moved and renamed the codecs, gave the codecs a new extension (.codec),

unified to a single codec-only API, made a new codeclib, disabled the building
of the *2wav plugins


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6812 a1c6a512-1295-4272-9138-f99709370657

13 files changed, 2233 insertions, 2 deletions

diff --git a/apps/codecs/Makefile b/apps/codecs/Makefile
index ffd8eab..970048e 100644
--- a/apps/codecs/Makefile
+++ b/apps/codecs/Makefile
@@ -10,20 +10,97 @@
 INCLUDES = -I$(FIRMDIR)/include -I$(FIRMDIR)/export -I$(FIRMDIR)/common \
  -I$(FIRMDIR)/drivers -I$(APPSDIR) -Ilib -I$(BUILDDIR)
 CFLAGS = $(GCCOPTS) $(INCLUDES) $(TARGET) $(EXTRA_DEFINES) \
- -DMEM=${MEMORYSIZE}
+ -DMEM=${MEMORYSIZE} -DCODEC
 
 ifdef APPEXTRA
    INCLUDES += -I$(APPSDIR)/$(APPEXTRA)
 endif
 
+ifdef SOFTWARECODECS
+   CODECLIBS = -lmad -la52 -lFLAC -lTremor -lwavpack -lmusepack
+endif
+
+# we "borrow" the plugin LDS file
+LDS := $(APPSDIR)/plugins/plugin.lds
+
+LINKCODEC := $(OBJDIR)/codeclink.lds
+DEPFILE = $(OBJDIR)/dep-codecs
+
+# This sets up 'SRC' based on the files mentioned in SOURCES
+include $(TOOLSDIR)/makesrc.inc
+
+ROCKS := $(SRC:%.c=$(OBJDIR)/%.codec)
+SOURCES = $(SRC)
+ELFS := $(SRC:%.c=$(OBJDIR)/%.elf)
+OBJS := $(SRC:%.c=$(OBJDIR)/%.o)
+# as created by the cross-compiler for win32:
+DEFS := $(SRC:%.c=$(OBJDIR)/%.def)
+DIRS = .
+
 .PHONY: libmad liba52 libFLAC libTremor libwavpack dumb libmusepack
 
 OUTPUT = $(SOFTWARECODECS)
 
-all: $(OUTPUT)
+all: $(OUTPUT) $(ROCKS) $(DEPFILE)
+
+ifndef SIMVER
+$(OBJDIR)/%.elf: $(OBJDIR)/%.o $(LINKCODEC)
+	$(SILENT)(file=`basename $@`; \
+	echo "LD $$file"; \
+	$(CC) $(GCCOPTS) -O -nostdlib -o $@ $< -L$(BUILDDIR) $(CODECLIBS) -lcodec -lgcc -T$(LINKCODEC) -Wl,-Map,$(OBJDIR)/$*.map)
+
+$(OBJDIR)/%.codec : $(OBJDIR)/%.elf
+	@echo "OBJCOPY "`basename $@`
+	@$(OC) -O binary $< $@
+else
+
+ifeq ($(SIMVER), x11)
+###################################################
+# This is the X11 simulator version
+
+$(OBJDIR)/%.codec : $(OBJDIR)/%.o  $(BUILDDIR)/libplugin.a
+	@echo "LD "`basename $@`
+	@$(CC) $(CFLAGS) -shared $< -L$(BUILDDIR)  $(CODECLIBS) -lplugin -o $@
+ifeq ($(findstring CYGWIN,$(UNAME)),CYGWIN)
+# 'x' must be kept or you'll have "Win32 error 5"
+#     $ fgrep 5 /usr/include/w32api/winerror.h | head -1
+#         #define ERROR_ACCESS_DENIED 5L
+else
+	@chmod -x $@
+endif
+
+else # end of x11-simulator
+###################################################
+# This is the win32 simulator version
+DLLTOOLFLAGS = --export-all
+DLLWRAPFLAGS = -s --entry _DllMain@12 --target=i386-mingw32 -mno-cygwin
+
+$(OBJDIR)/%.codec : $(OBJDIR)/%.o  $(BUILDDIR)/libplugin.a
+	@echo "DLL "`basename $@`
+	@$(DLLTOOL) $(DLLTOOLFLAGS) -z $(OBJDIR)/$*.def $<
+	@$(DLLWRAP) $(DLLWRAPFLAGS) --def $(OBJDIR)/$*.def $< $(BUILDDIR)/libplugin.a \
+                $(patsubst -l%,$(BUILDDIR)/lib%.a,$(CODECLIBS)) -o $@
+ifeq ($(findstring CYGWIN,$(UNAME)),CYGWIN)
+# 'x' must be kept or you'll have "Win32 error 5"
+#     $ fgrep 5 /usr/include/w32api/winerror.h | head -1
+#         #define ERROR_ACCESS_DENIED 5L
+else
+	@chmod -x $@
+endif
+endif # end of win32-simulator
+
+endif # end of simulator section
 
 include $(TOOLSDIR)/make.inc
 
+$(BUILDDIR)/libcodec.a:
+	@echo "MAKE in codecs/lib"
+	@mkdir -p $(OBJDIR)/lib
+	@$(MAKE) -C lib OBJDIR=$(OBJDIR)/lib
+
+$(LINKCODEC): $(LDS)
+	@echo "build $@"
+	@cat $< | $(CC) -DMEMORYSIZE=$(MEMORYSIZE) -DCODEC $(INCLUDES) $(TARGET) $(DEFINES) -E -P - >$@
 
 libmad:
 	@echo "MAKE in libmad"
diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
new file mode 100644
index 0000000..d35854e
--- /dev/null
+++ b/apps/codecs/a52.c
@@ -0,0 +1,210 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+
+#include <inttypes.h>  /* Needed by a52.h */
+#include <codecs/liba52/config-a52.h>
+#include <codecs/liba52/a52.h>
+
+#include "playback.h"
+#include "lib/codeclib.h"
+
+#define BUFFER_SIZE 4096
+
+struct codec_api* rb;
+struct codec_api* ci;
+
+static float gain = 1;
+static a52_state_t * state;
+unsigned long samplesdone;
+unsigned long frequency;
+
+/* Two buffers used outside liba52 */
+static uint8_t buf[3840] IDATA_ATTR;
+static int16_t int16_samples[256*2] IDATA_ATTR;
+
+static inline int16_t convert (int32_t i)
+{
+    i >>= 15;
+    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
+}
+
+void output_audio(sample_t* samples,int flags) {
+  int i;
+
+  flags &= A52_CHANNEL_MASK | A52_LFE;
+
+  /* We may need to check the output format in flags - I'm not sure... */
+  for (i = 0; i < 256; i++) {
+    int16_samples[2*i] = convert (samples[i]);
+    int16_samples[2*i+1] = convert (samples[i+256]);
+  }
+
+  rb->yield();
+  while(!ci->audiobuffer_insert((unsigned char*)int16_samples,256*2*2))
+    rb->yield();
+}
+
+
+void a52_decode_data (uint8_t * start, uint8_t * end)
+{
+  static uint8_t * bufptr = buf;
+  static uint8_t * bufpos = buf + 7;
+
+  /*
+   * sample_rate and flags are static because this routine could
+   * exit between the a52_syncinfo() and the ao_setup(), and we want
+   * to have the same values when we get back !
+   */
+
+  static int sample_rate;
+  static int flags;
+  int bit_rate;
+  int len;
+
+  while (1) {
+    len = end - start;
+    if (!len)
+      break;
+    if (len > bufpos - bufptr)
+      len = bufpos - bufptr;
+    memcpy (bufptr, start, len);
+    bufptr += len;
+    start += len;
+    if (bufptr == bufpos) {
+      if (bufpos == buf + 7) {
+        int length;
+
+        length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate);
+        if (!length) {
+          DEBUGF("skip\n");
+          for (bufptr = buf; bufptr < buf + 6; bufptr++)
+            bufptr[0] = bufptr[1];
+          continue;
+        }
+        bufpos = buf + length;
+      } else {
+        // The following two defaults are taken from audio_out_oss.c:
+        level_t level;
+        sample_t bias;
+        int i;
+
+        /* This is the configuration for the downmixing: */
+        flags=A52_STEREO|A52_ADJUST_LEVEL|A52_LFE;
+        level=(1 << 26);
+        bias=0;
+
+        level = (level_t) (level * gain);
+
+        if (a52_frame (state, buf, &flags, &level, bias)) {
+          goto error;
+        }
+
+//        file_info->frames_decoded++;
+
+//        /* We assume this never changes */
+//        file_info->samplerate=sample_rate;
+          frequency=sample_rate;
+
+        // An A52 frame consists of 6 blocks of 256 samples
+        // So we decode and output them one block at a time
+        for (i = 0; i < 6; i++) {
+          if (a52_block (state)) {
+            goto error;
+          }
+
+          output_audio(a52_samples (state),flags);
+          samplesdone+=256;
+        }
+        ci->set_elapsed(samplesdone/(frequency/1000));
+        bufptr = buf;
+        bufpos = buf + 7;
+        continue;
+
+        error:
+
+        //logf("Error decoding A52 stream\n");
+        bufptr = buf;
+        bufpos = buf + 7;
+      }
+    }
+  }
+}
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+  size_t n;
+  unsigned char* filebuf;
+
+  /* Generic codec initialisation */
+  TEST_CODEC_API(api);
+
+  rb = api;
+  ci = (struct codec_api*)api;
+
+#ifndef SIMULATOR
+  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
+  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
+
+  next_track:
+
+  if (codec_init(api)) {
+      return CODEC_ERROR;
+  }
+
+  /* Intialise the A52 decoder and check for success */
+  state = a52_init (0); // Parameter is "accel"
+
+  /* The main decoding loop */
+
+  samplesdone=0;
+  while (1) {
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    filebuf=ci->request_buffer(&n,BUFFER_SIZE);
+
+    if (n==0) {      /* End of Stream */
+      break;
+    }
+  
+    a52_decode_data(filebuf,filebuf+n);
+
+    ci->advance_buffer(n);
+  }
+
+  if (ci->request_next_track())
+   goto next_track;
+
+//NOT NEEDED??:  a52_free (state);
+
+  return CODEC_OK;
+}
diff --git a/apps/codecs/flac.c b/apps/codecs/flac.c
new file mode 100644
index 0000000..93134bb
--- /dev/null
+++ b/apps/codecs/flac.c
@@ -0,0 +1,248 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2002 Björn Stenberg
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+
+#include <codecs/libFLAC/include/FLAC/seekable_stream_decoder.h>
+#include "playback.h"
+#include "lib/codeclib.h"
+
+#define FLAC_MAX_SUPPORTED_BLOCKSIZE 4608
+#define FLAC_MAX_SUPPORTED_CHANNELS  2
+
+static struct codec_api* rb;
+static uint32_t samplesdone;
+
+/* Called when the FLAC decoder needs some FLAC data to decode */
+FLAC__SeekableStreamDecoderReadStatus flac_read_handler(const FLAC__SeekableStreamDecoder *dec, 
+                                                        FLAC__byte buffer[], unsigned *bytes, void *data)
+{   struct codec_api* ci = (struct codec_api*)data;
+    (void)dec;
+
+    *bytes=(unsigned)(ci->read_filebuf(buffer,*bytes));
+
+    if (*bytes==0) {
+      return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
+    } else {
+      return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+    }
+}
+
+static unsigned char pcmbuf[FLAC_MAX_SUPPORTED_BLOCKSIZE*FLAC_MAX_SUPPORTED_CHANNELS*2] IDATA_ATTR;
+
+/* Called when the FLAC decoder has some decoded PCM data to write */
+FLAC__StreamDecoderWriteStatus flac_write_handler(const FLAC__SeekableStreamDecoder *dec, 
+  					          const FLAC__Frame *frame, 
+					          const FLAC__int32 * const buf[], 
+					          void *data)
+{
+   struct codec_api* ci = (struct codec_api*)data;
+   (void)dec;
+   unsigned int c_samp, c_chan, d_samp;
+   uint32_t data_size = frame->header.blocksize * frame->header.channels * 2; /* Assume 16-bit words */
+   uint32_t samples = frame->header.blocksize;
+   int yieldcounter = 0;
+
+
+   if (samples*frame->header.channels > (FLAC_MAX_SUPPORTED_BLOCKSIZE*FLAC_MAX_SUPPORTED_CHANNELS)) {
+     // ERROR!!!
+     DEBUGF("ERROR: samples*frame->header.channels=%d\n",samples*frame->header.channels);
+     return(FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE);
+   }
+
+   (void)dec;
+   for(c_samp = d_samp = 0; c_samp < samples; c_samp++) {
+     for(c_chan = 0; c_chan < frame->header.channels; c_chan++, d_samp++) {
+       pcmbuf[d_samp*2] = (buf[c_chan][c_samp]&0xff00)>>8;
+       pcmbuf[(d_samp*2)+1] = buf[c_chan][c_samp]&0xff;
+       if (yieldcounter++ == 100) {
+         rb->yield();
+         yieldcounter = 0;
+       }
+     }
+   } 
+
+   samplesdone+=samples;
+   ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
+   
+   rb->yield();
+   while (!ci->audiobuffer_insert(pcmbuf, data_size))
+     rb->yield();
+
+   return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+}
+
+void flac_metadata_handler(const FLAC__SeekableStreamDecoder *dec, 
+                           const FLAC__StreamMetadata *meta, void *data)
+{
+  /* Ignore metadata for now... */
+  (void)dec;
+  (void)meta;
+  (void)data;
+}
+
+
+void flac_error_handler(const FLAC__SeekableStreamDecoder *dec, 
+                        FLAC__StreamDecoderErrorStatus status, void *data)
+{
+    (void)dec;
+    (void)status;
+    (void)data;
+}
+
+FLAC__SeekableStreamDecoderSeekStatus flac_seek_handler (const FLAC__SeekableStreamDecoder *decoder, 
+                                                         FLAC__uint64 absolute_byte_offset,
+                                                         void *client_data)
+{
+  (void)decoder;
+  struct codec_api* ci = (struct codec_api*)client_data;
+
+  if (ci->seek_buffer(absolute_byte_offset)) {
+    return(FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK);
+  } else {
+    return(FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_ERROR);
+  }
+}
+
+FLAC__SeekableStreamDecoderTellStatus flac_tell_handler (const FLAC__SeekableStreamDecoder *decoder, 
+                                                         FLAC__uint64 *absolute_byte_offset, void *client_data)
+{ 
+  struct codec_api* ci = (struct codec_api*)client_data;
+
+  (void)decoder;
+  *absolute_byte_offset=ci->curpos;
+  return(FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK);
+}
+
+FLAC__SeekableStreamDecoderLengthStatus flac_length_handler (const FLAC__SeekableStreamDecoder *decoder, 
+                                                         FLAC__uint64 *stream_length, void *client_data)
+{
+  struct codec_api* ci = (struct codec_api*)client_data;
+
+  (void)decoder;
+  *stream_length=ci->filesize;
+  return(FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK);
+}
+
+FLAC__bool flac_eof_handler (const FLAC__SeekableStreamDecoder *decoder, 
+                             void *client_data)
+{
+  struct codec_api* ci = (struct codec_api*)client_data;
+
+  (void)decoder;
+  if (ci->curpos >= ci->filesize) {
+    return(true);
+  } else {
+    return(false);
+  }
+}
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    struct codec_api* ci = api;
+    FLAC__SeekableStreamDecoder* flacDecoder;
+
+  /* Generic codec initialisation */
+  TEST_CODEC_API(api);
+
+  /* if you are using a global api pointer, don't forget to copy it!
+     otherwise you will get lovely "I04: IllInstr" errors... :-) */
+  rb = api;
+
+#ifndef SIMULATOR
+  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
+  ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
+  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
+
+  next_track:
+
+  if (codec_init(api)) {
+    return CODEC_ERROR;
+  }
+
+  /* Create a decoder instance */
+
+  flacDecoder=FLAC__seekable_stream_decoder_new();
+
+  /* Set up the decoder and the callback functions - this must be done before init */
+
+  /* The following are required for stream_decoder and higher */
+  FLAC__seekable_stream_decoder_set_client_data(flacDecoder,ci);
+  FLAC__seekable_stream_decoder_set_write_callback(flacDecoder, flac_write_handler);
+  FLAC__seekable_stream_decoder_set_read_callback(flacDecoder, flac_read_handler);
+  FLAC__seekable_stream_decoder_set_metadata_callback(flacDecoder, flac_metadata_handler);
+  FLAC__seekable_stream_decoder_set_error_callback(flacDecoder, flac_error_handler);
+  FLAC__seekable_stream_decoder_set_metadata_respond(flacDecoder, FLAC__METADATA_TYPE_STREAMINFO);
+
+  /* The following are only for the seekable_stream_decoder */
+  FLAC__seekable_stream_decoder_set_seek_callback(flacDecoder, flac_seek_handler);
+  FLAC__seekable_stream_decoder_set_tell_callback(flacDecoder, flac_tell_handler);
+  FLAC__seekable_stream_decoder_set_length_callback(flacDecoder, flac_length_handler);
+  FLAC__seekable_stream_decoder_set_eof_callback(flacDecoder, flac_eof_handler);
+
+
+  /* QUESTION: What do we do when the init fails? */
+  if (FLAC__seekable_stream_decoder_init(flacDecoder)) {
+    return CODEC_ERROR;
+  }
+
+  /* The first thing to do is to parse the metadata */
+  FLAC__seekable_stream_decoder_process_until_end_of_metadata(flacDecoder);
+
+  samplesdone=0;
+  ci->set_elapsed(0);
+  /* The main decoder loop */
+  while (FLAC__seekable_stream_decoder_get_state(flacDecoder)!=FLAC__SEEKABLE_STREAM_DECODER_END_OF_STREAM) {
+    rb->yield();
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    if (ci->seek_time) {
+        int sample_loc;
+
+        sample_loc = ci->seek_time/1000 * ci->id3->frequency;
+        if (FLAC__seekable_stream_decoder_seek_absolute(flacDecoder,sample_loc)) {
+          samplesdone=sample_loc;
+          ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
+        }
+        ci->seek_time = 0;
+    }
+
+    FLAC__seekable_stream_decoder_process_single(flacDecoder);
+  }
+
+  /* Flush the libFLAC buffers */
+  FLAC__seekable_stream_decoder_finish(flacDecoder);
+
+  if (ci->request_next_track())
+    goto next_track;
+
+  return CODEC_OK;
+}
diff --git a/apps/codecs/lib/Makefile b/apps/codecs/lib/Makefile
new file mode 100644
index 0000000..08dcd2a
--- /dev/null
+++ b/apps/codecs/lib/Makefile
@@ -0,0 +1,46 @@
+#             __________               __   ___.
+#   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+#   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+#   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+#   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+#                     \/            \/     \/    \/            \/
+# $Id$
+#
+
+# ../.. for the codec.h in the apps dir
+# .. for stuff in the codecs dir
+# . for stuff in the codeclib dir
+INCLUDES=-I$(APPSDIR) -I.. -I. -I$(FIRMDIR)/include -I$(FIRMDIR)/export	\
+ -I$(FIRMDIR)/common -I$(BUILDDIR)
+
+ifdef APPEXTRA
+INCLUDES += -I$(APPSDIR)/$(APPEXTRA)
+endif
+
+CFLAGS = $(GCCOPTS) \
+$(INCLUDES) $(TARGET) $(EXTRA_DEFINES) -DMEM=${MEMORYSIZE} -DCODEC
+
+# This sets up 'SRC' based on the files mentioned in SOURCES
+include $(TOOLSDIR)/makesrc.inc
+
+SOURCES = $(SRC)
+OBJS := $(SRC:%.c=$(OBJDIR)/%.o)
+DEPFILE = $(OBJDIR)/dep-codeclib
+DIRS = .
+
+OUTPUT = $(BUILDDIR)/libcodec.a
+
+all: $(OUTPUT)
+
+$(OUTPUT): $(OBJS)
+	@echo "AR+RANLIB $@"
+	@$(AR) ruv $@ $+ >/dev/null 2>&1
+	@$(RANLIB) $@
+
+include $(TOOLSDIR)/make.inc
+
+clean:
+	@echo "cleaning lib"
+	@rm -f $(OBJS) $(OUTPUT) $(DEPFILE)
+
+-include $(DEPFILE)
diff --git a/apps/codecs/lib/codeclib.c b/apps/codecs/lib/codeclib.c
new file mode 100644
index 0000000..3aa6216
--- /dev/null
+++ b/apps/codecs/lib/codeclib.c
@@ -0,0 +1,38 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+/* "helper functions" common to all codecs  */
+
+#include "plugin.h"
+#include "playback.h"
+#include "codeclib.h"
+#include "xxx2wav.h"
+
+struct codec_api *local_rb;
+
+int codec_init(struct codec_api* rb)
+{
+    local_rb = rb;
+    
+    xxx2wav_set_api(rb);
+    mem_ptr = 0;
+    mallocbuf = (unsigned char *)rb->get_codec_memory((size_t *)&bufsize);
+  
+    return 0;
+}
diff --git a/apps/codecs/lib/codeclib.h b/apps/codecs/lib/codeclib.h
new file mode 100644
index 0000000..116f210
--- /dev/null
+++ b/apps/codecs/lib/codeclib.h
@@ -0,0 +1,49 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "config.h"
+#include "codecs.h"
+
+/* Various codec "helper functions" */
+
+#if CONFIG_CPU == MCF5249 && !defined(SIMULATOR)
+#define ICODE_ATTR	__attribute__ ((section(".icode")))
+#define IDATA_ATTR	__attribute__ ((section(".idata")))
+#define USE_IRAM	1
+#else
+#define ICODE_ATTR	
+#define IDATA_ATTR	 
+#endif
+
+extern int mem_ptr;
+extern int bufsize;
+extern unsigned char* mallocbuf;  // 512K from the start of MP3 buffer
+
+void* codec_malloc(size_t size);
+void* codec_calloc(size_t nmemb, size_t size);
+void* codec_alloca(size_t size);
+void* codec_realloc(void* ptr, size_t size);
+void codec_free(void* ptr);
+void *memcpy(void *dest, const void *src, size_t n);
+void *memset(void *s, int c, size_t n);
+int memcmp(const void *s1, const void *s2, size_t n);
+void* memmove(const void *s1, const void *s2, size_t n);
+
+int codec_init(struct codec_api* rb);
+
diff --git a/apps/codecs/lib/xxx2wav.c b/apps/codecs/lib/xxx2wav.c
new file mode 100644
index 0000000..1bc1837
--- /dev/null
+++ b/apps/codecs/lib/xxx2wav.c
@@ -0,0 +1,270 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+/* Various "helper functions" common to all the xxx2wav decoder plugins  */
+
+#if (CONFIG_HWCODEC == MASNONE)
+/* software codec platforms, not for simulator */
+
+#include "codecs.h"
+#include "xxx2wav.h"
+
+static struct codec_api* local_rb;
+
+int mem_ptr;
+int bufsize;
+unsigned char* audiobuf;     // The actual audio buffer from Rockbox
+unsigned char* mallocbuf;  // 512K from the start of audio buffer
+unsigned char* filebuf;    // The rest of the audio buffer
+
+void* codec_malloc(size_t size)
+{
+    void* x;
+
+    x=&mallocbuf[mem_ptr];
+    mem_ptr+=(size+3)&~3; // Keep memory 32-bit aligned (if it was already?)
+/*
+  if(TIME_AFTER(*(local_rb->current_tick), last_tick + HZ)) {
+      char s[32];
+      static long last_tick = 0;
+      local_rb->snprintf(s,30,"Memory used: %d",mem_ptr);
+      local_rb->lcd_putsxy(0,80,s);
+
+      last_tick = *(local_rb->current_tick);
+      local_rb->lcd_update();
+  }*/
+    return(x);
+}
+
+void* codec_calloc(size_t nmemb, size_t size)
+{
+    void* x;
+    x = codec_malloc(nmemb*size);
+    local_rb->memset(x,0,nmemb*size);
+    return(x);
+}
+
+void* codec_alloca(size_t size)
+{
+    void* x;
+    x = codec_malloc(size);
+    return(x);
+}
+
+void codec_free(void* ptr) {
+    (void)ptr;
+}
+
+void* codec_realloc(void* ptr, size_t size) {
+  void* x;
+  (void)ptr;
+  x = codec_malloc(size);
+  return(x);
+}
+
+void *memcpy(void *dest, const void *src, size_t n) {
+  return(local_rb->memcpy(dest,src,n));
+}
+
+void *memset(void *s, int c, size_t n) {
+  return(local_rb->memset(s,c,n));
+}
+
+int memcmp(const void *s1, const void *s2, size_t n) {
+  return(local_rb->memcmp(s1,s2,n));
+}
+
+void* memchr(const void *s, int c, size_t n) {
+  /* TO DO: Implement for Tremor */
+  (void)s;
+  (void)c;
+  (void)n;
+  return(NULL);
+}
+
+void* memmove(const void *s1, const void *s2, size_t n) {
+  char* dest=(char*)s1;
+  char* src=(char*)s2;
+  size_t i;
+
+  for (i=0;i<n;i++) { dest[i]=src[i]; }
+  //  while(n>0) { *(dest++)=*(src++); n--; }
+  return(dest);
+}
+
+void qsort(void *base, size_t nmemb, size_t size, int(*compar)(const void *, const void *)) {
+  local_rb->qsort(base,nmemb,size,compar);
+}
+
+void display_status(file_info_struct* file_info) {
+  char s[32];
+  unsigned long ticks_taken;
+  unsigned long long speed;
+  unsigned long xspeed;
+  static long last_tick = 0;
+
+  if(TIME_AFTER(*(local_rb->current_tick), last_tick + HZ)) {
+      local_rb->snprintf(s,32,"Bytes read: %d",file_info->curpos);
+      local_rb->lcd_putsxy(0,0,s);
+      local_rb->snprintf(s,32,"Samples Decoded: %d",file_info->current_sample);
+      local_rb->lcd_putsxy(0,20,s);
+      local_rb->snprintf(s,32,"Frames Decoded: %d",file_info->frames_decoded);
+      local_rb->lcd_putsxy(0,40,s);
+
+      ticks_taken=*(local_rb->current_tick)-file_info->start_tick;
+
+      /* e.g.:
+       ticks_taken=500
+       sam_fmt.rate=44,100
+       samples_decoded=172,400
+       (samples_decoded/sam_fmt.rate)*100=400 (time it should have taken)
+       % Speed=(400/500)*100=80%
+      */
+
+      if (ticks_taken==0) { ticks_taken=1; } // Avoid fp exception.
+
+      speed=(100*file_info->current_sample)/file_info->samplerate;
+      xspeed=(speed*10000)/ticks_taken;
+      local_rb->snprintf(s,32,"Speed %ld.%02ld %% Secs: %d",(xspeed/100),(xspeed%100),ticks_taken/100);
+      local_rb->lcd_putsxy(0,60,s);
+
+      last_tick = *(local_rb->current_tick);
+      local_rb->lcd_update();
+  }
+}
+
+#if 0
+static unsigned char wav_header[44]={'R','I','F','F',    //  0 - ChunkID
+                              0,0,0,0,            //  4 - ChunkSize (filesize-8)
+                              'W','A','V','E',    //  8 - Format
+                              'f','m','t',' ',    // 12 - SubChunkID
+                              16,0,0,0,           // 16 - SubChunk1ID  // 16 for PCM
+                              1,0,                // 20 - AudioFormat (1=16-bit)
+                              2,0,                // 22 - NumChannels
+                              0,0,0,0,            // 24 - SampleRate in Hz
+                              0,0,0,0,            // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
+                              4,0,                // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
+                              16,0,               // 34 - BitsPerSample
+                              'd','a','t','a',    // 36 - Subchunk2ID
+                              0,0,0,0             // 40 - Subchunk2Size
+                             };
+#endif
+
+void xxx2wav_set_api(struct codec_api* rb)
+{
+    local_rb = rb;
+}
+
+#if 0
+int local_init(char* infilename, char* outfilename,
+               file_info_struct* file_info,
+               struct codec_api* rb)
+{
+  char s[32];
+  int i,n,bytesleft;
+
+  local_rb=rb;
+
+  mem_ptr=0;
+  audiobuf=local_rb->plugin_get_audio_buffer(&bufsize);
+  mallocbuf=audiobuf;
+  filebuf=&audiobuf[MALLOC_BUFSIZE];
+
+  local_rb->snprintf(s,32,"audio bufsize: %d",bufsize);
+  local_rb->lcd_putsxy(0,100,s);
+  local_rb->lcd_update();
+
+  file_info->infile=local_rb->open(infilename,O_RDONLY);
+  file_info->outfile=local_rb->creat(outfilename,O_WRONLY);
+  local_rb->write(file_info->outfile,wav_header,sizeof(wav_header));
+  file_info->curpos=0;
+  file_info->current_sample=0;
+  file_info->frames_decoded=0;
+  file_info->filesize=local_rb->filesize(file_info->infile);
+
+  local_rb->splash(HZ, true, "in: %d, size: %d", file_info->infile, file_info->filesize);
+  
+  if (file_info->filesize > (bufsize-MALLOC_BUFSIZE)) {
+    local_rb->close(file_info->infile);
+    local_rb->splash(HZ*2, true, "File too large");
+    return(1);
+  }
+
+  local_rb->snprintf(s,32,"Loading file...");
+  local_rb->lcd_putsxy(0,0,s);
+  local_rb->lcd_update();
+
+  bytesleft=file_info->filesize;
+  i=0;
+  while (bytesleft > 0) {
+    n=local_rb->read(file_info->infile,&filebuf[i],bytesleft);
+    if (n < 0) {
+      local_rb->close(file_info->infile);
+      local_rb->splash(HZ*2, true, "ERROR READING FILE");
+      return(1);
+    }      
+    i+=n; bytesleft-=n;
+  }
+  local_rb->close(file_info->infile);
+  local_rb->lcd_clear_display();
+  return(0);
+}
+
+void close_wav(file_info_struct* file_info)
+{
+  int x;
+  int filesize=local_rb->filesize(file_info->outfile);
+
+  /* We assume 16-bit, Stereo */
+
+  local_rb->lseek(file_info->outfile,0,SEEK_SET);
+
+  // ChunkSize
+  x=filesize-8;
+  wav_header[4]=(x&0xff);
+  wav_header[5]=(x&0xff00)>>8;
+  wav_header[6]=(x&0xff0000)>>16;
+  wav_header[7]=(x&0xff000000)>>24;
+
+  // Samplerate
+  wav_header[24]=file_info->samplerate&0xff;
+  wav_header[25]=(file_info->samplerate&0xff00)>>8;
+  wav_header[26]=(file_info->samplerate&0xff0000)>>16;
+  wav_header[27]=(file_info->samplerate&0xff000000)>>24;
+
+  // ByteRate
+  x=file_info->samplerate*4;
+  wav_header[28]=(x&0xff);
+  wav_header[29]=(x&0xff00)>>8;
+  wav_header[30]=(x&0xff0000)>>16;
+  wav_header[31]=(x&0xff000000)>>24;
+  
+  // Subchunk2Size
+  x=filesize-44;
+  wav_header[40]=(x&0xff);
+  wav_header[41]=(x&0xff00)>>8;
+  wav_header[42]=(x&0xff0000)>>16;
+  wav_header[43]=(x&0xff000000)>>24;
+
+  local_rb->write(file_info->outfile,wav_header,sizeof(wav_header));
+  local_rb->close(file_info->outfile);
+}
+#endif /* 0 */
+
+#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/codecs/lib/xxx2wav.h b/apps/codecs/lib/xxx2wav.h
new file mode 100644
index 0000000..1fa7dc9
--- /dev/null
+++ b/apps/codecs/lib/xxx2wav.h
@@ -0,0 +1,69 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+/* Various "helper functions" common to all the xxx2wav decoder plugins  */
+
+#if CONFIG_CPU == MCF5249 && !defined(SIMULATOR)
+#define ICODE_ATTR	__attribute__ ((section(".icode")))
+#define IDATA_ATTR	__attribute__ ((section(".idata")))
+#define USE_IRAM	1
+#else
+#define ICODE_ATTR	
+#define IDATA_ATTR	 
+#endif
+
+/* the main data structure of the program */
+typedef struct {
+    int infile;
+    int outfile;
+    off_t curpos;
+    off_t filesize;
+    int samplerate;
+    int bitspersample;
+    int channels;
+    int frames_decoded;
+    unsigned long total_samples;
+    unsigned long current_sample;
+    unsigned long start_tick;
+} file_info_struct;
+
+#define MALLOC_BUFSIZE (512*1024)
+
+extern int mem_ptr;
+extern int bufsize;
+extern unsigned char* mp3buf;     // The actual MP3 buffer from Rockbox
+extern unsigned char* mallocbuf;  // 512K from the start of MP3 buffer
+extern unsigned char* filebuf;    // The rest of the MP3 buffer
+
+void* codec_malloc(size_t size);
+void* codec_calloc(size_t nmemb, size_t size);
+void* codec_alloca(size_t size);
+void* codec_realloc(void* ptr, size_t size);
+void codec_free(void* ptr);
+void *memcpy(void *dest, const void *src, size_t n);
+void *memset(void *s, int c, size_t n);
+int memcmp(const void *s1, const void *s2, size_t n);
+void* memmove(const void *s1, const void *s2, size_t n);
+
+void display_status(file_info_struct* file_info);
+int local_init(char* infilename, char* outfilename,
+               file_info_struct* file_info,
+               struct codec_api* rb);
+void close_wav(file_info_struct* file_info);
+void xxx2wav_set_api(struct codec_api* rb);
diff --git a/apps/codecs/mpa.c b/apps/codecs/mpa.c
new file mode 100644
index 0000000..beb71d7
--- /dev/null
+++ b/apps/codecs/mpa.c
@@ -0,0 +1,521 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+
+#include <codecs/libmad/mad.h>
+
+#include "playback.h"
+#include "mp3data.h"
+#include "lib/codeclib.h"
+
+static struct codec_api* rb;
+
+struct mad_stream  Stream IDATA_ATTR;
+struct mad_frame  Frame IDATA_ATTR;
+struct mad_synth  Synth IDATA_ATTR;
+mad_timer_t      Timer;
+struct dither d0, d1;
+
+/* The following function is used inside libmad - let's hope it's never
+   called. 
+*/
+
+void abort(void) {
+}
+
+/* The "dither" code to convert the 24-bit samples produced by libmad was
+   taken from the coolplayer project - coolplayer.sourceforge.net */
+
+struct dither {
+	mad_fixed_t error[3];
+	mad_fixed_t random;
+};
+
+# define SAMPLE_DEPTH	16
+# define scale(x, y)	dither((x), (y))
+
+/*
+ * NAME:		prng()
+ * DESCRIPTION:	32-bit pseudo-random number generator
+ */
+static __inline
+unsigned long prng(unsigned long state)
+{
+  return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
+}
+
+/*
+ * NAME:        dither()
+ * DESCRIPTION:	dither and scale sample
+ */
+static __inline
+signed int dither(mad_fixed_t sample, struct dither *dither)
+{
+  unsigned int scalebits;
+  mad_fixed_t output, mask, random;
+
+  enum {
+    MIN = -MAD_F_ONE,
+    MAX =  MAD_F_ONE - 1
+  };
+
+  /* noise shape */
+  sample += dither->error[0] - dither->error[1] + dither->error[2];
+
+  dither->error[2] = dither->error[1];
+  dither->error[1] = dither->error[0] / 2;
+
+  /* bias */
+  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
+
+  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
+  mask = (1L << scalebits) - 1;
+
+  /* dither */
+  random  = prng(dither->random);
+  output += (random & mask) - (dither->random & mask);
+
+  //dither->random = random;
+
+  /* clip */
+  if (output > MAX) {
+    output = MAX;
+
+    if (sample > MAX)
+      sample = MAX;
+  }
+  else if (output < MIN) {
+    output = MIN;
+
+    if (sample < MIN)
+      sample = MIN;
+  }
+
+  /* quantize */
+  output &= ~mask;
+
+  /* error feedback */
+  dither->error[0] = sample - output;
+
+  /* scale */
+  return output >> scalebits;
+}
+
+static __inline
+signed int detect_silence(mad_fixed_t sample)
+{
+  unsigned int scalebits;
+  mad_fixed_t output, mask;
+
+  enum {
+    MIN = -MAD_F_ONE,
+    MAX =  MAD_F_ONE - 1
+  };
+
+  /* bias */
+  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
+
+  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
+  mask = (1L << scalebits) - 1;
+
+  /* clip */
+  if (output > MAX) {
+    output = MAX;
+
+    if (sample > MAX)
+      sample = MAX;
+  }
+  else if (output < MIN) {
+    output = MIN;
+
+    if (sample < MIN)
+      sample = MIN;
+  }
+
+  /* quantize */
+  output &= ~mask;
+
+  /* scale */
+  output >>= scalebits + 4;
+  
+  if (output == 0x00 || output == 0xff)
+    return 1;
+    
+  return 0;
+}
+#define SHRT_MAX 32767
+
+#define INPUT_CHUNK_SIZE        8192
+#define OUTPUT_BUFFER_SIZE	65536 /* Must be an integer multiple of 4. */
+
+unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
+unsigned char *OutputPtr;
+unsigned char *GuardPtr=NULL;
+const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
+long resampled_data[2][5000]; /* enough to cope with 11khz upsampling */
+
+mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
+unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
+/* TODO: what latency does layer 1 have? */
+int mpeg_latency[3] = { 0, 481, 529 };
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+#undef DEBUG_GAPLESS
+
+struct resampler {
+    long last_sample, phase, delta;
+};
+
+#if CONFIG_CPU==MCF5249 && !defined(SIMULATOR)
+
+#define INIT() asm volatile ("move.l #0xb0, %macsr") /* frac, round, clip */
+#define FRACMUL(x, y) \
+({ \
+    long t; \
+    asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
+                  "movclr.l %%acc0, %[t]\n\t" \
+                  : [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
+    t; \
+})
+
+#else
+
+#define INIT()
+#define FRACMUL(x, y) (long)(((long long)(x)*(long long)(y)) << 1)
+#endif
+
+/* linear resampling, introduces one sample delay, because of our inability to
+   look into the future at the end of a frame */
+long downsample(long *in, long *out, int num, struct resampler *s)
+{
+    long i = 1, pos;
+    long last = s->last_sample;
+
+    INIT();
+    pos = s->phase >> 16;
+    /* check if we need last sample of previous frame for interpolation */
+    if (pos > 0)
+        last = in[pos - 1];
+    out[0] = last + FRACMUL((s->phase & 0xffff) << 15, in[pos] - last);
+    s->phase += s->delta;
+    while ((pos = s->phase >> 16) < num) {
+        out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
+        s->phase += s->delta;
+    }
+    /* wrap phase accumulator back to start of next frame */
+    s->phase -= num << 16;
+    s->last_sample = in[num - 1];
+    return i;
+}
+
+long upsample(long *in, long *out, int num, struct resampler *s)
+{
+    long i = 0, pos;
+
+    INIT();
+    while ((pos = s->phase >> 16) == 0) {
+        out[i++] = s->last_sample + FRACMUL((s->phase & 0xffff) << 15, in[pos] - s->last_sample);
+        s->phase += s->delta;
+    }
+    while ((pos = s->phase >> 16) < num) {
+        out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
+        s->phase += s->delta;
+    }
+    /* wrap phase accumulator back to start of next frame */
+    s->phase -= num << 16;
+    s->last_sample = in[num - 1];
+    return i;
+}
+
+long resample(long *in, long *out, int num, struct resampler *s)
+{
+    if (s->delta >= (1 << 16))
+        return downsample(in, out, num, s);
+    else
+        return upsample(in, out, num, s);
+}
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    struct codec_api *ci = api;
+    struct mp3info *info;
+    int Status=0;
+    size_t size;
+    int file_end;
+    unsigned short Sample;
+    char *InputBuffer;
+    unsigned int samplecount;
+    unsigned int samplesdone;
+    bool first_frame;
+#ifdef DEBUG_GAPLESS
+    bool first = true;
+    int fd;
+#endif
+    int i;
+    int yieldcounter = 0;
+    int stop_skip, start_skip;
+    struct resampler lr = { 0, 0, 0 }, rr = { 0, 0, 0 };
+    long length;  
+    /* Generic codec inititialisation */
+    (void)parm;
+
+    TEST_CODEC_API(api);
+    rb = api;
+
+#ifdef USE_IRAM
+    rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+    /* This function sets up the buffers and reads the file into RAM */
+  
+    if (codec_init(api)) {
+        return CODEC_ERROR;
+    }
+
+    /* Create a decoder instance */
+
+    ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
+    ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
+  
+    memset(&Stream, 0, sizeof(struct mad_stream));
+    memset(&Frame, 0, sizeof(struct mad_frame));
+    memset(&Synth, 0, sizeof(struct mad_synth));
+    memset(&Timer, 0, sizeof(mad_timer_t));
+  
+    mad_stream_init(&Stream);
+    mad_frame_init(&Frame);
+    mad_synth_init(&Synth);
+    mad_timer_reset(&Timer);
+
+    /* We do this so libmad doesn't try to call codec_calloc() */
+    memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
+    Frame.overlap = &mad_frame_overlap;
+    Stream.main_data = &mad_main_data;
+    /* This label might need to be moved above all the init code, but I don't
+       think reiniting the codec is necessary for MPEG. It might even be unwanted
+       for gapless playback */
+  next_track:
+  
+#ifdef DEBUG_GAPLESS
+    if (first)
+        fd = rb->open("/first.pcm", O_WRONLY | O_CREAT);
+    else
+        fd = rb->open("/second.pcm", O_WRONLY | O_CREAT);
+    first = false;
+#endif
+  
+    info = ci->mp3data;
+    first_frame = false;
+    file_end = 0;
+    OutputPtr = OutputBuffer;
+    
+    while (!*ci->taginfo_ready)
+        rb->yield();
+  
+    ci->request_buffer(&size, ci->id3->first_frame_offset);
+    ci->advance_buffer(size);
+ 
+    if (info->enc_delay >= 0 && info->enc_padding >= 0) {
+        stop_skip = info->enc_padding - mpeg_latency[info->layer];
+        if (stop_skip < 0) stop_skip = 0;
+        start_skip = info->enc_delay + mpeg_latency[info->layer];
+    } else {
+        stop_skip = 0;
+        /* We want to skip this amount anyway */
+        start_skip = mpeg_latency[info->layer];
+    }
+  
+    /* NOTE: currently this doesn't work, the below calculated samples_count
+       seems to be right, but sometimes libmad just can't supply us with
+       all the data we need... */
+    if (info->frame_count) {
+        /* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
+           it's probably not correct at all for MPEG2 and layer 1 */
+        samplecount = info->frame_count*1152 - (start_skip + stop_skip);
+        samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
+    } else {
+        samplecount = ci->id3->length * (ci->id3->frequency / 100) / 10;
+        samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
+    }
+    /* rb->snprintf(buf2, sizeof(buf2), "sc: %d", samplecount);
+       rb->splash(0, true, buf2);
+       rb->snprintf(buf2, sizeof(buf2), "length: %d", ci->id3->length);
+       rb->splash(HZ*5, true, buf2);
+       rb->snprintf(buf2, sizeof(buf2), "frequency: %d", ci->id3->frequency);
+       rb->splash(HZ*5, true, buf2); */
+    lr.delta = rr.delta = ci->id3->frequency*65536/44100; 
+    /* This is the decoding loop. */
+    while (1) {
+        rb->yield();
+        if (ci->stop_codec || ci->reload_codec) { 
+            break ;
+        }
+    
+        if (ci->seek_time) {
+            unsigned int sample_loc;
+            int newpos;
+        
+            sample_loc = ci->seek_time/1000 * ci->id3->frequency;
+            newpos = ci->mp3_get_filepos(ci->seek_time-1);
+            if (ci->seek_buffer(newpos)) {
+                if (sample_loc >= samplecount + samplesdone)
+                    break ;
+                samplecount += samplesdone - sample_loc;
+                samplesdone = sample_loc;
+            }
+            ci->seek_time = 0;
+        }
+
+        /* Lock buffers */
+        if (Stream.error == 0) {
+            InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
+            if (size == 0 || InputBuffer == NULL)
+                break ;
+            mad_stream_buffer(&Stream, InputBuffer, size);
+        }
+    
+        //if ((int)ci->curpos >= ci->id3->first_frame_offset)
+        //first_frame = true;
+        
+        if(mad_frame_decode(&Frame,&Stream))
+        {
+            if (Stream.error == MAD_FLAG_INCOMPLETE || Stream.error == MAD_ERROR_BUFLEN) {
+                // rb->splash(HZ*1, true, "Incomplete");
+                /* This makes the codec to support partially corrupted files too. */
+                if (file_end == 30)
+                    break ;
+        
+                /* Fill the buffer */
+                Stream.error = 0;
+                file_end++;
+                continue ;
+            }
+            else if(MAD_RECOVERABLE(Stream.error))
+            {
+                if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr)
+                {
+                    // rb->splash(HZ*1, true, "Recoverable...!");
+                }
+                continue;
+            }
+            else if(Stream.error==MAD_ERROR_BUFLEN) {
+                //rb->splash(HZ*1, true, "Buflen error");
+                break ;
+            } else {
+                //rb->splash(HZ*1, true, "Unrecoverable error");
+                Status=1;
+                break;
+            }
+        }
+        if (Stream.next_frame)
+            ci->advance_buffer_loc((void *)Stream.next_frame);
+        file_end = false;
+        /* ?? Do we need the timer module? */
+        // mad_timer_add(&Timer,Frame.header.duration);
+
+/* DAVE: This can be used to attenuate the audio */
+//    if(DoFilter)
+//      ApplyFilter(&Frame);   
+
+        mad_synth_frame(&Synth,&Frame);
+    
+        //if (!first_frame) {
+        //samplecount -= Synth.pcm.length;
+        //continue ;
+        //}
+    
+        /* Convert MAD's numbers to an array of 16-bit LE signed integers */
+        /* We skip start_skip number of samples here, this should only happen for
+           very first frame in the stream. */
+        /* TODO: possible for start_skip to exceed one frames worth of samples? */
+        length = resample((long *)&Synth.pcm.samples[0][start_skip], resampled_data[0], Synth.pcm.length, &lr);
+        if (MAD_NCHANNELS(&Frame.header) == 2)
+            resample((long *)&Synth.pcm.samples[1][start_skip], resampled_data[1], Synth.pcm.length, &rr);
+        for (i = 0;i<length;i++)
+        {
+            start_skip = 0; /* not very elegant, and might want to keep this value */
+            samplesdone++;
+            //if (ci->mp3data->padding > 0) {
+            //  ci->mp3data->padding--;
+            //  continue ;
+            //}
+            /*if (!first_frame) {
+              if (detect_silence(Synth.pcm.samples[0][i]))
+              continue ;
+              first_frame = true;
+              }*/
+      
+            /* Left channel */
+            Sample=scale(resampled_data[0][i],&d0);
+            *(OutputPtr++)=Sample>>8;
+            *(OutputPtr++)=Sample&0xff;
+
+            /* Right channel. If the decoded stream is monophonic then
+             * the right output channel is the same as the left one.
+             */
+            if(MAD_NCHANNELS(&Frame.header)==2)
+                Sample=scale(resampled_data[1][i],&d1);
+            *(OutputPtr++)=Sample>>8;
+            *(OutputPtr++)=Sample&0xff;
+      
+            samplecount--;
+            if (samplecount == 0) {
+#ifdef DEBUG_GAPLESS
+                rb->write(fd, OutputBuffer, (int)OutputPtr-(int)OutputBuffer);
+#endif
+                while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr-(int)OutputBuffer))
+                    rb->yield();
+                goto song_end;
+            }
+  
+            if (yieldcounter++ == 200) {
+                rb->yield();
+                yieldcounter = 0;
+            }
+      
+            /* Flush the buffer if it is full. */
+            if(OutputPtr==OutputBufferEnd)
+            {
+#ifdef DEBUG_GAPLESS
+                rb->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
+#endif
+                while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE))
+                    rb->yield();
+                OutputPtr=OutputBuffer;
+            }
+        }
+        ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
+    }
+  
+  song_end:
+#ifdef DEBUG_GAPLESS
+    rb->close(fd);
+#endif
+    Stream.error = 0;
+  
+    if (ci->request_next_track())
+        goto next_track;
+    return CODEC_OK;
+}
diff --git a/apps/codecs/mpc.c b/apps/codecs/mpc.c
new file mode 100644
index 0000000..9b4a616
--- /dev/null
+++ b/apps/codecs/mpc.c
@@ -0,0 +1,214 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Thom Johansen
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+#include "playback.h"
+#include "lib/codeclib.h"
+#include <codecs/libmusepack/musepack.h>
+
+static struct codec_api* rb;
+mpc_decoder decoder;
+
+/*
+  Our implementations of the mpc_reader callback functions.
+*/
+mpc_int32_t
+read_impl(void *data, void *ptr, mpc_int32_t size)
+{
+  struct codec_api* ci = (struct codec_api*)data;
+
+  return((mpc_int32_t)(ci->read_filebuf(ptr,size)));
+}
+
+bool
+seek_impl(void *data, mpc_int32_t offset)
+{  
+  struct codec_api* ci = (struct codec_api*)data;
+
+  /* WARNING: assumes we don't need to skip too far into the past,
+     this might not be supported by the buffering layer yet */
+  return ci->seek_buffer(offset);
+}
+
+mpc_int32_t
+tell_impl(void *data)
+{
+  struct codec_api* ci = (struct codec_api*)data;
+
+  return ci->curpos;
+}
+
+mpc_int32_t
+get_size_impl(void *data)
+{
+  struct codec_api* ci = (struct codec_api*)data;
+  return ci->filesize;
+}
+
+bool
+canseek_impl(void *data)
+{
+    (void)data;
+    return false;
+}
+
+static int
+shift_signed(MPC_SAMPLE_FORMAT val, int shift)
+{
+  if (shift > 0)
+    val <<= shift;
+  else if (shift < 0)
+    val >>= -shift;
+  return (int)val;
+}
+
+#define OUTPUT_BUFFER_SIZE 65536 /* Must be an integer multiple of 4. */
+
+unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
+/* temporary, we probably have better use for iram than this */
+MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH] IDATA_ATTR;
+unsigned char *OutputPtr=OutputBuffer;
+const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
+
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    struct codec_api* ci = api;
+  unsigned short Sample;
+  unsigned long samplesdone;
+  unsigned long frequency;
+  unsigned status = 1;
+  unsigned int i;
+  mpc_reader reader;
+
+  /* Generic codec inititialisation */
+
+  TEST_CODEC_API(api);
+  rb = api;
+
+#ifndef SIMULATOR
+  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
+  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
+
+  next_track:
+
+  if (codec_init(api)) {
+    return CODEC_ERROR;
+  }
+
+  /* Create a decoder instance */
+
+  reader.read = read_impl;
+  reader.seek = seek_impl;
+  reader.tell = tell_impl;
+  reader.get_size = get_size_impl;
+  reader.canseek = canseek_impl;
+  reader.data = ci;
+
+  /* read file's streaminfo data */
+  mpc_streaminfo info;
+  mpc_streaminfo_init(&info);
+  if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
+    return CODEC_ERROR;  
+  }
+  frequency=info.sample_freq;
+
+  /* instantiate a decoder with our file reader */
+  mpc_decoder_setup(&decoder, &reader);
+  if (!mpc_decoder_initialize(&decoder, &info)) {
+    return CODEC_ERROR;
+  }
+
+  /* Initialise the output buffer. */
+  OutputPtr=OutputBuffer;
+  OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
+
+  /* This is the decoding loop. */
+  samplesdone=0;
+  while (status != 0) {
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    status = mpc_decoder_decode(&decoder, sample_buffer, 0, 0);
+    if (status == (unsigned)(-1)) {
+      //decode error
+      return CODEC_ERROR;
+    }
+    else                    //status>0
+    {
+    //      file_info.current_sample += status;
+    //      file_info.frames_decoded++;
+      /* Convert musepack's numbers to an array of 16-bit BE signed integers */
+      for(i = 0; i < status*info.channels; i += info.channels)
+      {
+        /* Left channel */
+        Sample=shift_signed(sample_buffer[i], 16 - MPC_FIXED_POINT_SCALE_SHIFT);
+        *(OutputPtr++)=Sample>>8;
+        *(OutputPtr++)=Sample&0xff;
+
+        /* Right channel. If the decoded stream is monophonic then
+         * the right output channel is the same as the left one.
+         */
+        if(info.channels==2) {
+          Sample=shift_signed(sample_buffer[i + 1], 16 - MPC_FIXED_POINT_SCALE_SHIFT);
+        }
+        *(OutputPtr++)=Sample>>8;
+        *(OutputPtr++)=Sample&0xff;
+
+        samplesdone++;
+
+        /* Flush the buffer if it is full. */
+        if(OutputPtr==OutputBufferEnd)
+        {
+          rb->yield();
+          while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE)) {
+            rb->yield();
+          }
+
+          ci->set_elapsed(samplesdone/(frequency/1000));
+          OutputPtr=OutputBuffer;
+        }
+      }
+    }
+  }
+
+  /* Flush the remaining data in the output buffer */
+  if (OutputPtr > OutputBuffer) {
+    rb->yield();
+    while (!ci->audiobuffer_insert(OutputBuffer, OutputPtr-OutputBuffer)) {
+      rb->yield();
+    }
+  }
+
+  if (ci->request_next_track())
+    goto next_track;
+
+  return CODEC_OK;
+}
+
diff --git a/apps/codecs/vorbis.c b/apps/codecs/vorbis.c
new file mode 100644
index 0000000..969fdf3
--- /dev/null
+++ b/apps/codecs/vorbis.c
@@ -0,0 +1,168 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2002 Björn Stenberg
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+#include "kernel.h"
+#include "codecs.h"
+
+#include "Tremor/ivorbisfile.h"
+#include "playback.h"
+#include "lib/codeclib.h"
+
+static struct codec_api* rb;
+
+/* Some standard functions and variables needed by Tremor */
+
+int errno;
+
+size_t strlen(const char *s)
+{
+    return(rb->strlen(s));
+}
+
+char *strcpy(char *dest, const char *src)
+{
+    return(rb->strcpy(dest,src));
+}
+
+char *strcat(char *dest, const char *src)
+{
+    return(rb->strcat(dest,src));
+}
+
+size_t read_handler(void *ptr, size_t size, size_t nmemb, void *datasource)
+{
+    return rb->read_filebuf(ptr, nmemb*size);
+}
+
+int seek_handler(void *datasource, ogg_int64_t offset, int whence)
+{
+    /* We are not seekable at the moment */
+    (void)datasource;
+    (void)offset;
+    (void)whence;
+    return -1;
+}
+
+int close_handler(void *datasource)
+{
+    (void)datasource;
+    return 0;
+}
+
+long tell_handler(void *datasource)
+{
+    return rb->curpos;
+}
+
+#ifdef USE_IRAM
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+
+/* reserve the PCM buffer in the IRAM area */
+static char pcmbuf[4096] IDATA_ATTR;
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    ov_callbacks callbacks;
+    OggVorbis_File vf;
+    vorbis_info* vi;
+
+    int error;
+    long n;
+    int current_section;
+    int eof;
+#if BYTE_ORDER == BIG_ENDIAN
+    int i;
+    char x;
+#endif
+
+    TEST_CODEC_API(api);
+
+    /* if you are using a global api pointer, don't forget to copy it!
+       otherwise you will get lovely "I04: IllInstr" errors... :-) */
+    rb = api;
+
+#ifdef USE_IRAM
+    rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+    
+    rb->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
+    rb->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*64));
+    
+    /* We need to flush reserver memory every track load. */
+  next_track:
+    if (codec_init(rb)) {
+        return CODEC_ERROR;
+    }
+
+
+    /* Create a decoder instance */
+
+    callbacks.read_func=read_handler;
+    callbacks.seek_func=seek_handler;
+    callbacks.tell_func=tell_handler;
+    callbacks.close_func=close_handler;
+
+    error=ov_open_callbacks(rb,&vf,NULL,0,callbacks);
+    
+    vi=ov_info(&vf,-1);
+
+    if (vi==NULL) {
+        // rb->splash(HZ*2, true, "Vorbis Error");
+        return CODEC_ERROR;
+    }
+  
+    eof=0;
+    while (!eof) {
+        /* Read host-endian signed 16 bit PCM samples */
+        n=ov_read(&vf,pcmbuf,sizeof(pcmbuf),&current_section);
+
+        if (n==0) {
+            eof=1;
+        }
+        else if (n < 0) {
+            DEBUGF("Error decoding frame\n");
+        } else {
+            rb->yield();
+            if (rb->stop_codec || rb->reload_codec)
+                break ;
+        
+            rb->yield();
+            while (!rb->audiobuffer_insert(pcmbuf, n))
+                rb->yield();
+
+            rb->set_elapsed(ov_time_tell(&vf));
+
+#if BYTE_ORDER == BIG_ENDIAN
+            for (i=0;i<n;i+=2) { 
+                x=pcmbuf[i]; pcmbuf[i]=pcmbuf[i+1]; pcmbuf[i+1]=x;
+            }
+#endif
+        }
+    }
+  
+    if (rb->request_next_track())
+        goto next_track;
+    
+    return CODEC_OK;
+}
+
diff --git a/apps/codecs/wav.c b/apps/codecs/wav.c
new file mode 100644
index 0000000..d750b64
--- /dev/null
+++ b/apps/codecs/wav.c
@@ -0,0 +1,136 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 Dave Chapman
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+#include "playback.h"
+#include "lib/codeclib.h"
+
+#define BYTESWAP(x) (((x>>8) & 0xff) | ((x<<8) & 0xff00))
+
+/* Number of bytes to process in one iteration */
+#define WAV_CHUNK_SIZE (1024*4)
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    struct codec_api* rb = api;
+    struct codec_api* ci = api;
+  unsigned long samplerate,numbytes,totalsamples,samplesdone,nsamples;
+  int channels,bytespersample,bitspersample;
+  unsigned int i;
+  size_t n;
+  int endofstream;
+  unsigned char* header;
+  unsigned short* wavbuf;
+
+  /* Generic codec initialisation */
+  TEST_CODEC_API(api);
+
+  /* if you are using a global api pointer, don't forget to copy it!
+     otherwise you will get lovely "I04: IllInstr" errors... :-) */
+  rb = api;
+
+#ifndef SIMULATOR
+  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
+  ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
+  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
+
+  next_track:
+
+  if (codec_init(api)) {
+    return CODEC_ERROR;
+  }
+
+  /* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
+
+  header=ci->request_buffer(&n,44);
+  if (n!=44) {
+    return CODEC_ERROR;
+  }
+  if ((memcmp(header,"RIFF",4)!=0) || (memcmp(&header[8],"WAVEfmt",7)!=0)) {
+    return CODEC_ERROR;
+  }
+
+  samplerate=header[24]|(header[25]<<8)|(header[26]<<16)|(header[27]<<24);
+  bitspersample=header[34];
+  channels=header[22];
+  bytespersample=((bitspersample/8)*channels);
+  numbytes=(header[40]|(header[41]<<8)|(header[42]<<16)|(header[43]<<24));
+  totalsamples=numbytes/bytespersample;
+
+  if ((bitspersample!=16) || (channels != 2)) {
+    return CODEC_ERROR;
+  }
+
+  ci->advance_buffer(44);
+
+  /* The main decoder loop */
+
+  samplesdone=0;
+  ci->set_elapsed(0);
+  endofstream=0;
+  while (!endofstream) {
+    if (ci->stop_codec || ci->reload_codec) {
+      break;
+    }
+
+    wavbuf=ci->request_buffer(&n,WAV_CHUNK_SIZE);
+
+    if (n==0) break; /* End of stream */
+
+    nsamples=(n/bytespersample);
+
+    /* WAV files can contain extra data at the end - so we can't just
+       process until the end of the file */
+
+    if (samplesdone+nsamples > totalsamples) {
+      nsamples=(totalsamples-samplesdone);
+      n=nsamples*bytespersample;
+      endofstream=1;
+    }
+
+    /* Byte-swap data */
+    for (i=0;i<n/2;i++) {
+      wavbuf[i]=BYTESWAP(wavbuf[i]);
+    }
+
+    samplesdone+=nsamples;
+    ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
+   
+    rb->yield();
+    while (!ci->audiobuffer_insert((unsigned char*)wavbuf, n))
+      rb->yield();
+
+    ci->advance_buffer(n);
+  }
+
+  if (ci->request_next_track())
+    goto next_track;
+
+  return CODEC_OK;
+}
diff --git a/apps/codecs/wavpack.c b/apps/codecs/wavpack.c
new file mode 100644
index 0000000..e18dac6
--- /dev/null
+++ b/apps/codecs/wavpack.c
@@ -0,0 +1,185 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
+ *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
+ *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
+ *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
+ *                     \/            \/     \/    \/            \/
+ * $Id$
+ *
+ * Copyright (C) 2005 David Bryant
+ *
+ * All files in this archive are subject to the GNU General Public License.
+ * See the file COPYING in the source tree root for full license agreement.
+ *
+ * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
+ * KIND, either express or implied.
+ *
+ ****************************************************************************/
+
+#include "codec.h"
+
+#include <codecs/libwavpack/wavpack.h>
+#include "playback.h"
+#include "lib/codeclib.h"
+
+static struct codec_api *rb;
+static struct codec_api *ci;
+
+#define BUFFER_SIZE 4096
+
+static long temp_buffer [BUFFER_SIZE] IDATA_ATTR;
+
+static long read_callback (void *buffer, long bytes)
+{
+    return ci->read_filebuf (buffer, bytes);
+}
+
+#ifndef SIMULATOR
+extern char iramcopy[];
+extern char iramstart[];
+extern char iramend[];
+#endif
+
+/* this is the codec entry point */
+enum codec_status codec_start(struct codec_api* api, void* parm)
+{
+    WavpackContext *wpc;
+    char error [80];
+    int bps, nchans;
+
+    /* Generic codec initialisation */
+    TEST_CODEC_API(api);
+
+    rb = api;
+    ci = api;
+
+#ifndef SIMULATOR
+    rb->memcpy(iramstart, iramcopy, iramend-iramstart);
+#endif
+
+    ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*10));
+    ci->configure(CODEC_SET_FILEBUF_WATERMARK, (int *)(1024*512));
+    ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
+
+    next_track:
+
+    if (codec_init(api))
+        return CODEC_ERROR;
+
+    /* Create a decoder instance */
+
+    wpc = WavpackOpenFileInput (read_callback, error);
+
+    if (!wpc)
+        return CODEC_ERROR;
+
+    bps = WavpackGetBytesPerSample (wpc);
+    nchans = WavpackGetReducedChannels (wpc);
+
+    ci->set_elapsed (0);
+
+    /* The main decoder loop */
+
+    while (1) {
+        long nsamples;  
+
+        if (ci->seek_time && ci->taginfo_ready && ci->id3->length) {
+            int curpos_ms = (WavpackGetSampleIndex (wpc) + 220) / 441 * 10;
+            int n, d, skip;
+
+            if (ci->seek_time > curpos_ms) {
+                n = ci->seek_time - curpos_ms;
+                d = ci->id3->length - curpos_ms;
+                skip = (int)((long long)(ci->filesize - ci->curpos) * n / d);
+                ci->seek_buffer (ci->curpos + skip);
+            }
+            else {
+                n = curpos_ms - ci->seek_time;
+                d = curpos_ms;
+                skip = (int)((long long) ci->curpos * n / d);
+                ci->seek_buffer (ci->curpos - skip);
+            }
+
+            wpc = WavpackOpenFileInput (read_callback, error);
+            ci->seek_time = 0;
+
+            if (!wpc)
+                break;
+
+            ci->set_elapsed ((int)((long long) WavpackGetSampleIndex (wpc) * 1000 / 44100));
+            rb->yield ();
+        }
+
+        nsamples = WavpackUnpackSamples (wpc, temp_buffer, BUFFER_SIZE / 2);  
+
+        if (!nsamples || ci->stop_codec || ci->reload_codec)
+            break;
+
+        /* convert mono to stereo here, in place */
+
+        if (nchans == 1) {
+            long *dst = temp_buffer + (nsamples * 2);
+            long *src = temp_buffer + nsamples;
+            long count = nsamples;
+
+            while (count--) {
+                *--dst = *--src;
+                *--dst = *src;
+                if (!(count & 0x7f))
+                    rb->yield ();
+            }
+        }
+
+        if (bps == 1) {
+            short *dst = (short *) temp_buffer;
+            long *src = temp_buffer;
+            long count = nsamples;
+
+            while (count--) {
+                *dst++ = *src++ << 8;
+                *dst++ = *src++ << 8;
+                if (!(count & 0x7f))
+                    rb->yield ();
+            }
+        }
+        else if (bps == 2) {
+            short *dst = (short *) temp_buffer;
+            long *src = temp_buffer;
+            long count = nsamples;
+
+            while (count--) {
+                *dst++ = *src++;
+                *dst++ = *src++;
+                if (!(count & 0x7f))
+                    rb->yield ();
+            }
+        }
+        else {
+            short *dst = (short *) temp_buffer;
+            int shift = (bps - 2) * 8;
+            long *src = temp_buffer;
+            long count = nsamples;
+
+            while (count--) {
+                *dst++ = *src++ >> shift;
+                *dst++ = *src++ >> shift;
+                if (!(count & 0x7f))
+                    rb->yield ();
+            }
+        }
+
+        if (ci->stop_codec || ci->reload_codec)
+            break;
+
+        while (!ci->audiobuffer_insert ((char *) temp_buffer, nsamples * 4))
+            rb->yield ();
+
+        ci->set_elapsed ((WavpackGetSampleIndex (wpc) + 220) / 441 * 10);
+    }
+
+    if (ci->request_next_track())
+        goto next_track;
+
+    return CODEC_OK;
+}