Add Sun Audio codec. (FS#10433)

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24955 a1c6a512-1295-4272-9138-f99709370657

3 files changed, 321 insertions, 0 deletions

diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES
index 9787caa..b8a86ae 100644
--- a/apps/codecs/SOURCES
+++ b/apps/codecs/SOURCES
@@ -28,6 +28,7 @@ aiff.c
 speex.c
 adx.c
 smaf.c
+au.c
 #if defined(HAVE_RECORDING) && !defined(SIMULATOR)
 /* encoders */
 aiff_enc.c
diff --git a/apps/codecs/au.c b/apps/codecs/au.c
new file mode 100644
index 0000000..5632bf7
--- /dev/null
+++ b/apps/codecs/au.c
@@ -0,0 +1,319 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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
+
+/* Sun Audio file (Au file format) codec
+ *
+ * References
+ * [1] Sun Microsystems, Inc., Header file for Audio, .au, 1992
+ *     URL http://www.opengroup.org/public/pubs/external/auformat.html
+ * [2] Wikipedia, Au file format, URL: http://en.wikipedia.org/wiki/Sun_Audio
+ */
+
+#define PCM_SAMPLE_SIZE (1024*2)
+
+static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
+
+enum
+{
+    AU_FORMAT_UNSUPPORT = 0, /* unsupported format */
+    AU_FORMAT_MULAW,         /* G.711 MULAW */
+    AU_FORMAT_PCM,           /* Linear PCM */
+    AU_FORMAT_IEEE_FLOAT,    /* IEEE float */
+    AU_FORMAT_ALAW,          /* G.711 ALAW */
+};
+
+static int support_formats[28][2] = {
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_MULAW,      8  }, /* G.711 MULAW */
+  { AU_FORMAT_PCM,        8  }, /* Linear PCM 8bit (signed) */
+  { AU_FORMAT_PCM,        16 }, /* Linear PCM 16bit (signed, big endian) */
+  { AU_FORMAT_PCM,        24 }, /* Linear PCM 24bit (signed, big endian) */
+  { AU_FORMAT_PCM,        32 }, /* Linear PCM 32bit (signed, big endian) */
+  { AU_FORMAT_IEEE_FLOAT, 32 }, /* Linear PCM float 32bit (signed, big endian) */
+  { AU_FORMAT_IEEE_FLOAT, 64 }, /* Linear PCM float 64bit (signed, big endian) */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* Fragmented sample data */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* DSP program */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 8bit fixed point */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit fixed point */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 24bit fixed point */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 32bit fixed point */
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear with emphasis */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear compressed */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* 16bit linear with emphasis and compression */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* Music kit DSP commands */
+  { AU_FORMAT_UNSUPPORT,  0  },
+  { AU_FORMAT_UNSUPPORT,  0  }, /* G.721 MULAW */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* G.722 */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* G.723 3bit */
+  { AU_FORMAT_UNSUPPORT,  0  }, /* G.723 5bit */
+  { AU_FORMAT_ALAW,       8  }, /* G.711 ALAW */
+};
+
+const struct pcm_entry au_codecs[] = {
+    { AU_FORMAT_MULAW,      get_itut_g711_mulaw_codec },
+    { AU_FORMAT_PCM,        get_linear_pcm_codec      },
+    { AU_FORMAT_IEEE_FLOAT, get_ieee_float_codec      },
+    { AU_FORMAT_ALAW,       get_itut_g711_alaw_codec  },
+};
+
+#define NUM_FORMATS 4
+
+static const struct pcm_codec *get_au_codec(uint32_t formattag)
+{
+    int i;
+
+    for (i = 0; i < NUM_FORMATS; i++)
+    {
+        if (au_codecs[i].format_tag == formattag)
+        {
+            if (au_codecs[i].get_codec)
+                return au_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_au_format(unsigned int encoding, struct pcm_format *fmt)
+{
+    if (encoding > 27)
+    {
+        fmt->formattag = AU_FORMAT_UNSUPPORT;
+        fmt->bitspersample = 0;
+    }
+    else
+    {
+        fmt->formattag = support_formats[encoding][0];
+        fmt->bitspersample = support_formats[encoding][1];
+    }
+
+    return fmt->formattag;
+}
+
+/* this is the codec entry point */
+enum codec_status codec_main(void)
+{
+    int status = CODEC_OK;
+    struct pcm_format format;
+    uint32_t bytesdone, decodedsamples;
+    size_t n;
+    int bufcount;
+    int endofstream;
+    unsigned char *buf;
+    uint8_t *aubuf;
+    off_t firstblockposn;     /* position of the first block in file */
+    const struct pcm_codec *codec;
+    int offset = 0;
+
+    /* Generic codec initialisation */
+    ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
+  
+next_track:
+    if (codec_init()) {
+        DEBUGF("codec_init() error\n");
+        status = 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;
+
+    /* set format */
+    buf = ci->request_buffer(&n, 24);
+    if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
+    {
+        /*
+         * headerless sun audio file
+         * It is decoded under conditions.
+         *     format:    G.711 mu-law
+         *     channel:   mono
+         *     frequency: 8000 kHz
+         */
+        offset = 0;
+        format.formattag     = AU_FORMAT_MULAW;
+        format.channels      = 1;
+        format.bitspersample = 8;
+        format.numbytes      = ci->id3->filesize;
+    }
+    else
+    {
+        /* parse header */
+
+        /* data offset */
+        offset = get_be32(buf + 4);
+        if (offset < 24)
+        {
+            DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
+            status = CODEC_ERROR;
+            goto done;
+        }
+        /* data size */
+        format.numbytes = get_be32(buf + 8);
+        if (format.numbytes == (uint32_t)0xffffffff)
+            format.numbytes = ci->id3->filesize - offset;
+        /* encoding */
+        format.formattag = convert_au_format(get_be32(buf + 12), &format);
+        if (format.formattag == AU_FORMAT_UNSUPPORT)
+        {
+            DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
+            status = CODEC_ERROR;
+            goto done;
+        }
+        /* skip sample rate */
+        format.channels = get_be32(buf + 20);
+        if (format.channels == 0) {
+            DEBUGF("CODEC_ERROR: sun audio 0-channels file\n");
+            status = CODEC_ERROR;
+            goto done;
+        }
+    }
+
+    /* advance to first WAVE chunk */
+    ci->advance_buffer(offset);
+
+    firstblockposn = offset;
+
+    decodedsamples = 0;
+    codec = 0;
+    bytesdone = 0;
+
+    /* blockalign = 1 sample */
+    format.blockalign = format.bitspersample * format.channels >> 3;
+
+    /* get codec */
+    codec = get_au_codec(format.formattag);
+    if (!codec)
+    {
+        DEBUGF("CODEC_ERROR: unsupport sun audio format: %lx\n", format.formattag);
+        status = CODEC_ERROR;
+        goto done;
+    }
+
+    if (!codec->set_format(&format))
+    {
+        status = CODEC_ERROR;
+        goto done;
+    }
+
+    if (format.numbytes == 0) {
+        DEBUGF("CODEC_ERROR: data size is 0\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");
+        status = 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\n");
+        status = CODEC_ERROR;
+        goto done;
+    }
+
+    /* The main decoder loop */
+    endofstream = 0;
+
+    while (!endofstream) {
+        ci->yield();
+        if (ci->stop_codec || ci->new_track) {
+            break;
+        }
+
+        if (ci->seek_time) {
+            /* 2nd args(read_buffer) is unnecessary in the format which Sun Audio supports.  */
+            struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, NULL);
+
+            decodedsamples = newpos->samples;
+            if (newpos->pos > format.numbytes)
+                break;
+            if (ci->seek_buffer(firstblockposn + newpos->pos))
+            {
+                bytesdone = newpos->pos;
+            }
+            ci->seek_complete();
+        }
+
+        aubuf = (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(aubuf, 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);
+    }
+    status = CODEC_OK;
+
+done:
+    if (ci->request_next_track())
+        goto next_track;
+
+exit:
+    return status;
+}
diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make
index 6a86517..cf1a057 100644
--- a/apps/codecs/codecs.make
+++ b/apps/codecs/codecs.make
@@ -91,6 +91,7 @@ $(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a
 $(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a
 $(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
 $(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a
+$(CODECDIR)/au.codec : $(CODECDIR)/libpcm.a
 
 $(CODECS): $(CODECLIB) # this must be last in codec dependency list