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Code policed. Converted to generate internal s3.28 format directly (next

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task is removing use of interleaved audio). Could not test ADPCM due to
difficulty in finding files it could play, so tell me if it is broken.


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@9136 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen 2006-03-20 20:32:19 +00:00
parent d8eaefe29a
commit 0968536c4b
1 changed files with 333 additions and 374 deletions
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167
apps/codecs/wav.c
View file

@ -22,12 +22,15 @@
CODEC_HEADER CODEC_HEADER
/* Macro that sign extends an unsigned byte */
#define SE(x) ((int32_t)((int8_t)(x)))
struct codec_api *rb; struct codec_api *rb;
/* This codec support WAVE files with the following formats: /* This codec support WAVE files with the following formats:
* - PCM, up to 32 bits, supporting 32 bits playback when useful. * - PCM, up to 32 bits, supporting 32 bits playback when useful.
* - ALAW and MULAW (16 bits compressed on 8 bits). * - ALAW and MULAW (16 bits compressed on 8 bits).
* - DVI_ADPCM (16 bits compressed on 4 bits). * - DVI_ADPCM (16 bits compressed on 3 or 4 bits).
* *
* For a good documentation on WAVE files, see: * For a good documentation on WAVE files, see:
* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html * http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
@ -100,8 +103,6 @@ extern char iedata[];
extern char iend[]; extern char iend[];
#endif #endif
/* Those are lookup tables, so they should be in the idata section
* (fast but small RAM on the coldfire processor) */
static const int16_t alaw2linear16[256] ICONST_ATTR = { static const int16_t alaw2linear16[256] ICONST_ATTR = {
-5504, -5248, -6016, -5760, -4480, -4224, -4992, -5504, -5248, -6016, -5760, -4480, -4224, -4992,
-4736, -7552, -7296, -8064, -7808, -6528, -6272, -4736, -7552, -7296, -8064, -7808, -6528, -6272,
@ -195,17 +196,20 @@ static const uint16_t dvi_adpcm_steptab[ 89 ] ICONST_ATTR = {
7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767 }; 32767 };
static const int dvi_adpcm_indextab4[ 8 ] ICONST_ATTR = { -1, -1, -1, -1, 2, 4, 6, 8 };
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 const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
static int16_t int16_samples[WAV_CHUNK_SIZE] IBSS_ATTR; static int32_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
static enum codec_status static enum codec_status
decode_dvi_adpcm(struct codec_api *ci, decode_dvi_adpcm(struct codec_api *ci,
const uint8_t *buf, const uint8_t *buf,
int n, int n,
uint16_t channels, uint16_t bitspersample, uint16_t channels, uint16_t bitspersample,
int16_t *pcmout, int32_t *pcmout,
size_t *pcmoutsize); size_t *pcmoutsize);
/* this is the codec entry point */ /* this is the codec entry point */
@ -219,17 +223,15 @@ enum codec_status codec_start(struct codec_api* api)
int bytespersample = 0; int bytespersample = 0;
uint16_t bitspersample; uint16_t bitspersample;
uint32_t i; uint32_t i;
size_t n, wavbufsize; size_t n, bufsize;
int endofstream; int endofstream;
unsigned char *buf; unsigned char *buf;
uint16_t* wavbuf; uint8_t *wavbuf;
long chunksize; long chunksize;
uint16_t formattag = 0; uint16_t formattag = 0;
uint16_t blockalign = 0; uint16_t blockalign = 0;
uint32_t avgbytespersec = 0; uint32_t avgbytespersec = 0;
off_t firstblockposn; /* position of the first block in file */ off_t firstblockposn; /* position of the first block in file */
int shortorlong = 1; /* do we output shorts (1) or longs (2)? */
int32_t * const int32_samples = (int32_t*)int16_samples;
/* Generic codec initialisation */ /* Generic codec initialisation */
rb = api; rb = api;
@ -240,13 +242,13 @@ enum codec_status codec_start(struct codec_api* api)
ci->memset(iedata, 0, iend - iedata); ci->memset(iedata, 0, iend - iedata);
#endif #endif
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512)); ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256)); ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
ci->configure(DSP_DITHER, (bool *)false); ci->configure(DSP_DITHER, (bool *)false);
next_track: next_track:
if (codec_init(api)) { if (codec_init(api)) {
i = CODEC_ERROR; i = CODEC_ERROR;
goto exit; goto exit;
@ -310,8 +312,7 @@ enum codec_status codec_start(struct codec_api* api)
goto exit; goto exit;
} }
samplesperblock = buf[26]|(buf[27]<<8); samplesperblock = buf[26]|(buf[27]<<8);
} } else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
if (size < 22) { if (size < 22) {
DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is " DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
"missing extension\n"); "missing extension\n");
@ -325,18 +326,14 @@ enum codec_status codec_start(struct codec_api* api)
formattag = buf[32]|(buf[33]<<8); formattag = buf[32]|(buf[33]<<8);
} }
} }
} } else if (memcmp(buf, "data", 4) == 0) {
else if (memcmp(buf,"data",4)==0) {
numbytes = i; numbytes = i;
i = 0; /* advance to the beginning of data */ i = 0; /* advance to the beginning of data */
} } else if (memcmp(buf, "fact", 4) == 0) {
else if (memcmp(buf,"fact",4)==0) {
/* dwSampleLength */ /* dwSampleLength */
if (i>=4) { if (i >= 4)
totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24)); totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
} } else {
}
else {
DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n", DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
buf[0], buf[1], buf[2], buf[3], i); buf[0], buf[1], buf[2], buf[3], i);
} }
@ -388,23 +385,11 @@ enum codec_status codec_start(struct codec_api* api)
goto exit; goto exit;
} }
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency)); ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
if (bitspersample <= 16) {
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
} else {
shortorlong = 2;
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *) (32));
ci->configure(DSP_SET_CLIP_MAX, (long *) (2147483647));
ci->configure(DSP_SET_CLIP_MIN, (long *) (-2147483647-1));
}
if (channels == 2) { if (channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED); ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
} else if (channels == 1) { } else if (channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO); ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_MONO);
} else { } else {
DEBUGF("CODEC_ERROR: more than 2 channels\n"); DEBUGF("CODEC_ERROR: more than 2 channels\n");
i = CODEC_ERROR; i = CODEC_ERROR;
@ -418,19 +403,17 @@ enum codec_status codec_start(struct codec_api* api)
/* for PCM and derived formats only */ /* for PCM and derived formats only */
bytespersample = (((bitspersample - 1)/8 + 1)*channels); bytespersample = (((bitspersample - 1)/8 + 1)*channels);
totalsamples = numbytes/bytespersample; totalsamples = numbytes/bytespersample;
} } else {
else {
DEBUGF("CODEC_ERROR: cannot compute totalsamples\n"); DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
i = CODEC_ERROR; i = CODEC_ERROR;
goto exit; goto exit;
} }
} }
firstblockposn = (1024-n); firstblockposn = 1024 - n;
ci->advance_buffer(firstblockposn); ci->advance_buffer(firstblockposn);
/* The main decoder loop */ /* The main decoder loop */
bytesdone = 0; bytesdone = 0;
ci->set_elapsed(0); ci->set_elapsed(0);
endofstream = 0; endofstream = 0;
@ -440,16 +423,14 @@ enum codec_status codec_start(struct codec_api* api)
chunksize = (1 + avgbytespersec / (50*blockalign))*blockalign; chunksize = (1 + avgbytespersec / (50*blockalign))*blockalign;
/* check that the output buffer is big enough (convert to samplespersec, /* check that the output buffer is big enough (convert to samplespersec,
then round to the blockalign multiple below) */ then round to the blockalign multiple below) */
if (((uint64_t)chunksize*ci->id3->frequency*channels*shortorlong) if (((uint64_t)chunksize*ci->id3->frequency*channels*sizeof(long))
/(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) { /(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec
/ ((uint64_t)ci->id3->frequency * channels * shortorlong /((uint64_t)ci->id3->frequency*channels*sizeof(long)
*blockalign))*blockalign; *blockalign))*blockalign;
} }
while (!endofstream) { while (!endofstream) {
uint8_t *wavbuf8;
ci->yield(); ci->yield();
if (ci->stop_codec || ci->reload_codec) { if (ci->stop_codec || ci->reload_codec) {
break; break;
@ -464,13 +445,11 @@ enum codec_status codec_start(struct codec_api* api)
/ (1000LL*blockalign))*blockalign; / (1000LL*blockalign))*blockalign;
if (newpos > numbytes) if (newpos > numbytes)
break; break;
if (ci->seek_buffer(firstblockposn + newpos)) { if (ci->seek_buffer(firstblockposn + newpos))
bytesdone = newpos; bytesdone = newpos;
}
ci->seek_complete(); ci->seek_complete();
} }
wavbuf=ci->request_buffer((long *)&n,chunksize); wavbuf = (uint8_t *)ci->request_buffer((long *)&n, chunksize);
wavbuf8 = (uint8_t*)wavbuf;
if (n == 0) if (n == 0)
break; /* End of stream */ break; /* End of stream */
@ -480,48 +459,41 @@ enum codec_status codec_start(struct codec_api* api)
endofstream = 1; endofstream = 1;
} }
wavbufsize = sizeof(int16_samples);
if (formattag == WAVE_FORMAT_PCM) { if (formattag == WAVE_FORMAT_PCM) {
if (bitspersample > 24) { if (bitspersample > 24) {
for (i = 0; i < n; i += 4) { for (i = 0; i < n; i += 4) {
int32_samples[i/4]=(int32_t)(wavbuf8[i]|(wavbuf8[i+1]<<8)| samples[i/4] = (wavbuf[i] >> 3)|
(wavbuf8[i+2]<<16)|(wavbuf8[i+3]<<24)); (wavbuf[i + 1]<<5)|(wavbuf[i + 2]<<13)|
(SE(wavbuf[i + 3])<<21);
} }
wavbufsize = n; bufsize = n;
} } else if (bitspersample > 16) {
else if (bitspersample > 16) {
for (i = 0; i < n; i += 3) { for (i = 0; i < n; i += 3) {
int32_samples[i/3]=(int32_t)((wavbuf8[i]<<8)| samples[i/3] = (wavbuf[i]<<5)|
(wavbuf8[i+1]<<16)|(wavbuf8[i+2]<<24)); (wavbuf[i + 1]<<13)|(SE(wavbuf[i + 2])<<21);
} }
wavbufsize = n*4/3; bufsize = n*4/3;
} else if (bitspersample > 8) {
for (i = 0; i < n; i += 2) {
samples[i/2] = (wavbuf[i]<<13)|(SE(wavbuf[i + 1])<<21);
} }
else if (bitspersample > 8) { bufsize = n*2;
/* Byte-swap data. */ } else {
for (i=0;i<n/2;i++) {
int16_samples[i]=(int16_t)letoh16(wavbuf[i]);
}
wavbufsize = n;
}
else {
for (i = 0; i < n; i++) { for (i = 0; i < n; i++) {
int16_samples[i] = (wavbuf8[i]<<8) - 0x8000; samples[i] = (wavbuf[i] - 0x80)<<21;
} }
wavbufsize = n*2; bufsize = n*4;
} }
} } else if (formattag == WAVE_FORMAT_ALAW
else if (formattag == WAVE_FORMAT_ALAW || formattag == IBM_FORMAT_ALAW) { || formattag == IBM_FORMAT_ALAW) {
for (i=0;i<n;i++) { for (i = 0; i < n; i++)
int16_samples[i] = alaw2linear16[wavbuf8[i]]; samples[i] = alaw2linear16[wavbuf[i]] << 13;
} bufsize = n*4;
wavbufsize = n*2; } else if (formattag == WAVE_FORMAT_MULAW
} || formattag == IBM_FORMAT_MULAW) {
else if (formattag == WAVE_FORMAT_MULAW || formattag == IBM_FORMAT_MULAW) { for (i = 0; i < n; i++)
for (i=0;i<n;i++) { samples[i] = ulaw2linear16[wavbuf[i]] << 13;
int16_samples[i] = ulaw2linear16[wavbuf8[i]]; bufsize = n*4;
}
wavbufsize = n*2;
} }
else if (formattag == WAVE_FORMAT_DVI_ADPCM) { else if (formattag == WAVE_FORMAT_DVI_ADPCM) {
unsigned int nblocks = chunksize/blockalign; unsigned int nblocks = chunksize/blockalign;
@ -530,35 +502,25 @@ enum codec_status codec_start(struct codec_api* api)
size_t decodedsize = samplesperblock*channels; size_t decodedsize = samplesperblock*channels;
if (decode_dvi_adpcm(ci, ((uint8_t *)wavbuf) + i*blockalign, if (decode_dvi_adpcm(ci, ((uint8_t *)wavbuf) + i*blockalign,
blockalign, channels, bitspersample, blockalign, channels, bitspersample,
int16_samples+i*samplesperblock*channels, samples + i*samplesperblock*channels,
&decodedsize) &decodedsize) != CODEC_OK)
!= CODEC_OK) {
i = CODEC_ERROR; i = CODEC_ERROR;
goto exit; goto exit;
} }
if (decodedsize != samplesperblock) { bufsize = nblocks*samplesperblock*channels*2;
i = CODEC_ERROR; } else {
goto exit;
}
}
wavbufsize = nblocks*samplesperblock*channels*2;
}
else {
DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag); DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
i = CODEC_ERROR; i = CODEC_ERROR;
goto exit; goto exit;
} }
while (!ci->pcmbuf_insert((char*)int16_samples, wavbufsize)) { while (!ci->pcmbuf_insert((char *)samples, bufsize))
ci->yield(); ci->yield();
}
ci->advance_buffer(n); ci->advance_buffer(n);
bytesdone += n; bytesdone += n;
if (bytesdone >= numbytes) { if (bytesdone >= numbytes)
endofstream = 1; endofstream = 1;
}
ci->set_elapsed(bytesdone*1000LL/avgbytespersec); ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
} }
@ -575,7 +537,7 @@ decode_dvi_adpcm(struct codec_api* ci,
const uint8_t *buf, const uint8_t *buf,
int n, int n,
uint16_t channels, uint16_t bitspersample, uint16_t channels, uint16_t bitspersample,
int16_t *pcmout, int32_t *pcmout,
size_t *pcmoutsize) size_t *pcmoutsize)
{ {
size_t nsamples = 0; size_t nsamples = 0;
@ -625,7 +587,6 @@ decode_dvi_adpcm(struct codec_api* ci,
buf += 4; buf += 4;
n -= 4; n -= 4;
} }
for (i = 0; i < 8; i++) { for (i = 0; i < 8; i++) {
for (c = 0; c < channels; c++) { for (c = 0; c < channels; c++) {
step = dvi_adpcm_steptab[stepindex[c]]; step = dvi_adpcm_steptab[stepindex[c]];
@ -661,20 +622,18 @@ decode_dvi_adpcm(struct codec_api* ci,
sample[c] -= diff; sample[c] -= diff;
if (sample[c] < -32768) if (sample[c] < -32768)
sample[c] = -32768; sample[c] = -32768;
} } else {
else {
sample[c] += diff; sample[c] += diff;
if (sample[c] > 32767) if (sample[c] > 32767)
sample[c] = 32767; sample[c] = 32767;
} }
/* output the new sample */ /* output the new sample */
pcmout[nsamples] = sample[c]; pcmout[nsamples] = sample[c] << 13;
nsamples++; nsamples++;
} }
} }
} }
} } else { /* bitspersample == 3 */
else { /* bitspersample == 3 */
while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) { while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) {
for (c = 0; c < channels; c++) { for (c = 0; c < channels; c++) {
uint16_t bitstream = 0; uint16_t bitstream = 0;
@ -737,7 +696,7 @@ decode_dvi_adpcm(struct codec_api* ci,
sample[c] = 32767; sample[c] = 32767;
} }
/* output the new sample */ /* output the new sample */
pcmout[nsamples] = sample[c]; pcmout[nsamples] = sample[c] << 13;
nsamples++; nsamples++;
} }
} }