forked from len0rd/rockbox
Code policed. Converted to generate internal s3.28 format directly (next
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
parent
d8eaefe29a
commit
0968536c4b
1 changed files with 333 additions and 374 deletions
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@ -22,12 +22,15 @@
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CODEC_HEADER
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/* Macro that sign extends an unsigned byte */
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#define SE(x) ((int32_t)((int8_t)(x)))
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struct codec_api *rb;
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/* This codec support WAVE files with the following formats:
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* - PCM, up to 32 bits, supporting 32 bits playback when useful.
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* - ALAW and MULAW (16 bits compressed on 8 bits).
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* - DVI_ADPCM (16 bits compressed on 4 bits).
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* - DVI_ADPCM (16 bits compressed on 3 or 4 bits).
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*
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* For a good documentation on WAVE files, see:
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* http://www.tsp.ece.mcgill.ca/MMSP/Documents/AudioFormats/WAVE/WAVE.html
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@ -100,8 +103,6 @@ extern char iedata[];
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extern char iend[];
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#endif
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/* Those are lookup tables, so they should be in the idata section
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* (fast but small RAM on the coldfire processor) */
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static const int16_t alaw2linear16[256] ICONST_ATTR = {
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-5504, -5248, -6016, -5760, -4480, -4224, -4992,
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-4736, -7552, -7296, -8064, -7808, -6528, -6272,
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@ -195,17 +196,20 @@ static const uint16_t dvi_adpcm_steptab[ 89 ] ICONST_ATTR = {
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7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
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15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
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32767 };
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static const int dvi_adpcm_indextab4[ 8 ] ICONST_ATTR = { -1, -1, -1, -1, 2, 4, 6, 8 };
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static const int dvi_adpcm_indextab4[8] ICONST_ATTR = {
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-1, -1, -1, -1, 2, 4, 6, 8 };
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static const int dvi_adpcm_indextab3[4] ICONST_ATTR = { -1, -1, 1, 2 };
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static int16_t int16_samples[WAV_CHUNK_SIZE] IBSS_ATTR;
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static int32_t samples[WAV_CHUNK_SIZE] IBSS_ATTR;
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static enum codec_status
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decode_dvi_adpcm(struct codec_api *ci,
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const uint8_t *buf,
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int n,
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uint16_t channels, uint16_t bitspersample,
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int16_t *pcmout,
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int32_t *pcmout,
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size_t *pcmoutsize);
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/* this is the codec entry point */
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@ -219,17 +223,15 @@ enum codec_status codec_start(struct codec_api* api)
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int bytespersample = 0;
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uint16_t bitspersample;
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uint32_t i;
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size_t n, wavbufsize;
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size_t n, bufsize;
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int endofstream;
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unsigned char *buf;
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uint16_t* wavbuf;
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uint8_t *wavbuf;
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long chunksize;
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uint16_t formattag = 0;
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uint16_t blockalign = 0;
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uint32_t avgbytespersec = 0;
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off_t firstblockposn; /* position of the first block in file */
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int shortorlong = 1; /* do we output shorts (1) or longs (2)? */
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int32_t * const int32_samples = (int32_t*)int16_samples;
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/* Generic codec initialisation */
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rb = api;
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@ -240,13 +242,13 @@ enum codec_status codec_start(struct codec_api* api)
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ci->memset(iedata, 0, iend - iedata);
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#endif
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)28);
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ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
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ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
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ci->configure(DSP_DITHER, (bool *)false);
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next_track:
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if (codec_init(api)) {
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i = CODEC_ERROR;
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goto exit;
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@ -310,8 +312,7 @@ enum codec_status codec_start(struct codec_api* api)
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goto exit;
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}
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samplesperblock = buf[26]|(buf[27]<<8);
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}
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else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
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} else if (formattag == WAVE_FORMAT_EXTENSIBLE) {
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if (size < 22) {
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DEBUGF("CODEC_ERROR: WAVE_FORMAT_EXTENSIBLE is "
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"missing extension\n");
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@ -325,18 +326,14 @@ enum codec_status codec_start(struct codec_api* api)
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formattag = buf[32]|(buf[33]<<8);
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}
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}
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}
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else if (memcmp(buf,"data",4)==0) {
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} else if (memcmp(buf, "data", 4) == 0) {
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numbytes = i;
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i = 0; /* advance to the beginning of data */
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}
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else if (memcmp(buf,"fact",4)==0) {
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} else if (memcmp(buf, "fact", 4) == 0) {
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/* dwSampleLength */
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if (i>=4) {
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if (i >= 4)
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totalsamples = (buf[8]|(buf[9]<<8)|(buf[10]<<16)|(buf[11]<<24));
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}
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}
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else {
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} else {
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DEBUGF("unknown WAVE chunk: '%c%c%c%c', size=%lu\n",
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buf[0], buf[1], buf[2], buf[3], i);
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}
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goto exit;
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}
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ci->configure(CODEC_DSP_ENABLE, (bool *)true);
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ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
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if (bitspersample <= 16) {
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ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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} else {
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shortorlong = 2;
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ci->configure(DSP_DITHER, (bool *)false);
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ci->configure(DSP_SET_SAMPLE_DEPTH, (long *) (32));
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ci->configure(DSP_SET_CLIP_MAX, (long *) (2147483647));
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ci->configure(DSP_SET_CLIP_MIN, (long *) (-2147483647-1));
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}
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if (channels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_INTERLEAVED);
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} else if (channels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
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ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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i = CODEC_ERROR;
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@ -418,19 +403,17 @@ enum codec_status codec_start(struct codec_api* api)
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/* for PCM and derived formats only */
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bytespersample = (((bitspersample - 1)/8 + 1)*channels);
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totalsamples = numbytes/bytespersample;
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}
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else {
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} else {
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DEBUGF("CODEC_ERROR: cannot compute totalsamples\n");
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i = CODEC_ERROR;
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goto exit;
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}
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}
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firstblockposn = (1024-n);
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firstblockposn = 1024 - n;
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ci->advance_buffer(firstblockposn);
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/* The main decoder loop */
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bytesdone = 0;
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ci->set_elapsed(0);
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endofstream = 0;
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@ -440,16 +423,14 @@ enum codec_status codec_start(struct codec_api* api)
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chunksize = (1 + avgbytespersec / (50*blockalign))*blockalign;
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/* check that the output buffer is big enough (convert to samplespersec,
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then round to the blockalign multiple below) */
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if (((uint64_t)chunksize*ci->id3->frequency*channels*shortorlong)
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if (((uint64_t)chunksize*ci->id3->frequency*channels*sizeof(long))
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/(uint64_t)avgbytespersec >= WAV_CHUNK_SIZE) {
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chunksize = ((uint64_t)WAV_CHUNK_SIZE*avgbytespersec
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/ ((uint64_t)ci->id3->frequency * channels * shortorlong
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/((uint64_t)ci->id3->frequency*channels*sizeof(long)
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*blockalign))*blockalign;
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}
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while (!endofstream) {
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uint8_t *wavbuf8;
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ci->yield();
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if (ci->stop_codec || ci->reload_codec) {
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break;
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@ -464,13 +445,11 @@ enum codec_status codec_start(struct codec_api* api)
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/ (1000LL*blockalign))*blockalign;
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if (newpos > numbytes)
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break;
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if (ci->seek_buffer(firstblockposn + newpos)) {
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if (ci->seek_buffer(firstblockposn + newpos))
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bytesdone = newpos;
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}
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ci->seek_complete();
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}
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wavbuf=ci->request_buffer((long *)&n,chunksize);
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wavbuf8 = (uint8_t*)wavbuf;
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wavbuf = (uint8_t *)ci->request_buffer((long *)&n, chunksize);
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if (n == 0)
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break; /* End of stream */
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endofstream = 1;
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}
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wavbufsize = sizeof(int16_samples);
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if (formattag == WAVE_FORMAT_PCM) {
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if (bitspersample > 24) {
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for (i = 0; i < n; i += 4) {
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int32_samples[i/4]=(int32_t)(wavbuf8[i]|(wavbuf8[i+1]<<8)|
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(wavbuf8[i+2]<<16)|(wavbuf8[i+3]<<24));
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samples[i/4] = (wavbuf[i] >> 3)|
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(wavbuf[i + 1]<<5)|(wavbuf[i + 2]<<13)|
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(SE(wavbuf[i + 3])<<21);
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}
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wavbufsize = n;
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}
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else if (bitspersample > 16) {
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bufsize = n;
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} else if (bitspersample > 16) {
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for (i = 0; i < n; i += 3) {
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int32_samples[i/3]=(int32_t)((wavbuf8[i]<<8)|
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(wavbuf8[i+1]<<16)|(wavbuf8[i+2]<<24));
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samples[i/3] = (wavbuf[i]<<5)|
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(wavbuf[i + 1]<<13)|(SE(wavbuf[i + 2])<<21);
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}
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wavbufsize = n*4/3;
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bufsize = n*4/3;
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} else if (bitspersample > 8) {
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for (i = 0; i < n; i += 2) {
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samples[i/2] = (wavbuf[i]<<13)|(SE(wavbuf[i + 1])<<21);
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}
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else if (bitspersample > 8) {
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/* Byte-swap data. */
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for (i=0;i<n/2;i++) {
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int16_samples[i]=(int16_t)letoh16(wavbuf[i]);
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}
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wavbufsize = n;
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}
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else {
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bufsize = n*2;
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} else {
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for (i = 0; i < n; i++) {
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int16_samples[i] = (wavbuf8[i]<<8) - 0x8000;
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samples[i] = (wavbuf[i] - 0x80)<<21;
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}
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wavbufsize = n*2;
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bufsize = n*4;
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}
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}
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else if (formattag == WAVE_FORMAT_ALAW || formattag == IBM_FORMAT_ALAW) {
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for (i=0;i<n;i++) {
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int16_samples[i] = alaw2linear16[wavbuf8[i]];
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}
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wavbufsize = n*2;
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}
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else if (formattag == WAVE_FORMAT_MULAW || formattag == IBM_FORMAT_MULAW) {
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for (i=0;i<n;i++) {
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int16_samples[i] = ulaw2linear16[wavbuf8[i]];
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}
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wavbufsize = n*2;
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} else if (formattag == WAVE_FORMAT_ALAW
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|| formattag == IBM_FORMAT_ALAW) {
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for (i = 0; i < n; i++)
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samples[i] = alaw2linear16[wavbuf[i]] << 13;
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bufsize = n*4;
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} else if (formattag == WAVE_FORMAT_MULAW
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|| formattag == IBM_FORMAT_MULAW) {
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for (i = 0; i < n; i++)
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samples[i] = ulaw2linear16[wavbuf[i]] << 13;
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bufsize = n*4;
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}
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else if (formattag == WAVE_FORMAT_DVI_ADPCM) {
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unsigned int nblocks = chunksize/blockalign;
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@ -530,35 +502,25 @@ enum codec_status codec_start(struct codec_api* api)
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size_t decodedsize = samplesperblock*channels;
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if (decode_dvi_adpcm(ci, ((uint8_t *)wavbuf) + i*blockalign,
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blockalign, channels, bitspersample,
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int16_samples+i*samplesperblock*channels,
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&decodedsize)
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!= CODEC_OK) {
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samples + i*samplesperblock*channels,
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&decodedsize) != CODEC_OK)
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i = CODEC_ERROR;
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goto exit;
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}
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if (decodedsize != samplesperblock) {
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i = CODEC_ERROR;
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goto exit;
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}
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}
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wavbufsize = nblocks*samplesperblock*channels*2;
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}
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else {
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bufsize = nblocks*samplesperblock*channels*2;
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} else {
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DEBUGF("CODEC_ERROR: unsupported format %x\n", formattag);
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i = CODEC_ERROR;
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goto exit;
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}
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while (!ci->pcmbuf_insert((char*)int16_samples, wavbufsize)) {
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while (!ci->pcmbuf_insert((char *)samples, bufsize))
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ci->yield();
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}
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ci->advance_buffer(n);
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bytesdone += n;
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if (bytesdone >= numbytes) {
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if (bytesdone >= numbytes)
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endofstream = 1;
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}
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ci->set_elapsed(bytesdone*1000LL/avgbytespersec);
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}
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@ -575,7 +537,7 @@ decode_dvi_adpcm(struct codec_api* ci,
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const uint8_t *buf,
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int n,
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uint16_t channels, uint16_t bitspersample,
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int16_t *pcmout,
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int32_t *pcmout,
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size_t *pcmoutsize)
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{
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size_t nsamples = 0;
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@ -625,7 +587,6 @@ decode_dvi_adpcm(struct codec_api* ci,
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buf += 4;
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n -= 4;
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}
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for (i = 0; i < 8; i++) {
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for (c = 0; c < channels; c++) {
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step = dvi_adpcm_steptab[stepindex[c]];
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@ -661,20 +622,18 @@ decode_dvi_adpcm(struct codec_api* ci,
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sample[c] -= diff;
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if (sample[c] < -32768)
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sample[c] = -32768;
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}
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else {
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} else {
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sample[c] += diff;
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if (sample[c] > 32767)
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sample[c] = 32767;
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}
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/* output the new sample */
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pcmout[nsamples] = sample[c];
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pcmout[nsamples] = sample[c] << 13;
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nsamples++;
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}
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}
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}
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}
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else { /* bitspersample == 3 */
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} else { /* bitspersample == 3 */
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while (n >= channels*12 && (nsamples + 32*channels) <= *pcmoutsize) {
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for (c = 0; c < channels; c++) {
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uint16_t bitstream = 0;
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@ -737,7 +696,7 @@ decode_dvi_adpcm(struct codec_api* ci,
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sample[c] = 32767;
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}
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/* output the new sample */
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pcmout[nsamples] = sample[c];
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pcmout[nsamples] = sample[c] << 13;
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nsamples++;
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}
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}
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Reference in a new issue