1 files changed, 492 insertions, 0 deletions

diff --git a/lib/rbcodec/codecs/smaf.c b/lib/rbcodec/codecs/smaf.c
new file mode 100644
index 0000000000..0203d1b4e3
--- /dev/null
+++ b/lib/rbcodec/codecs/smaf.c
@@ -0,0 +1,492 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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_AUDIO_TRACK_CHUNK = 0, /* PCM Audio Track */
+    SMAF_SCORE_TRACK_CHUNK,     /* Score 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 const int support_formats[2][3] = {
+    {SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM,        SMAF_FORMAT_UNSUPPORT },
+    {SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM     },
+};
+
+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 const 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(const uint8_t *buf)
+{
+    return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
+}
+
+static int convert_smaf_channels(unsigned int ch)
+{
+    return (ch >> 7) + 1;
+}
+
+static int convert_smaf_audio_format(unsigned int chunk, unsigned int audio_format)
+{
+    int idx = (audio_format & 0x70) >> 4;
+
+    if (idx < 3)
+        return support_formats[chunk][idx];
+
+    DEBUGF("CODEC_ERROR: unsupport audio format: %d\n", audio_format);
+    return SMAF_FORMAT_UNSUPPORT;
+}
+
+static int convert_smaf_audio_basebit(unsigned int basebit)
+{
+    if (basebit < 4)
+        return basebits[basebit];
+
+    DEBUGF("CODEC_ERROR: illegal basebit: %d\n", basebit);
+    return 0;
+}
+
+static unsigned int search_chunk(const unsigned char *name, int nlen, off_t *pos)
+{
+    const unsigned char *buf;
+    unsigned int chunksize;
+    size_t size;
+
+    while (true)
+    {
+        buf = ci->request_buffer(&size, 8);
+        if (size < 8)
+            break;
+
+        chunksize = get_be32(buf + 4);
+        ci->advance_buffer(8);
+        *pos += 8;
+        if (memcmp(buf, name, nlen) == 0)
+            return chunksize;
+
+        ci->advance_buffer(chunksize);
+        *pos += chunksize;
+    }
+    DEBUGF("CODEC_ERROR: missing '%s' chunk\n", name);
+    return 0;
+}
+
+static bool parse_audio_track(struct pcm_format *fmt, unsigned int chunksize, off_t *pos)
+{
+    const unsigned char *buf;
+    size_t size;
+
+    /* search PCM Audio Track Chunk */
+    ci->advance_buffer(chunksize);
+    *pos += chunksize;
+    if (search_chunk("ATR", 3, pos) == 0)
+    {
+        DEBUGF("CODEC_ERROR: missing PCM Audio Track Chunk\n");
+        return false;
+    }
+
+    /*
+     * get format
+     *  buf
+     *    +0: Format Type
+     *    +1: Sequence Type
+     *    +2: bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: frequency
+     *    +3: bit 4-7: base bit
+     *    +4: TimeBase_D
+     *    +5: TimeBase_G
+     *
+     * Note: If PCM Audio Track does not include Sequence Data Chunk,
+     *       tmp+6 is the start position of Wave Data Chunk.
+     */
+    buf = ci->request_buffer(&size, 6);
+    if (size < 6)
+    {
+        DEBUGF("CODEC_ERROR: smaf is too small\n");
+        return false;
+    }
+
+    fmt->formattag     = convert_smaf_audio_format(SMAF_AUDIO_TRACK_CHUNK, buf[2]);
+    fmt->channels      = convert_smaf_channels(buf[2]);
+    fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
+
+    /* search Wave Data Chunk */
+    ci->advance_buffer(6);
+    *pos += 6;
+    fmt->numbytes = search_chunk("Awa", 3, pos);
+    if (fmt->numbytes == 0)
+    {
+        DEBUGF("CODEC_ERROR: missing Wave Data Chunk\n");
+        return false;
+    }
+
+    return true;
+}
+
+static bool parse_score_track(struct pcm_format *fmt, off_t *pos)
+{
+    const unsigned char *buf;
+    unsigned int chunksize;
+    size_t size;
+
+    /* parse Optional Data Chunk */
+    buf = ci->request_buffer(&size, 13);
+    if (size < 13)
+    {
+        DEBUGF("CODEC_ERROR: smaf is too small\n");
+        return false;
+    }
+
+    if (memcmp(buf + 5, "OPDA", 4) != 0)
+    {
+        DEBUGF("CODEC_ERROR: missing Optional Data Chunk\n");
+        return false;
+    }
+
+    /* Optional Data Chunk size */
+    chunksize = get_be32(buf + 9);
+
+    /* search Score Track Chunk */
+    ci->advance_buffer(13 + chunksize);
+    *pos += (13 + chunksize);
+    if (search_chunk("MTR", 3, pos) == 0)
+    {
+        DEBUGF("CODEC_ERROR: missing Score Track Chunk\n");
+        return false;
+    }
+
+    /*
+     * search next chunk
+     * usually, next chunk ('M***') found within 40 bytes.
+     */
+    buf = ci->request_buffer(&size, 40);
+    if (size < 40)
+    {
+        DEBUGF("CODEC_ERROR: smaf is too small\n");
+        return false;
+    }
+
+    size = 0;
+    while (size < 40 && buf[size] != 'M')
+        size++;
+
+    if (size >= 40)
+    {
+        DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk");
+        return false;
+    }
+
+    /* search Score Track Stream PCM Data Chunk */
+    ci->advance_buffer(size);
+    *pos += size;
+    if (search_chunk("Mtsp", 4, pos) == 0)
+    {
+        DEBUGF("CODEC_ERROR: missing Score Track Stream PCM Data Chunk\n");
+        return false;
+    }
+
+    /*
+     * parse Score Track Stream Wave Data Chunk
+     *  buf
+     *    +4-7: chunk size (WaveType(3bytes) + wave data count)
+     *    +8:   bit 7 0:mono/1:stereo, bit 4-6 format, bit 0-3: base bit
+     *    +9:   frequency (MSB)
+     *    +10:  frequency (LSB)
+     */
+    buf = ci->request_buffer(&size, 9);
+    if (size < 9)
+    {
+        DEBUGF("CODEC_ERROR: smaf is too small\n");
+        return false;
+    }
+
+    if (memcmp(buf, "Mwa", 3) != 0)
+    {
+        DEBUGF("CODEC_ERROR: missing Score Track Stream Wave Data Chunk\n");
+        return false;
+    }
+
+    fmt->formattag     = convert_smaf_audio_format(SMAF_SCORE_TRACK_CHUNK, buf[8]);
+    fmt->channels      = convert_smaf_channels(buf[8]);
+    fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0xf);
+    fmt->numbytes      = get_be32(buf + 4) - 3;
+
+    *pos += 11;
+    return true;
+}
+
+static bool parse_header(struct pcm_format *fmt, off_t *pos)
+{
+    const unsigned char *buf;
+    unsigned int chunksize;
+    size_t size;
+
+    ci->memset(fmt, 0, sizeof(struct pcm_format));
+
+    /* check File Chunk and Contents Info Chunk */
+    buf = ci->request_buffer(&size, 16);
+    if (size < 16)
+    {
+        DEBUGF("CODEC_ERROR: smaf is too small\n");
+        return false;
+    }
+
+    if ((memcmp(buf, "MMMD", 4) != 0) || (memcmp(buf + 8, "CNTI", 4) != 0))
+    {
+        DEBUGF("CODEC_ERROR: does not smaf format\n");
+        return false;
+    }
+
+    chunksize = get_be32(buf + 12);
+    ci->advance_buffer(16);
+    *pos = 16;
+    if (chunksize > 5)
+    {
+        if (!parse_audio_track(fmt, chunksize, pos))
+            return false;
+    }
+    else if (!parse_score_track(fmt, pos))
+        return false;
+
+    /* data signess (default signed) */
+    fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM);
+
+    /* data is always big endian */
+    fmt->is_little_endian = false;
+
+    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;
+}
+
+/* this is the codec entry point */
+enum codec_status codec_main(enum codec_entry_call_reason reason)
+{
+    if (reason == CODEC_LOAD) {
+        /* Generic codec initialisation */
+        ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
+    }
+
+    return CODEC_OK;
+}
+
+/* this is called for each file to process */
+enum codec_status codec_run(void)
+{
+    uint32_t decodedsamples;
+    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;
+    intptr_t param;
+  
+    if (codec_init())
+        return CODEC_ERROR;
+
+    codec_set_replaygain(ci->id3);
+
+    /* Need to save offset for later use (cleared indirectly by advance_buffer) */
+    bytesdone = ci->id3->offset;
+
+    decodedsamples = 0;
+    codec = 0;
+
+    ci->seek_buffer(0);
+    if (!parse_header(&format, &firstblockposn))
+    {
+        return CODEC_ERROR;
+    }
+
+    codec = get_codec(format.formattag);
+    if (codec == 0)
+    {
+        DEBUGF("CODEC_ERROR: unsupport audio format: 0x%x\n", (int)format.formattag);
+        return CODEC_ERROR;
+    }
+
+    if (!codec->set_format(&format))
+    {
+        return CODEC_ERROR;
+    }
+
+    /* 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");
+        return CODEC_ERROR;
+    }
+
+    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");
+        return CODEC_ERROR;
+    }
+
+    ci->seek_buffer(firstblockposn);
+
+    /* make sure we're at the correct offset */
+    if (bytesdone > (uint32_t) firstblockposn)
+    {
+        /* Round down to previous block */
+        struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
+                                                     PCM_SEEK_POS, &read_buffer);
+
+        if (newpos->pos > format.numbytes)
+            return CODEC_OK;
+
+        if (ci->seek_buffer(firstblockposn + newpos->pos))
+        {
+            bytesdone      = newpos->pos;
+            decodedsamples = newpos->samples;
+        }
+    }
+    else
+    {
+        /* already where we need to be */
+        bytesdone = 0;
+    }
+
+    ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
+
+    /* The main decoder loop */
+    endofstream = 0;
+
+    while (!endofstream) {
+        enum codec_command_action action = ci->get_command(&param);
+
+        if (action == CODEC_ACTION_HALT)
+            break;
+
+        if (action == CODEC_ACTION_SEEK_TIME) {
+            struct pcm_pos *newpos = codec->get_seek_pos(param, PCM_SEEK_TIME,
+                                                         &read_buffer);
+
+            if (newpos->pos > format.numbytes)
+            {
+                ci->set_elapsed(ci->id3->length);
+                ci->seek_complete();
+                break;
+            }
+
+            if (ci->seek_buffer(firstblockposn + newpos->pos))
+            {
+                bytesdone      = newpos->pos;
+                decodedsamples = newpos->samples;
+            }
+
+            ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
+            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;
+        }
+
+        if (codec->decode(smafbuf, n, samples, &bufcount) == CODEC_ERROR)
+        {
+            DEBUGF("codec error\n");
+            return CODEC_ERROR;
+        }
+
+        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);
+    }
+
+    return CODEC_OK;
+}