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

15 files changed, 5 insertions, 2119 deletions

diff --git a/apps/plugins/CODECS b/apps/plugins/CODECS
deleted file mode 100644
index ea921a6..0000000
--- a/apps/plugins/CODECS
+++ /dev/null
@@ -1,7 +0,0 @@
-codecvorbis.elf
-codecmpa.elf
-codecflac.elf
-codecwav.elf
-codeca52.elf
-codecmpc.elf
-codecwavpack.elf
diff --git a/apps/plugins/Makefile b/apps/plugins/Makefile
index 44e055d..fbae027 100644
--- a/apps/plugins/Makefile
+++ b/apps/plugins/Makefile
@@ -22,7 +22,6 @@ endif
 
 LDS := plugin.lds
 LINKFILE := $(OBJDIR)/pluginlink.lds
-LINKCODEC := $(OBJDIR)/codeclink.lds
 DEPFILE = $(OBJDIR)/dep-plugins
 
 # This sets up 'SRC' based on the files mentioned in SOURCES
@@ -55,13 +54,7 @@ ifndef SIMVER
 $(OBJDIR)/%.elf: $(OBJDIR)/%.o $(LINKFILE) $(LINKCODEC) $(BUILDDIR)/libplugin.a
 	$(SILENT)(file=`basename $@`; \
 	echo "LD $$file"; \
-	match=`grep $$file CODECS`; \
-	if test -z "$$match"; then \
-	  LINKWITH=$(LINKFILE); \
-	else \
-	  LINKWITH=$(LINKCODEC); \
-	fi; \
-	$(CC) $(GCCOPTS) -O -nostdlib -o $@ $< -L$(BUILDDIR) $(CODECLIBS) -lplugin -lgcc -T$$LINKWITH -Wl,-Map,$(OBJDIR)/$*.map)
+	$(CC) $(GCCOPTS) -O -nostdlib -o $@ $< -L$(BUILDDIR) $(CODECLIBS) -lplugin -lgcc -T$(LINKFILE) -Wl,-Map,$(OBJDIR)/$*.map)
 
 $(OBJDIR)/%.rock : $(OBJDIR)/%.elf
 	@echo "OBJCOPY "`basename $@`
@@ -108,7 +101,7 @@ endif # end of simulator section
 include $(TOOLSDIR)/make.inc
 
 $(BUILDDIR)/libplugin.a:
-	@echo "MAKE in lib"
+	@echo "MAKE in plugin/lib"
 	@mkdir -p $(OBJDIR)/lib
 	@$(MAKE) -C lib OBJDIR=$(OBJDIR)/lib
 
@@ -116,10 +109,6 @@ $(LINKFILE): $(LDS)
 	@echo "build $@"
 	@cat $< | $(CC) -DMEMORYSIZE=$(MEMORYSIZE) $(INCLUDES) $(TARGET) $(DEFINES) -E -P - >$@
 
-$(LINKCODEC): $(LDS)
-	@echo "build $@"
-	@cat $< | $(CC) -DMEMORYSIZE=$(MEMORYSIZE) -DCODEC $(INCLUDES) $(TARGET) $(DEFINES) -E -P - >$@
-
 $(SUBDIRS):
 	@echo "MAKE in $@"
 	@mkdir -p $(OBJDIR)/$@
diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index 34f014d..933625a 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -67,22 +67,15 @@ alpine_cdc.c
 #endif
 
 #if CONFIG_HWCODEC == MASNONE /* software codec platforms */
+#if 0
 mpa2wav.c
 a52towav.c
 flac2wav.c
 vorbis2wav.c
-#ifdef IRIVER_H100
-codecvorbis.c
-codecmpa.c
-codecflac.c
-codecwav.c
-codeca52.c
-codecmpc.c
-codecwavpack.c
-#endif
 wv2wav.c
 mpc2wav.c
 midi2wav.c
+#endif
 iriverify.c
 #else
 splitedit.c
diff --git a/apps/plugins/codeca52.c b/apps/plugins/codeca52.c
deleted file mode 100644
index ced5ced..0000000
--- a/apps/plugins/codeca52.c
+++ /dev/null
@@ -1,210 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.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 plugin_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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  size_t n;
-  unsigned char* filebuf;
-
-  /* Generic plugin initialisation */
-  TEST_PLUGIN_API(api);
-
-  rb = api;
-  ci = (struct codec_api*)parm;
-
-#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, ci)) {
-  return PLUGIN_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 PLUGIN_OK;
-}
diff --git a/apps/plugins/codecflac.c b/apps/plugins/codecflac.c
deleted file mode 100644
index 1cc95b3..0000000
--- a/apps/plugins/codecflac.c
+++ /dev/null
@@ -1,248 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.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 plugin_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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct codec_api* ci = (struct codec_api*)parm;
-  FLAC__SeekableStreamDecoder* flacDecoder;
-
-  /* Generic plugin initialisation */
-  TEST_PLUGIN_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, ci)) {
-    return PLUGIN_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 PLUGIN_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 PLUGIN_OK;
-}
diff --git a/apps/plugins/codecmpa.c b/apps/plugins/codecmpa.c
deleted file mode 100644
index 768c3fe..0000000
--- a/apps/plugins/codecmpa.c
+++ /dev/null
@@ -1,520 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.h"
-
-#include <codecs/libmad/mad.h>
-
-#include "playback.h"
-#include "mp3data.h"
-#include "lib/codeclib.h"
-
-static struct plugin_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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct codec_api *ci = (struct codec_api *)parm;
-  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 plugin inititialisation */
-
-  TEST_PLUGIN_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, ci)) {
-    return PLUGIN_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 PLUGIN_OK;
-}
diff --git a/apps/plugins/codecmpc.c b/apps/plugins/codecmpc.c
deleted file mode 100644
index 19eb518..0000000
--- a/apps/plugins/codecmpc.c
+++ /dev/null
@@ -1,214 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.h"
-#include "playback.h"
-#include "lib/codeclib.h"
-#include <codecs/libmusepack/musepack.h>
-
-static struct plugin_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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct codec_api* ci = (struct codec_api*)parm;
-  unsigned short Sample;
-  unsigned long samplesdone;
-  unsigned long frequency;
-  unsigned status = 1;
-  unsigned int i;
-  mpc_reader reader;
-
-  /* Generic plugin inititialisation */
-
-  TEST_PLUGIN_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, ci)) {
-    return PLUGIN_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 PLUGIN_ERROR;  
-  }
-  frequency=info.sample_freq;
-
-  /* instantiate a decoder with our file reader */
-  mpc_decoder_setup(&decoder, &reader);
-  if (!mpc_decoder_initialize(&decoder, &info)) {
-    return PLUGIN_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 PLUGIN_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 PLUGIN_OK;
-}
-
diff --git a/apps/plugins/codecvorbis.c b/apps/plugins/codecvorbis.c
deleted file mode 100644
index 8e88471..0000000
--- a/apps/plugins/codecvorbis.c
+++ /dev/null
@@ -1,166 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.h"
-
-#include <codecs/Tremor/ivorbisfile.h>
-
-#include "playback.h"
-#include "lib/codeclib.h"
-
-static struct plugin_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) {
-    struct codec_api *p = (struct codec_api *) datasource;
-
-    return p->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) {
-    struct codec_api *p = (struct codec_api *) datasource;
-  return p->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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct codec_api *ci = (struct codec_api *)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_PLUGIN_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
-    
-  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
-  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*64));
-    
-  /* We need to flush reserver memory every track load. */
-  next_track:
-  if (codec_init(api, ci)) {
-    return PLUGIN_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(ci,&vf,NULL,0,callbacks);
-    
-  vi=ov_info(&vf,-1);
-
-  if (vi==NULL) {
-    // rb->splash(HZ*2, true, "Vorbis Error");
-    return PLUGIN_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 (ci->stop_codec || ci->reload_codec)
-        break ;
-        
-      rb->yield();
-      while (!ci->audiobuffer_insert(pcmbuf, n))
-        rb->yield();
-
-      ci->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 (ci->request_next_track())
-    goto next_track;
-    
-  return PLUGIN_OK;
-}
-
diff --git a/apps/plugins/codecwav.c b/apps/plugins/codecwav.c
deleted file mode 100644
index 230b316..0000000
--- a/apps/plugins/codecwav.c
+++ /dev/null
@@ -1,136 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct plugin_api* rb = (struct plugin_api*)api;
-  struct codec_api* ci = (struct codec_api*)parm;
-  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 plugin initialisation */
-  TEST_PLUGIN_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, ci)) {
-    return PLUGIN_ERROR;
-  }
-
-  /* FIX: Correctly parse WAV header - we assume canonical 44-byte header */
-
-  header=ci->request_buffer(&n,44);
-  if (n!=44) {
-    return PLUGIN_ERROR;
-  }
-  if ((memcmp(header,"RIFF",4)!=0) || (memcmp(&header[8],"WAVEfmt",7)!=0)) {
-    return PLUGIN_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 PLUGIN_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 PLUGIN_OK;
-}
diff --git a/apps/plugins/codecwavpack.c b/apps/plugins/codecwavpack.c
deleted file mode 100644
index 1e56dbd..0000000
--- a/apps/plugins/codecwavpack.c
+++ /dev/null
@@ -1,185 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.h"
-
-#include <codecs/libwavpack/wavpack.h>
-#include "playback.h"
-#include "lib/codeclib.h"
-
-static struct plugin_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 plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-    WavpackContext *wpc;
-    char error [80];
-    int bps, nchans;
-
-    /* Generic plugin initialisation */
-    TEST_PLUGIN_API(api);
-
-    rb = api;
-    ci = (struct codec_api*) parm;
-
-#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, ci))
-        return PLUGIN_ERROR;
-
-    /* Create a decoder instance */
-
-    wpc = WavpackOpenFileInput (read_callback, error);
-
-    if (!wpc)
-        return PLUGIN_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 PLUGIN_OK;
-}
diff --git a/apps/plugins/lib/SOURCES b/apps/plugins/lib/SOURCES
index f8ed798..0914f10 100644
--- a/apps/plugins/lib/SOURCES
+++ b/apps/plugins/lib/SOURCES
@@ -34,9 +34,8 @@ gray_verline.c
 #ifdef HAVE_LCD_CHARCELLS
 playergfx.c
 #endif
+#if 0
 #if CONFIG_HWCODEC == MASNONE /* software codec platforms */
 xxx2wav.c
-#ifdef IRIVER_H100
-codeclib.c
 #endif
 #endif
diff --git a/apps/plugins/lib/codeclib.c b/apps/plugins/lib/codeclib.c
deleted file mode 100644
index 9d9cc0b..0000000
--- a/apps/plugins/lib/codeclib.c
+++ /dev/null
@@ -1,37 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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  */
-
-#include "plugin.h"
-#include "playback.h"
-#include "codeclib.h"
-#include "xxx2wav.h"
-
-struct plugin_api* local_rb;
-
-int codec_init(struct plugin_api* rb, struct codec_api* ci) {
-    local_rb = rb;
-    
-    xxx2wav_set_api(rb);
-    mem_ptr = 0;
-    mallocbuf = (unsigned char *)ci->get_codec_memory((size_t *)&bufsize);
-  
-    return 0;
-}
diff --git a/apps/plugins/lib/codeclib.h b/apps/plugins/lib/codeclib.h
deleted file mode 100644
index 876e69b..0000000
--- a/apps/plugins/lib/codeclib.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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
-
-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 plugin_api* rb, struct codec_api* ci);
-
diff --git a/apps/plugins/lib/xxx2wav.c b/apps/plugins/lib/xxx2wav.c
deleted file mode 100644
index 8eadfb8..0000000
--- a/apps/plugins/lib/xxx2wav.c
+++ /dev/null
@@ -1,259 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 "plugin.h"
-#include "xxx2wav.h"
-
-static struct plugin_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();
-  }
-}
-
-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
-                             };
-
-
-void xxx2wav_set_api(struct plugin_api* rb)
-{
-    local_rb = rb;
-}
-
-int local_init(char* infilename, char* outfilename, file_info_struct* file_info, struct plugin_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 /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/lib/xxx2wav.h b/apps/plugins/lib/xxx2wav.h
deleted file mode 100644
index 7e1b942..0000000
--- a/apps/plugins/lib/xxx2wav.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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 plugin_api* rb);
-void close_wav(file_info_struct* file_info);
-void xxx2wav_set_api(struct plugin_api* rb);