028,
27004, 25980, 24956, 23932, 22908, 21884, 20860,
19836, 18812, 17788, 16764, 15996, 15484, 14972,
14460, 13948, 13436, 12924, 12412, 11900, 11388,
10876, 10364, 9852, 9340, 8828, 8316, 7932,
7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348,
4092, 3900, 3772, 3644, 3516, 3388, 3260,
3132, 3004, 2876, 2748, 2620, 2492, 2364,
2236, 2108, 1980, 1884, 1820, 1756, 1692,
1628, 1564, 1500, 1436, 1372, 1308, 1244,
1180, 1116, 1052, 988, 924, 876, 844,
812, 780, 748, 716, 684, 652, 620,
588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276,
260, 244, 228, 212, 196, 180, 164,
148, 132, 120, 112, 104, 96, 88,
80, 72, 64, 56, 48, 40, 32,
24, 16, 8, 0
};
static struct pcm_format *fmt;
static bool set_format(struct pcm_format *format, const unsigned char *fmtpos)
{
fmt = format;
(void)fmtpos;
if (fmt->bitspersample != 8)
{
DEBUGF("CODEC_ERROR: alaw and mulaw must have 8 bitspersample\n");
return false;
}
if (fmt->totalsamples == 0)
{
fmt->bytespersample = fmt->channels;
fmt->totalsamples = fmt->numbytes/fmt->bytespersample;
}
/* 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 */
fmt->chunksize = (1 + fmt->avgbytespersec / (50*fmt->blockalign))*fmt->blockalign;
/* check that the output buffer is big enough (convert to samplespersec,
then round to the blockalign multiple below) */
if ((((uint64_t)fmt->chunksize * ci->id3->frequency * fmt->channels * fmt->bitspersample)>>3)
/(uint64_t)fmt->avgbytespersec >= PCM_CHUNK_SIZE)
fmt->chunksize = ((uint64_t)PCM_CHUNK_SIZE * fmt->avgbytespersec
/((uint64_t)ci->id3->frequency * fmt->channels * 2
* fmt->blockalign)) * fmt->blockalign;
return true;
}
static uint32_t get_seek_pos(long 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)fmt->avgbytespersec*(seek_time - 1))
/ (1000LL*fmt->blockalign))*fmt->blockalign;
return newpos;
}
static int decode_alaw(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, int *outbufsize)
{
uint32_t i;
for (i = 0; i < inbufsize; i++)
outbuf[i] = alaw2linear16[inbuf[i]] << 13;
*outbufsize = (fmt->channels == 2) ? (inbufsize >> 1) : inbufsize;
return CODEC_OK;
}
static int decode_mulaw(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, int *outbufsize)
{
uint32_t i;
for (i = 0; i < inbufsize; i++)
outbuf[i] = ulaw2linear16[inbuf[i]] << 13;
*outbufsize = (fmt->channels == 2) ? (inbufsize >> 1) : inbufsize;
return CODEC_OK;
}
static const struct pcm_codec alaw_codec = {
set_format,
get_seek_pos,
decode_alaw,
};
static const struct pcm_codec mulaw_codec = {
set_format,
get_seek_pos,
decode_mulaw,
};
const struct pcm_codec *get_itut_g711_alaw_codec(void)
{
return &alaw_codec;
}
const struct pcm_codec *get_itut_g711_mulaw_codec(void)
{
return &mulaw_codec;
}

25
apps/codecs/libpcm/libpcm.make Normal file
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@ -0,0 +1,25 @@
# __________ __ ___.
# Open \______ \ ____ ____ | | _\_ |__ _______ ___
# Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
# Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
# Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
# \/ \/ \/ \/ \/
# $Id$
#
# libpcm
PCMSLIB := $(CODECDIR)/libpcm.a
PCMSLIB_SRC := $(call preprocess, $(APPSDIR)/codecs/libpcm/SOURCES)
PCMSLIB_OBJ := $(call c2obj, $(PCMSLIB_SRC))
OTHER_SRC += $(PCMSLIB_SRC)
$(PCMSLIB): $(PCMSLIB_OBJ)
$(SILENT)$(shell rm -f $@)
$(call PRINTS,AR $(@F))$(AR) rcs $@ $^ >/dev/null
PCMSFLAGS = $(filter-out -O%,$(CODECFLAGS))
PCMSFLAGS += -O1
$(CODECDIR)/libpcm/%.o: $(ROOTDIR)/apps/codecs/libpcm/%.c
$(SILENT)mkdir -p $(dir $@)
$(call PRINTS,CC $(subst $(ROOTDIR)/,,$<))$(CC) $(PCMSFLAGS) -c $< -o $@

163
apps/codecs/libpcm/linear_pcm.c Normal file
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@ -0,0 +1,163 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
* Copyright (C) 2009 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 "pcm_common.h"
/*
* Linear PCM
*/
static struct pcm_format *fmt;
static bool set_format(struct pcm_format *format, const unsigned char *fmtpos)
{
fmt = format;
(void)fmtpos;
if (fmt->bitspersample > 32)
{
DEBUGF("CODEC_ERROR: pcm with more than 32 bitspersample "
"is unsupported\n");
return false;
}
if (fmt->totalsamples == 0)
{
fmt->bytespersample = (((fmt->bitspersample - 1)/8 + 1)*fmt->channels);
fmt->totalsamples = fmt->numbytes/fmt->bytespersample;
}
/* 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 */
fmt->chunksize = (1 + fmt->avgbytespersec / (50*fmt->blockalign))*fmt->blockalign;
return true;
}
static uint32_t get_seek_pos(long 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)fmt->avgbytespersec*(seek_time - 1))
/ (1000LL*fmt->blockalign))*fmt->blockalign;
return newpos;
}
static int decode(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, int *outbufsize)
{
uint32_t i;
if (fmt->bitspersample > 24)
{
for (i = 0; i < inbufsize; i += 4)
{
if (fmt->is_little_endian)
{
if (fmt->is_signed)
outbuf[i/4] = (inbuf[i]>>3)|(inbuf[i+1]<<5)|(inbuf[i+2]<<13)|(SE(inbuf[i+3])<<21);
else
outbuf[i/4] = (inbuf[i]>>3)|(inbuf[i+1]<<5)|(inbuf[i+2]<<13)|(SFT(inbuf[i+3])<<21);
}
else
{
if (fmt->is_signed)
outbuf[i/4] = (inbuf[i+3]>>3)|(inbuf[i+2]<<5)|(inbuf[i+1]<<13)|(SE(inbuf[i])<<21);
else
outbuf[i/4] = (inbuf[i+3]>>3)|(inbuf[i+2]<<5)|(inbuf[i+1]<<13)|(SFT(inbuf[i])<<21);
}
}
*outbufsize = inbufsize >> 2;
}
else if (fmt->bitspersample > 16)
{
for (i = 0; i < inbufsize; i += 3)
{
if (fmt->is_little_endian)
{
if (fmt->is_signed)
outbuf[i/3] = (inbuf[i]<<5)|(inbuf[i+1]<<13)|(SE(inbuf[i+2])<<21);
else
outbuf[i/3] = (inbuf[i]<<5)|(inbuf[i+1]<<13)|(SFT(inbuf[i+2])<<21);
}
else
{
if (fmt->is_signed)
outbuf[i/3] = (inbuf[i+2]<<5)|(inbuf[i+1]<<13)|(SE(inbuf[i])<<21);
else
outbuf[i/3] = (inbuf[i+2]<<5)|(inbuf[i+1]<<13)|(SFT(inbuf[i])<<21);
}
}
*outbufsize = inbufsize/3;
}
else if (fmt->bitspersample > 8)
{
for (i = 0; i < inbufsize; i += 2)
{
if (fmt->is_little_endian)
{
if (fmt->is_signed)
outbuf[i/2] = (inbuf[i]<<13)|(SE(inbuf[i+1])<<21);
else
outbuf[i/2] = (inbuf[i]<<13)|(SFT(inbuf[i+1])<<21);
}
else
{
if (fmt->is_signed)
outbuf[i/2] = (inbuf[i+1]<<13)|(SE(inbuf[i])<<21);
else
outbuf[i/2] = (inbuf[i+1]<<13)|(SFT(inbuf[i])<<21);
}
}
*outbufsize = inbufsize >> 1;
}
else
{
for (i = 0; i < inbufsize; i++) {
if (fmt->is_signed)
outbuf[i] = SE(inbuf[i])<<21;
else
outbuf[i] = SFT(inbuf[i])<<21;
}
*outbufsize = inbufsize;
}
if (fmt->channels == 2)
*outbufsize >>= 1;
return CODEC_OK;
}
static const struct pcm_codec codec = {
set_format,
get_seek_pos,
decode,
};
const struct pcm_codec *get_linear_pcm_codec(void)
{
return &codec;
}

120
apps/codecs/libpcm/pcm_common.h Normal file
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@ -0,0 +1,120 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 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.
*
****************************************************************************/
#ifndef CODEC_LIBPCMS_PCM_COMMON_H
#define CODEC_LIBPCMS_PCM_COMMON_H
#include <sys/types.h>
#include <stdbool.h>
#include <inttypes.h>
/*
* PCM_CHUNK_SIZE has the size only of storing the sample at 1/50 seconds.
* But it might not be 1/50 seconds according to the format.
* Please confirm the source file of each format.
*/
#define PCM_CHUNK_SIZE (4096*2)
/* Macro that sign extends an unsigned byte */
#define SE(x) ((int32_t)((int8_t)(x)))
/* Macro that shift to -0x80. (0 .. 127 to -128 .. -1, 128 .. 255 to 0 .. 127) */
#define SFT(x) ((int32_t)x-0x80)
struct pcm_format {
/*
* RIFF: wFormatTag (in 'fmt ' chunk)
* AIFF: compressionType (in 'COMM' chunk)
*/
uint32_t formattag;
/*
* RIFF: wChannels (in 'fmt ' chunk)
* AIFF: numChannels (in 'COMM' chunk)
*/
uint16_t channels;
/*
* RIFF: dwSamplesPerSec (in 'fmt ' chunk)
* AIFF: sampleRate (in 'COMM' chunk)
*/
uint32_t samplespersec;
/* RIFF: dwAvgBytesPerSec (in 'fmt ' chunk) */
uint32_t avgbytespersec;
/*
* RIFF: wBlockAlign (in 'fmt ' chunk)
* AIFF: blockSize (in 'SSND' chunk)
*/
uint16_t blockalign;
/*
* RIFF: wBitsPerSample (in 'fmt ' chunk)
* AIFF: sampleSize (in 'COMM' chunk)
*/
uint16_t bitspersample;
/* RIFF: wSize (in 'fmt ' chunk) */
uint16_t size;
/* RIFF: dSamplesPerBlock (in 'fmt ' chunk) */
uint16_t samplesperblock;
/* RIFF: wTotalSamples (in 'fact' chunk) */
uint16_t totalsamples;
/* the following values are not RIFF/AIFF chunk values */
/* bytes per sample */
int bytespersample;
/* chunk size */
long chunksize;
/* data size */
uint32_t numbytes;
/*
* data endian
* true: little endian, false: big endian
*/
bool is_little_endian;
/*
* data signess
* true: signed, false: unsigned
*/
bool is_signed;
};
struct pcm_codec {
bool (*set_format)(struct pcm_format *format, const unsigned char *fmtpos);
uint32_t (*get_seek_pos)(long seek_time);
int (*decode)(const uint8_t *inbuf, size_t inbufsize,
int32_t *outbuf, int *outbufsize);
};
struct pcm_entry {
uint32_t format_tag;
const struct pcm_codec *(*get_codec)(void);
};
#endif

37
apps/codecs/libpcm/support_formats.h Normal file
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@ -0,0 +1,37 @@
/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 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.
*
****************************************************************************/
#ifndef CODEC_LIBPCMS_SUPPORT_FORMATS_H
#define CODEC_LIBPCMS_SUPPORT_FORMATS_H
#include "pcm_common.h"
/* Linear PCM */
const struct pcm_codec *get_linear_pcm_codec(void);
/* ITU-T G.711 A-law */
const struct pcm_codec *get_itut_g711_alaw_codec(void);
/* ITU-T G.711 mu-law */
const struct pcm_codec *get_itut_g711_mulaw_codec(void);
/* Intel DVI ADPCM */
const struct pcm_codec *get_dvi_adpcm_codec(void);
#endif

642
apps/codecs/wav.c
View file

@ -8,6 +8,7 @@
* $Id$
*
* Copyright (C) 2005 Dave Chapman
* Copyright (C) 2009 Yoshihisa Uchida
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
@ -21,16 +22,11 @@
#include "codeclib.h"
#include "inttypes.h"
#include "codecs/libpcm/support_formats.h"
CODEC_HEADER
/* Macro that sign extends an unsigned byte */
#define SE(x) ((int32_t)((int8_t)(x)))
/* This codec support WAVE files with the following formats:
* - PCM, up to 32 bits, supporting 32 bits playback when useful.
* - ALAW and MULAW (16 bits compressed on 8 bits).
* - DVI_ADPCM (16 bits compressed on 3 or 4 bits).
/* WAVE (RIFF) codec:
*
* For a good documentation on WAVE files, see:
* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
@ -40,198 +36,74 @@ CODEC_HEADER
* For sample WAV files, see:
* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/Samples.html
*
* The most common formats seem to be PCM, ADPCM, DVI_ADPCM, IEEE_FLOAT,
* ALAW and MULAW
*/
/* These constants are from RFC 2361. */
static int32_t samples[PCM_CHUNK_SIZE] IBSS_ATTR;
/* 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_VSELP = 0x0004, /* Compaq Computer's VSELP */
WAVE_FORMAT_IBM_CVSD = 0x0005, /* IBM CVSD */
WAVE_FORMAT_ALAW = 0x0006, /* Microsoft ALAW */
WAVE_FORMAT_MULAW = 0x0007, /* Microsoft MULAW */
WAVE_FORMAT_OKI_ADPCM = 0x0010, /* OKI ADPCM */
WAVE_FORMAT_DVI_ADPCM = 0x0011, /* Intel's DVI ADPCM */
WAVE_FORMAT_MEDIASPACE_ADPCM = 0x0012, /* Videologic's MediaSpace ADPCM */
WAVE_FORMAT_SIERRA_ADPCM = 0x0013, /* Sierra ADPCM */
WAVE_FORMAT_G723_ADPCM = 0x0014, /* G.723 ADPCM */
WAVE_FORMAT_DIGISTD = 0x0015, /* DSP Solutions' DIGISTD */
WAVE_FORMAT_DIGIFIX = 0x0016, /* DSP Solutions' DIGIFIX */
WAVE_FORMAT_DIALOGIC_OKI_ADPCM = 0x0017, /* Dialogic OKI ADPCM */
WAVE_FORMAT_MEDIAVISION_ADPCM = 0x0018, /* MediaVision ADPCM */
WAVE_FORMAT_CU_CODEC = 0x0019, /* HP CU */
WAVE_FORMAT_YAMAHA_ADPCM = 0x0020, /* Yamaha ADPCM */
WAVE_FORMAT_SONARC = 0x0021, /* Speech Compression's Sonarc */
WAVE_FORMAT_DSP_TRUESPEECH = 0x0022, /* DSP Group's True Speech */
WAVE_FORMAT_ECHOSC1 = 0x0023, /* Echo Speech's EchoSC1 */
WAVE_FORMAT_AUDIOFILE_AF36 = 0x0024, /* Audiofile AF36 */
WAVE_FORMAT_APTX = 0x0025, /* APTX */
WAVE_FORMAT_DOLBY_AC2 = 0x0030, /* Dolby AC2 */
WAVE_FORMAT_GSM610 = 0x0031, /* GSM610 */
WAVE_FORMAT_MSNAUDIO = 0x0032, /* MSNAudio */
WAVE_FORMAT_ANTEX_ADPCME = 0x0033, /* Antex ADPCME */
WAVE_FORMAT_MPEG = 0x0050, /* MPEG */
WAVE_FORMAT_MPEGLAYER3 = 0x0055, /* MPEG layer 3 */
WAVE_FORMAT_LUCENT_G723 = 0x0059, /* Lucent G.723 */
WAVE_FORMAT_G726_ADPCM = 0x0064, /* G.726 ADPCM */
WAVE_FORMAT_G722_ADPCM = 0x0065, /* G.722 ADPCM */
IBM_FORMAT_MULAW = 0x0101, /* same as WAVE_FORMAT_MULAW */
IBM_FORMAT_ALAW = 0x0102, /* same as WAVE_FORMAT_ALAW */
IBM_FORMAT_ADPCM = 0x0103,
WAVE_FORMAT_CREATIVE_ADPCM = 0x0200,
WAVE_FORMAT_EXTENSIBLE = 0xFFFE
};
/* Maximum number of bytes to process in one iteration */
/* for 44.1kHz stereo 16bits, this represents 0.023s ~= 1/50s */
#define WAV_CHUNK_SIZE (1024*2)
static const int16_t alaw2linear16[256] ICONST_ATTR = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992,
-4736, -7552, -7296, -8064, -7808, -6528, -6272,
-7040, -6784, -2752, -2624, -3008, -2880, -2240,
-2112, -2496, -2368, -3776, -3648, -4032, -3904,
-3264, -3136, -3520, -3392, -22016, -20992, -24064,
-23040, -17920, -16896, -19968, -18944, -30208, -29184,
-32256, -31232, -26112, -25088, -28160, -27136, -11008,
-10496, -12032, -11520, -8960, -8448, -9984, -9472,
-15104, -14592, -16128, -15616, -13056, -12544, -14080,
-13568, -344, -328, -376, -360, -280, -264,
-312, -296, -472, -456, -504, -488, -408,
-392, -440, -424, -88, -72, -120, -104,
-24, -8, -56, -40, -216, -200, -248,
-232, -152, -136, -184, -168, -1376, -1312,
-1504, -1440, -1120, -1056, -1248, -1184, -1888,
-1824, -2016, -1952, -1632, -1568, -1760, -1696,
-688, -656, -752, -720, -560, -528, -624,
-592, -944, -912, -1008, -976, -816, -784,
-880, -848, 5504, 5248, 6016, 5760, 4480,
4224, 4992, 4736, 7552, 7296, 8064, 7808,
6528, 6272, 7040, 6784, 2752, 2624, 3008,
2880, 2240, 2112, 2496, 2368, 3776, 3648,
4032, 3904, 3264, 3136, 3520, 3392, 22016,
20992, 24064, 23040, 17920, 16896, 19968, 18944,
30208, 29184, 32256, 31232, 26112, 25088, 28160,
27136, 11008, 10496, 12032, 11520, 8960, 8448,
9984, 9472, 15104, 14592, 16128, 15616, 13056,
12544, 14080, 13568, 344, 328, 376, 360,
280, 264, 312, 296, 472, 456, 504,
488, 408, 392, 440, 424, 88, 72,
120, 104, 24, 8, 56, 40, 216,
200, 248, 232, 152, 136, 184, 168,
1376, 1312, 1504, 1440, 1120, 1056, 1248,
1184, 1888, 1824, 2016, 1952, 1632, 1568,
1760, 1696, 688, 656, 752, 720, 560,
528, 624, 592, 944, 912, 1008, 976,
816, 784, 880, 848
const struct pcm_entry wave_codecs[] = {
{ WAVE_FORMAT_UNKNOWN, 0 },
{ WAVE_FORMAT_PCM, get_linear_pcm_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 },
{ IBM_FORMAT_MULAW, get_itut_g711_mulaw_codec },
{ IBM_FORMAT_ALAW, get_itut_g711_alaw_codec },
};
static const int16_t ulaw2linear16[256] ICONST_ATTR = {
-32124, -31100, -30076, -29052, -28028, -27004, -25980,
-24956, -23932, -22908, -21884, -20860, -19836, -18812,
-17788, -16764, -15996, -15484, -14972, -14460, -13948,
-13436, -12924, -12412, -11900, -11388, -10876, -10364,
-9852, -9340, -8828, -8316, -7932, -7676, -7420,
-7164, -6908, -6652, -6396, -6140, -5884, -5628,
-5372, -5116, -4860, -4604, -4348, -4092, -3900,
-3772, -3644, -3516, -3388, -3260, -3132, -3004,
-2876, -2748, -2620, -2492, -2364, -2236, -2108,
-1980, -1884, -1820, -1756, -1692, -1628, -1564,
-1500, -1436, -1372, -1308, -1244, -1180, -1116,
-1052, -988, -924, -876, -844, -812, -780,
-748, -716, -684, -652, -620, -588, -556,
-524, -492, -460, -428, -396, -372, -356,
-340, -324, -308, -292, -276, -260, -244,
-228, -212, -196, -180, -164, -148, -132,
-120, -112, -104, -96, -88, -80, -72,
-64, -56, -48, -40, -32, -24, -16,
-8, 0, 32124, 31100, 30076, 29052, 28028,
27004, 25980, 24956, 23932, 22908, 21884, 20860,
19836, 18812, 17788, 16764, 15996, 15484, 14972,
14460, 13948, 13436, 12924, 12412, 11900, 11388,
10876, 10364, 9852, 9340, 8828, 8316, 7932,
7676, 7420, 7164, 6908, 6652, 6396, 6140,
5884, 5628, 5372, 5116, 4860, 4604, 4348,
4092, 3900, 3772, 3644, 3516, 3388, 3260,
3132, 3004, 2876, 2748, 2620, 2492, 2364,
2236, 2108, 1980, 1884, 1820, 1756, 1692,
1628, 1564, 1500, 1436, 1372, 1308, 1244,
1180, 1116, 1052, 988, 924, 876, 844,
812, 780, 748, 716, 684, 652, 620,
588, 556, 524, 492, 460, 428, 396,
372, 356, 340, 324, 308, 292, 276,
260, 244, 228, 212, 196, 180, 164,
148, 132, 120, 112, 104, 96, 88,
80, 72, 64, 56, 48, 40, 32,
24, 16, 8, 0
};
#define NUM_FORMATS 7
static const uint16_t dvi_adpcm_steptab[89] ICONST_ATTR = {
7, 8, 9, 10, 11, 12, 13, 14,
16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66,
73, 80, 88, 97, 107, 118, 130, 143,
157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658,
724, 796, 876, 963, 1060, 1166, 1282, 1411,
1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024,
3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484,
7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767 };
static const struct pcm_codec *get_wave_codec(uint32_t formattag)
{
int i;
static const int dvi_adpcm_indextab4[8] ICONST_ATTR = {
-1, -1, -1, -1, 2, 4, 6, 8 };
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 const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
static int32_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
static enum codec_status
decode_dvi_adpcm(struct codec_api *ci,
const uint8_t *buf,
int n,
uint16_t channels, uint16_t bitspersample,
int32_t *pcmout,
size_t *pcmoutsize);
/* this is the codec entry point */
enum codec_status codec_main(void)
{
uint32_t numbytes, bytesdone;
uint32_t totalsamples = 0;
uint16_t channels = 0;
uint16_t samplesperblock = 0;
int bytespersample = 0;
uint16_t bitspersample;
int status = CODEC_OK;
struct pcm_format format;
uint32_t bytesdone, decodedbytes;
uint32_t i;
size_t n;
int bufcount;
int endofstream;
unsigned char *buf;
uint8_t *wavbuf;
long chunksize;
uint16_t formattag = 0;
uint16_t blockalign = 0;
uint32_t avgbytespersec = 0;
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;
DEBUGF("codec_init() error\n");
status = CODEC_ERROR;
goto exit;
}
@ -246,11 +118,12 @@ next_track:
/* get RIFF chunk header */
buf = ci->request_buffer(&n, 12);
if (n < 12) {
i = CODEC_ERROR;
DEBUGF("request_buffer error\n");
status = CODEC_ERROR;
goto done;
}
if ((memcmp(buf, "RIFF", 4) != 0) || (memcmp(&buf[8], "WAVE", 4) != 0)) {
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
@ -258,17 +131,21 @@ next_track:
ci->advance_buffer(12);
firstblockposn = 12;
bitspersample = 0;
numbytes = 0;
totalsamples = 0;
ci->memset(&format, 0, sizeof(struct pcm_format));
format.is_signed = true;
format.is_little_endian = true;
decodedbytes = 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 */
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
@ -278,54 +155,70 @@ next_track:
if (i < 16) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk size=%lu < 16\n",
(unsigned long)i);
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
/* wFormatTag */
formattag=buf[8]|(buf[9]<<8);
format.formattag=buf[8]|(buf[9]<<8);
/* wChannels */
channels=buf[10]|(buf[11]<<8);
format.channels=buf[10]|(buf[11]<<8);
/* skipping dwSamplesPerSec */
/* dwAvgBytesPerSec */
avgbytespersec = buf[16]|(buf[17]<<8)|(buf[18]<<16)|(buf[19]<<24);
format.avgbytespersec = buf[16]|(buf[17]<<8)|(buf[18]<<16)|(buf[19]<<24);
/* wBlockAlign */
blockalign=buf[20]|(buf[21]<<8);
format.blockalign=buf[20]|(buf[21]<<8);
/* wBitsPerSample */
bitspersample=buf[22]|(buf[23]<<8);
if (formattag != WAVE_FORMAT_PCM) {
uint16_t size;
format.bitspersample=buf[22]|(buf[23]<<8);
if (format.formattag != WAVE_FORMAT_PCM) {
if (i < 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=%ld<18).\n",
formattag, (long)i);
format.formattag, (long)i);
}
size = buf[24]|(buf[25]<<8);
if (formattag == WAVE_FORMAT_DVI_ADPCM) {
if (size < 2) {
DEBUGF("CODEC_ERROR: dvi_adpcm is missing "
"SamplesPerBlock value\n");
i = CODEC_ERROR;
goto done;
else
{
format.size = buf[24]|(buf[25]<<8);
if (format.formattag != WAVE_FORMAT_EXTENSIBLE)
format.samplesperblock = buf[26]|(buf[27]<<8);
else {
if (format.size < 22) {
DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
"missing extension\n");
status = CODEC_ERROR;
goto done;
}
/* wValidBitsPerSample */
format.bitspersample = buf[26]|(buf[27]<<8);
/* skipping dwChannelMask (4bytes) */
/* SubFormat (only get the first two bytes) */
format.formattag = buf[32]|(buf[33]<<8);
}
samplesperblock = buf[26]|(buf[27]<<8);
} else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
if (size < 22) {
DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
"missing extension\n");
i = CODEC_ERROR;
goto done;
}
/* wValidBitsPerSample */
bitspersample = buf[26]|(buf[27]<<8);
/* skipping dwChannelMask (4bytes) */
/* SubFormat (only get the first two bytes) */
formattag = buf[32]|(buf[33]<<8);
}
}
/* get codec */
codec = get_wave_codec(format.formattag);
if (!codec)
{
DEBUGF("CODEC_ERROR: unsupport wave format %x\n", 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;
/* set format, parse codec specific tag, check format, and calculate chunk size */
if (!codec->set_format(&format, buf))
{
status = CODEC_ERROR;
goto done;
}
} else if (memcmp(buf, "data", 4) == 0) {
numbytes = i;
format.numbytes = i;
/* advance to start of data */
ci->advance_buffer(8);
firstblockposn += 8;
@ -333,7 +226,8 @@ next_track:
} else if (memcmp(buf, "fact", 4) == 0) {
/* dwSampleLength */
if (i >= 4)
totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
format.totalsamples =
(buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
} else {
DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
buf[0], buf[1], buf[2], buf[3], (unsigned long)i);
@ -346,71 +240,40 @@ next_track:
firstblockposn += i + 8;
}
if (channels == 0) {
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");
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
if (numbytes == 0) {
if (format.numbytes == 0) {
DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
i = CODEC_ERROR;
goto done;
}
if (formattag != WAVE_FORMAT_PCM && totalsamples == 0) {
/* This is non-fatal for some formats */
DEBUGF("CODEC_WARNING: non-PCM WAVE doesn't have a 'fact' chunk\n");
}
if (formattag == WAVE_FORMAT_ALAW || formattag == WAVE_FORMAT_MULAW ||
formattag == IBM_FORMAT_ALAW || formattag == IBM_FORMAT_MULAW) {
if (bitspersample != 8) {
DEBUGF("CODEC_ERROR: alaw and mulaw must have 8 bitspersample\n");
i = CODEC_ERROR;
goto done;
}
bytespersample = channels;
}
if (formattag == WAVE_FORMAT_DVI_ADPCM
&& bitspersample != 4 && bitspersample != 3) {
DEBUGF("CODEC_ERROR: dvi_adpcm must have 3 or 4 bitspersample\n");
i = CODEC_ERROR;
goto done;
}
if (formattag == WAVE_FORMAT_PCM && bitspersample > 32) {
DEBUGF("CODEC_ERROR: pcm with more than 32 bitspersample "
"is unsupported\n");
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (channels == 2) {
if (format.channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (channels == 1) {
} else if (format.channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
} else {
DEBUGF("CODEC_ERROR: more than 2 channels\n");
i = CODEC_ERROR;
status = CODEC_ERROR;
goto done;
}
if (totalsamples == 0) {
if (formattag == WAVE_FORMAT_PCM ||
formattag == WAVE_FORMAT_ALAW || formattag == WAVE_FORMAT_MULAW ||
formattag == IBM_FORMAT_ALAW || formattag == IBM_FORMAT_MULAW) {
/* for PCM and derived formats only */
bytespersample = (((bitspersample - 1)/8 + 1)*channels);
totalsamples = numbytes/bytespersample;
} else {
DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
i = CODEC_ERROR;
goto done;
}
}
/* make sure we're at the correct offset */
if (bytesdone > (uint32_t) firstblockposn) {
/* Round down to previous block */
uint32_t offset = bytesdone - bytesdone % blockalign;
uint32_t offset = bytesdone - bytesdone % format.blockalign;
ci->advance_buffer(offset-firstblockposn);
bytesdone = offset - firstblockposn;
@ -421,18 +284,6 @@ next_track:
/* The main decoder loop */
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*blockalign))*blockalign;
/* check that the output buffer is big enough (convert to samplespersec,
then round to the blockalign multiple below) */
if (((uint64_t)chunksize*ci->id3->frequency*channels*2)
/(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec
/((uint64_t)ci->id3->frequency*channels*2
*blockalign))*blockalign;
}
while (!endofstream) {
ci->yield();
@ -441,286 +292,47 @@ next_track:
}
if (ci->seek_time) {
uint32_t newpos;
uint32_t newpos = codec->get_seek_pos(ci->seek_time);
/* 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*blockalign))*blockalign;
if (newpos > numbytes)
if (newpos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos))
{
bytesdone = newpos;
}
ci->seek_complete();
}
wavbuf = (uint8_t *)ci->request_buffer(&n, chunksize);
wavbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > numbytes) {
n = numbytes - bytesdone;
if (bytesdone + n > format.numbytes) {
n = format.numbytes - bytesdone;
endofstream = 1;
}
if (formattag == WAVE_FORMAT_PCM) {
if (bitspersample > 24) {
for (i = 0; i < n; i += 4) {
samples[i/4] = (wavbuf[i] >> 3)|
(wavbuf[i + 1]<<5)|(wavbuf[i + 2]<<13)|
(SE(wavbuf[i + 3])<<21);
}
bufcount = n >> 2;
} else if (bitspersample > 16) {
for (i = 0; i < n; i += 3) {
samples[i/3] = (wavbuf[i]<<5)|
(wavbuf[i + 1]<<13)|(SE(wavbuf[i + 2])<<21);
}
bufcount = n/3;
} else if (bitspersample > 8) {
for (i = 0; i < n; i += 2) {
samples[i/2] = (wavbuf[i]<<13)|(SE(wavbuf[i + 1])<<21);
}
bufcount = n >> 1;
} else {
for (i = 0; i < n; i++) {
samples[i] = (wavbuf[i] - 0x80)<<21;
}
bufcount = n;
}
if (channels == 2)
bufcount >>= 1;
} else if (formattag == WAVE_FORMAT_ALAW
|| formattag == IBM_FORMAT_ALAW) {
for (i = 0; i < n; i++)
samples[i] = alaw2linear16[wavbuf[i]] << 13;
bufcount = (channels == 2) ? (n >> 1) : n;
} else if (formattag == WAVE_FORMAT_MULAW
|| formattag == IBM_FORMAT_MULAW) {
for (i = 0; i < n; i++)
samples[i] = ulaw2linear16[wavbuf[i]] << 13;
bufcount = (channels == 2) ? (n >> 1) : n;
}
else if (formattag == WAVE_FORMAT_DVI_ADPCM) {
unsigned int nblocks = chunksize/blockalign;
for (i = 0; i < nblocks; i++) {
size_t decodedsize = samplesperblock*channels;
if (decode_dvi_adpcm(ci, wavbuf + i*blockalign,
blockalign, channels, bitspersample,
samples + i*samplesperblock*channels,
&decodedsize) != CODEC_OK) {
i = CODEC_ERROR;
goto done;
}
}
bufcount = nblocks*samplesperblock;
} else {
DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
i = CODEC_ERROR;
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;
if (bytesdone >= numbytes)
decodedbytes += bufcount;
if (bytesdone >= format.numbytes)
endofstream = 1;
ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
ci->set_elapsed(decodedbytes*1000LL/ci->id3->frequency);
}
i = CODEC_OK;
status = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return i;
return status;
}
static enum codec_status
decode_dvi_adpcm(struct codec_api *ci,
const uint8_t *buf,
int n,
uint16_t channels, uint16_t bitspersample,
int32_t *pcmout,
size_t *pcmoutsize)
{
size_t nsamples = 0;
int sample[2];
int samplecode[32][2];
int i;
int stepindex[2];
int c;
int diff;
int step;
int codem;
int code;
(void)ci;
if (bitspersample != 4 && bitspersample != 3) {
DEBUGF("decode_dvi_adpcm: wrong bitspersample\n");
return CODEC_ERROR;
}
/* decode block header */
for (c = 0; c < channels && n >= 4; c++) {
/* decode + push first sample */
sample[c] = (short)(buf[0]|(buf[1]<<8));/* need cast for sign-extend */
pcmout[c] = sample[c] << 13;
nsamples++;
stepindex[c] = buf[2];
/* check for step table index overflow */
if (stepindex[c] > 88) {
DEBUGF("decode_dvi_adpcm: stepindex[%d]=%d>88\n",c,stepindex[c]);
return CODEC_ERROR;
}
buf += 4;
n -= 4;
}
if (bitspersample == 4) {
while (n>= channels*4 && (nsamples + 8*channels) <= *pcmoutsize) {
for (c = 0; c < channels; c++) {
samplecode[0][c] = buf[0]&0xf;
samplecode[1][c] = buf[0]>>4;
samplecode[2][c] = buf[1]&0xf;
samplecode[3][c] = buf[1]>>4;
samplecode[4][c] = buf[2]&0xf;
samplecode[5][c] = buf[2]>>4;
samplecode[6][c] = buf[3]&0xf;
samplecode[7][c] = buf[3]>>4;
buf += 4;
n -= 4;
}
for (i = 0; i < 8; i++) {
for (c = 0; c < channels; c++) {
step = dvi_adpcm_steptab[stepindex[c]];
codem = samplecode[i][c];
code = codem & 0x07;
/* adjust the step table index */
stepindex[c] += dvi_adpcm_indextab4[code];
/* check for step table index overflow and underflow */
if (stepindex[c] > 88)
stepindex[c] = 88;
else if (stepindex[c] < 0)
stepindex[c] = 0;
/* calculate the difference */
#ifdef STRICT_IMA
diff = 0;
if (code & 4)
diff += step;
step = step >> 1;
if (code & 2)
diff += step;
step = step >> 1;
if (code & 1)
diff += step;
step = step >> 1;
diff += step;
#else
diff = ((code + code + 1) * step) >> 3; /* faster */
#endif
/* check the sign bit */
/* check for overflow and underflow errors */
if (code != codem) {
sample[c] -= diff;
if (sample[c] < -32768)
sample[c] = -32768;
} else {
sample[c] += diff;
if (sample[c] > 32767)
sample[c] = 32767;
}
/* output the new sample */
pcmout[nsamples] = sample[c] << 13;
nsamples++;
}
}
}
} else { /* bitspersample == 3 */
while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) {
for (c = 0; c < channels; c++) {
uint16_t bitstream = 0;
int bitsread = 0;
for (i = 0; i < 32 && n > 0; i++) {
if (bitsread < 3) {
/* read 8 more bits */
bitstream |= buf[0]<<bitsread;
bitsread += 8;
n--;
buf++;
}
samplecode[i][c] = bitstream & 7;
bitstream = bitstream>>3;
bitsread -= 3;
}
if (bitsread != 0) {
/* 32*3 = 3 words, so we should end with bitsread==0 */
DEBUGF("decode_dvi_adpcm: error in implementation\n");
return CODEC_ERROR;
}
}
for (i = 0; i < 32; i++) {
for (c = 0; c < channels; c++) {
step = dvi_adpcm_steptab[stepindex[c]];
codem = samplecode[i][c];
code = codem & 0x03;
/* adjust the step table index */
stepindex[c] += dvi_adpcm_indextab3[code];
/* check for step table index overflow and underflow */
if (stepindex[c] > 88)
stepindex[c] = 88;
else if (stepindex[c] < 0)
stepindex[c] = 0;
/* calculate the difference */
#ifdef STRICT_IMA
diff = 0;
if (code & 2)
diff += step;
step = step >> 1;
if (code & 1)
diff += step;
step = step >> 1;
diff += step;
#else
diff = ((code + code + 1) * step) >> 3; /* faster */
#endif
/* check the sign bit */
/* check for overflow and underflow errors */
if (code != codem) {
sample[c] -= diff;
if (sample[c] < -32768)
sample[c] = -32768;
}
else {
sample[c] += diff;
if (sample[c] > 32767)
sample[c] = 32767;
}
/* output the new sample */
pcmout[nsamples] = sample[c] << 13;
nsamples++;
}
}
}
}
if (nsamples > *pcmoutsize) {
DEBUGF("decode_dvi_adpcm: output buffer overflow!\n");
return CODEC_ERROR;
}
*pcmoutsize = nsamples;
if (n != 0) {
DEBUGF("decode_dvi_adpcm: n=%d unprocessed bytes\n", n);
}
return CODEC_OK;
}

2
apps/metadata/aiff.c
View file

@ -48,7 +48,7 @@ bool get_aiff_metadata(int fd, struct mp3entry* id3)
}
if ((memcmp(buf, "FORM",4) != 0)
|| (memcmp(&buf[8], "AIFF", 4) !=0 ))
|| ((memcmp(&buf[8], "AIFF", 4) !=0) && (memcmp(&buf[8], "AIFC", 4) !=0)))
{
return false;
}