diff options
author | Yoshihisa Uchida <uchida@rockbox.org> | 2010-02-24 11:46:29 +0000 |
---|---|---|
committer | Yoshihisa Uchida <uchida@rockbox.org> | 2010-02-24 11:46:29 +0000 |
commit | 45e009a364d3ee105bf6b2ebb2e28445c9c6e3fd (patch) | |
tree | 7c0cff93773ebb8d88a4fc0fecd126fcbee46124 /apps/codecs | |
parent | aa58715a54cb62638b263b7942c46e25ec928534 (diff) |
add SMAF codec (.mmf extension)(FS#10432)
This codec supports only wave data (ADPCM and PCM).
It does not support MIDI, picture, and movie.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24878 a1c6a512-1295-4272-9138-f99709370657
Diffstat (limited to 'apps/codecs')
-rw-r--r-- | apps/codecs/SOURCES | 1 | ||||
-rw-r--r-- | apps/codecs/codecs.make | 1 | ||||
-rw-r--r-- | apps/codecs/smaf.c | 435 |
3 files changed, 437 insertions, 0 deletions
diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES index 4c847c23e0..9787caa122 100644 --- a/apps/codecs/SOURCES +++ b/apps/codecs/SOURCES @@ -27,6 +27,7 @@ shorten.c aiff.c speex.c adx.c +smaf.c #if defined(HAVE_RECORDING) && !defined(SIMULATOR) /* encoders */ aiff_enc.c diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make index 633f35b273..6a86517119 100644 --- a/apps/codecs/codecs.make +++ b/apps/codecs/codecs.make @@ -90,6 +90,7 @@ $(CODECDIR)/atrac3_rm.codec : $(CODECDIR)/libatrac.a $(CODECDIR)/librm.a $(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a $(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a $(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a +$(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a $(CODECS): $(CODECLIB) # this must be last in codec dependency list diff --git a/apps/codecs/smaf.c b/apps/codecs/smaf.c new file mode 100644 index 0000000000..227227ada8 --- /dev/null +++ b/apps/codecs/smaf.c @@ -0,0 +1,435 @@ +/*************************************************************************** + * __________ __ ___. + * Open \______ \ ____ ____ | | _\_ |__ _______ ___ + * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / + * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < + * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ + * \/ \/ \/ \/ \/ + * $Id$ + * + * Copyright (c) 2010 Yoshihisa Uchida + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY + * KIND, either express or implied. + * + ****************************************************************************/ + +#include "codeclib.h" +#include "codecs/libpcm/support_formats.h" + +CODEC_HEADER + +/* + * SMAF (Synthetic music Mobile Application Format) + * + * References + * [1] YAMAHA Corporation, Synthetic music Mobile Application Format Ver.3.05, 2002 + */ + +enum { + SMAF_TRACK_CHUNK_SCORE = 0, /* Score Track */ + SMAF_TRACK_CHUNK_AUDIO, /* PCM Audio Track */ +}; + +/* SMAF supported codec formats */ +enum { + SMAF_FORMAT_UNSUPPORT = 0, /* unsupported format */ + SMAF_FORMAT_SIGNED_PCM, /* 2's complement PCM */ + SMAF_FORMAT_UNSIGNED_PCM, /* Offset Binary PCM */ + SMAF_FORMAT_ADPCM, /* YAMAHA ADPCM */ +}; + +static int support_formats[2][3] = { + {SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM }, + {SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT }, +}; + +static const struct pcm_entry pcm_codecs[] = { + { SMAF_FORMAT_SIGNED_PCM, get_linear_pcm_codec }, + { SMAF_FORMAT_UNSIGNED_PCM, get_linear_pcm_codec }, + { SMAF_FORMAT_ADPCM, get_yamaha_adpcm_codec }, +}; + +#define NUM_FORMATS 3 + +static int basebits[4] = { 4, 8, 12, 16 }; + +#define PCM_SAMPLE_SIZE (2048*2) + +static int32_t samples[PCM_SAMPLE_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; +} + +static unsigned int get_be32(uint8_t *buf) +{ + return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3]; +} + +static int convert_smaf_audio_format(int track_chunk, unsigned int audio_format) +{ + if (audio_format > 3) + return SMAF_FORMAT_UNSUPPORT; + + return support_formats[track_chunk][audio_format]; +} + +static int convert_smaf_audio_basebit(unsigned int basebit) +{ + if (basebit > 4) + return 0; + return basebits[basebit]; +} + +static bool parse_audio_track(struct pcm_format *fmt, + unsigned char **stbuf, unsigned char *endbuf) +{ + unsigned char *buf = *stbuf; + int chunksize; + + buf += 8; + fmt->channels = ((buf[2] & 0x80) >> 7) + 1; + fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_AUDIO, + (buf[2] >> 4) & 0x07); + if (fmt->formattag == SMAF_FORMAT_UNSUPPORT) + { + DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n", (buf[2] >> 4) & 0x07); + return false; + } + fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4); + if (fmt->bitspersample == 0) + { + DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n", buf[3] >> 4); + return false; + } + buf += 6; + while (buf < endbuf) + { + chunksize = get_be32(buf + 4) + 8; + if (memcmp(buf, "Awa", 3) == 0) + { + fmt->numbytes = get_be32(buf + 4); + buf += 8; + return true; + } + buf += chunksize; + } + DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n"); + return false; +} + +static bool parse_score_track(struct pcm_format *fmt, + unsigned char **stbuf, unsigned char *endbuf) +{ + unsigned char *buf = *stbuf; + int chunksize; + + if (buf[9] != 0x00) + { + DEBUGF("CODEC_ERROR: score track chunk unsupport sequence type %d\n", buf[9]); + return false; + } + + /* + * skip to the next chunk. + * MA-2/MA-3/MA-5: padding 16 bytes + * MA-7: padding 32 bytes + */ + if (buf[3] < 7) + buf += 28; + else + buf += 44; + + while (buf < endbuf) + { + chunksize = get_be32(buf + 4) + 8; + if (memcmp(buf, "Mtsp", 4) == 0) + { + buf += 8; + if (memcmp(buf, "Mwa", 3) != 0) + { + DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n"); + return false; + } + fmt->numbytes = get_be32(buf + 4) - 3; + fmt->channels = ((buf[8] & 0x80) >> 7) + 1; + fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_SCORE, + (buf[8] >> 4) & 0x07); + if (fmt->formattag == SMAF_FORMAT_UNSUPPORT) + { + DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n", + (buf[8] >> 4) & 0x07); + return false; + } + fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0x0f); + if (fmt->bitspersample == 0) + { + DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n", + buf[8] & 0x0f); + return false; + } + buf += 11; + return true; + } + buf += chunksize; + } + + DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n"); + return false; +} + +static bool parse_header(struct pcm_format *fmt, size_t *pos) +{ + unsigned char *buf, *stbuf, *endbuf; + size_t chunksize; + + ci->memset(fmt, 0, sizeof(struct pcm_format)); + + /* assume the SMAF pcm data position is less than 1024 bytes */ + stbuf = ci->request_buffer(&chunksize, 1024); + if (chunksize < 1024) + return false; + + buf = stbuf; + endbuf = stbuf + chunksize; + + if (memcmp(buf, "MMMD", 4) != 0) + { + DEBUGF("CODEC_ERROR: does not smaf format %c%c%c%c\n", + buf[0], buf[1], buf[2], buf[3]); + return false; + } + buf += 8; + + while (buf < endbuf) + { + chunksize = get_be32(buf + 4) + 8; + if (memcmp(buf, "ATR", 3) == 0) + { + if (!parse_audio_track(fmt, &buf, endbuf)) + return false; + break; + } + if (memcmp(buf, "MTR", 3) == 0) + { + if (!parse_score_track(fmt, &buf, endbuf)) + return false; + break; + } + buf += chunksize; + } + + if (buf >= endbuf) + { + DEBUGF("CODEC_ERROR: unsupported smaf format\n"); + return false; + } + + /* blockalign */ + if (fmt->formattag == SMAF_FORMAT_SIGNED_PCM || + fmt->formattag == SMAF_FORMAT_UNSIGNED_PCM) + fmt->blockalign = fmt->channels * fmt->bitspersample >> 3; + + /* data signess (default signed) */ + fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM); + + fmt->is_little_endian = false; + + /* sets pcm data position */ + *pos = buf - stbuf; + + return true; +} + +static struct pcm_format format; +static uint32_t bytesdone; + +static uint8_t *read_buffer(size_t *realsize) +{ + uint8_t *buffer = (uint8_t *)ci->request_buffer(realsize, format.chunksize); + if (bytesdone + (*realsize) > format.numbytes) + *realsize = format.numbytes - bytesdone; + bytesdone += *realsize; + ci->advance_buffer(*realsize); + return buffer; +} + +enum codec_status codec_main(void) +{ + int status = CODEC_OK; + uint32_t decodedsamples; + uint32_t i = CODEC_OK; + size_t n; + int bufcount; + int endofstream; + uint8_t *smafbuf; + 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; + goto exit; + } + + while (!*ci->taginfo_ready && !ci->stop_codec) + ci->sleep(1); + + codec_set_replaygain(ci->id3); + + ci->memset(&format, 0, sizeof(struct pcm_format)); + format.is_signed = true; + format.is_little_endian = false; + + decodedsamples = 0; + codec = 0; + + if (!parse_header(&format, &n)) + { + i = CODEC_ERROR; + goto done; + } + + codec = get_codec(format.formattag); + if (codec == 0) + { + DEBUGF("CODEC_ERROR: unsupport audio format: 0x%lx\n", format.formattag); + i = CODEC_ERROR; + goto done; + } + + if (!codec->set_format(&format)) + { + i = CODEC_ERROR; + goto done; + } + + /* common format check */ + if (format.channels == 0) { + DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n"); + status = CODEC_ERROR; + goto done; + } + if (format.samplesperblock == 0) { + DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n"); + status = CODEC_ERROR; + goto done; + } + if (format.blockalign == 0) + { + DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n"); + i = CODEC_ERROR; + goto done; + } + if (format.numbytes == 0) { + DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n"); + status = CODEC_ERROR; + goto done; + } + + /* check chunksize */ + if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels + > PCM_SAMPLE_SIZE) + format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign; + if (format.chunksize == 0) + { + DEBUGF("CODEC_ERROR: chunksize is 0\n"); + i = CODEC_ERROR; + goto done; + } + + ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); + + if (format.channels == 2) { + ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED); + } else if (format.channels == 1) { + ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO); + } else { + DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n"); + i = CODEC_ERROR; + goto done; + } + + firstblockposn = 1024 - n; + ci->advance_buffer(firstblockposn); + + /* The main decoder loop */ + bytesdone = 0; + ci->set_elapsed(0); + endofstream = 0; + + while (!endofstream) { + ci->yield(); + if (ci->stop_codec || ci->new_track) + break; + + if (ci->seek_time) { + struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, &read_buffer); + + decodedsamples = newpos->samples; + if (newpos->pos > format.numbytes) + break; + if (ci->seek_buffer(firstblockposn + newpos->pos)) + { + bytesdone = newpos->pos; + } + ci->seek_complete(); + } + smafbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize); + + if (n == 0) + break; /* End of stream */ + + if (bytesdone + n > format.numbytes) { + n = format.numbytes - bytesdone; + endofstream = 1; + } + + status = codec->decode(smafbuf, n, samples, &bufcount); + if (status == CODEC_ERROR) + { + DEBUGF("codec error\n"); + goto done; + } + + ci->pcmbuf_insert(samples, NULL, bufcount); + + ci->advance_buffer(n); + bytesdone += n; + decodedsamples += bufcount; + if (bytesdone >= format.numbytes) + endofstream = 1; + + ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency); + } + i = CODEC_OK; + +done: + if (ci->request_next_track()) + goto next_track; + +exit: + return i; +} + |