removed old codec leftovers

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6919 a1c6a512-1295-4272-9138-f99709370657

9 files changed, 0 insertions, 1574 deletions

diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index 933625aac2..76d26e7e59 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -67,15 +67,6 @@ alpine_cdc.c
 #endif
 
 #if CONFIG_HWCODEC == MASNONE /* software codec platforms */
-#if 0
-mpa2wav.c
-a52towav.c
-flac2wav.c
-vorbis2wav.c
-wv2wav.c
-mpc2wav.c
-midi2wav.c
-#endif
 iriverify.c
 #else
 splitedit.c
diff --git a/apps/plugins/a52towav.c b/apps/plugins/a52towav.c
deleted file mode 100644
index f6769abd2b..0000000000
--- a/apps/plugins/a52towav.c
+++ /dev/null
@@ -1,217 +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"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include <inttypes.h>  /* Needed by a52.h */
-
-#include <codecs/liba52/config-a52.h>
-#include <codecs/liba52/a52.h>
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-
-static struct plugin_api* rb;
-
-#ifdef WORDS_BIGENDIAN
-#define LE_S16(x) ( (uint16_t) ( ((uint16_t)(x) >> 8) | ((uint16_t)(x) << 8) ) )
-#else
-#define LE_S16(x) (x)
-#endif
-
-
-static float gain = 1;
-static a52_state_t * state;
-
-static inline int16_t convert (int32_t i)
-{
-    i >>= 15;
-    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
-}
-
-void ao_play(file_info_struct* file_info,sample_t* samples,int flags) {
-  int i;
-  static int16_t int16_samples[256*2];
-
-  flags &= A52_CHANNEL_MASK | A52_LFE;
-
-  if (flags==A52_STEREO) {
-    for (i = 0; i < 256; i++) {
-        int16_samples[2*i] = LE_S16(convert (samples[i]));
-        int16_samples[2*i+1] = LE_S16(convert (samples[i+256]));
-    }
-  } else {
-    DEBUGF("ERROR: unsupported format: %d\n",flags);
-  }
-
-  /* FIX: Buffer the disk write to write larger amounts at one */
-  i=rb->write(file_info->outfile,int16_samples,256*2*2);
-}
-
-
-void a52_decode_data (file_info_struct* file_info, uint8_t * start, uint8_t * end)
-{
-    static uint8_t buf[3840];
-    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;
-
-                // 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;
-                    }
-		    ao_play (file_info, a52_samples (state),flags);
-                    file_info->current_sample+=256;
-		}
-		bufptr = buf;
-		bufpos = buf + 7;
-		continue;
-	    error:
-		DEBUGF("error\n");
-		bufptr = buf;
-		bufpos = buf + 7;
-	    }
-	}
-    }
-}
-
-
-#define BUFFER_SIZE 4096
-
-#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* file)
-{
-  file_info_struct file_info;
-
-  /* Generic plugin initialisation */
-
-  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 (local_init(file,"/ac3test.wav",&file_info,api)) {
-    return PLUGIN_ERROR;
-  }
-
-  /* Intialise the A52 decoder and check for success */
-  state = a52_init (0); // Parameter is "accel"
-
-  if (state == NULL) {
-    rb->splash(HZ*2, true, "a52_init failed");
-    return PLUGIN_ERROR;
-  }
-
-  /* The main decoding loop */
-
-  file_info.start_tick=*(rb->current_tick);
-  rb->button_clear_queue();
-
-  while (file_info.curpos < file_info.filesize) {
-
-    if ((file_info.curpos+BUFFER_SIZE) < file_info.filesize) {   
-      a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.curpos+BUFFER_SIZE]);
-      file_info.curpos+=BUFFER_SIZE;
-    } else {
-      a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.filesize-1]);
-      file_info.curpos=file_info.filesize;
-    }
-
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) {
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }
-  close_wav(&file_info);
-
-  /* Cleanly close and exit */
-
-//NOT NEEDED:  a52_free (state);
-
-  rb->splash(HZ*2, true, "FINISHED!");
-  return PLUGIN_OK;
-}
-#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/flac2wav.c b/apps/plugins/flac2wav.c
deleted file mode 100644
index 84b5ed15b2..0000000000
--- a/apps/plugins/flac2wav.c
+++ /dev/null
@@ -1,237 +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"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include <codecs/libFLAC/include/FLAC/seekable_stream_decoder.h>
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-
-#define FLAC_MAX_SUPPORTED_BLOCKSIZE 4608
-#define FLAC_MAX_SUPPORTED_CHANNELS  2
-
-static struct plugin_api* rb;
-
-/* 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)
-{   (void)dec;
-
-    file_info_struct *p = (file_info_struct *) data;
-
-    if (p->curpos >= p->filesize) {
-      return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
-    }
-
-    rb->memcpy(buffer,&filebuf[p->curpos],*bytes);
-    p->curpos+=*bytes;
-  
-    return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
-}
-
-/* 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)
-{
-    unsigned int c_samp, c_chan, d_samp;
-    file_info_struct *p = (file_info_struct *) data;
-    uint32_t data_size = frame->header.blocksize * frame->header.channels * (p->bitspersample / 8);
-    uint32_t samples = frame->header.blocksize;
-
-    // FIXME: This should not be on the stack!
-    static unsigned char ldb[FLAC_MAX_SUPPORTED_BLOCKSIZE*FLAC_MAX_SUPPORTED_CHANNELS*2];
-
-    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;
-    (void)data_size;
-    for(c_samp = d_samp = 0; c_samp < samples; c_samp++) {
-	for(c_chan = 0; c_chan < frame->header.channels; c_chan++, d_samp++) {
-	    ldb[d_samp*2] = buf[c_chan][c_samp]&0xff;
-	    ldb[(d_samp*2)+1] = (buf[c_chan][c_samp]&0xff00)>>8;
-	}
-    } 
-
-    rb->write(p->outfile,ldb,data_size);
-
-    p->current_sample += samples;
-
-    return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
-}
-
-void flac_metadata_handler(const FLAC__SeekableStreamDecoder *dec, 
-                           const FLAC__StreamMetadata *meta, void *data)
-{
-    file_info_struct *p = (file_info_struct *) data;
-    (void)dec;
-
-    if(meta->type == FLAC__METADATA_TYPE_STREAMINFO) {
-	p->bitspersample = meta->data.stream_info.bits_per_sample;
-	p->samplerate = meta->data.stream_info.sample_rate;
-	p->channels = meta->data.stream_info.channels;
-//	FLAC__ASSERT(meta->data.stream_info.total_samples < 0x100000000); /* we can handle < 4 gigasamples */
-	p->total_samples = (unsigned) 
-	    (meta->data.stream_info.total_samples & 0xffffffff);
-	p->current_sample = 0;
-    }
-}
-
-
-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;
-  file_info_struct *p = (file_info_struct *) client_data;
-  rb->lseek(p->infile,SEEK_SET,absolute_byte_offset);
-  return(FLAC__SEEKABLE_STREAM_DECODER_SEEK_STATUS_OK);
-}
-
-FLAC__SeekableStreamDecoderTellStatus flac_tell_handler (const FLAC__SeekableStreamDecoder *decoder, 
-                                                         FLAC__uint64 *absolute_byte_offset, void *client_data)
-{ 
-  file_info_struct *p = (file_info_struct *) client_data;
-
-  (void)decoder;
-  *absolute_byte_offset=rb->lseek(p->infile,SEEK_CUR,0);
-  return(FLAC__SEEKABLE_STREAM_DECODER_TELL_STATUS_OK);
-}
-
-FLAC__SeekableStreamDecoderLengthStatus flac_length_handler (const FLAC__SeekableStreamDecoder *decoder, 
-                                                         FLAC__uint64 *stream_length, void *client_data)
-{
-  file_info_struct *p = (file_info_struct *) client_data;
-
-  (void)decoder;
-  *stream_length=p->filesize;
-  return(FLAC__SEEKABLE_STREAM_DECODER_LENGTH_STATUS_OK);
-}
-
-FLAC__bool flac_eof_handler (const FLAC__SeekableStreamDecoder *decoder, 
-                             void *client_data)
-{
-  file_info_struct *p = (file_info_struct *) client_data;
-
-  (void)decoder;
-  if (p->curpos >= p->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* file)
-{
-  FLAC__SeekableStreamDecoder* flacDecoder;
-  file_info_struct file_info;
-
-  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
-
-  /* This function sets up the buffers and reads the file into RAM */
-
-  if (local_init(file,"/flactest.wav",&file_info,api)) {
-    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,&file_info);
-  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);
-
-  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);
-
-  file_info.frames_decoded=0;
-  file_info.start_tick=*(rb->current_tick);
-  rb->button_clear_queue();
-
-  while (FLAC__seekable_stream_decoder_get_state(flacDecoder)!=2) {
-    FLAC__seekable_stream_decoder_process_single(flacDecoder);
-    file_info.frames_decoded++;
-
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) {
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }
-
-  close_wav(&file_info);
-  rb->splash(HZ*2, true, "FINISHED!");
-
-  /* Flush internal buffers etc */
-//No need for this.  flacResult=FLAC__seekable_stream_decoder_reset(flacDecoder);
-
-  //  audio_close();
-
-  return PLUGIN_OK;
-}
-#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/lib/SOURCES b/apps/plugins/lib/SOURCES
index 396ed062ac..49abf854da 100644
--- a/apps/plugins/lib/SOURCES
+++ b/apps/plugins/lib/SOURCES
@@ -34,8 +34,3 @@ gray_verline.c
 #ifdef HAVE_LCD_CHARCELLS
 playergfx.c
 #endif
-#if 0
-#if CONFIG_HWCODEC == MASNONE /* software codec platforms */
-xxx2wav.c
-#endif
-#endif
diff --git a/apps/plugins/midi2wav.c b/apps/plugins/midi2wav.c
deleted file mode 100644
index 241d56defb..0000000000
--- a/apps/plugins/midi2wav.c
+++ /dev/null
@@ -1,232 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
- *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
- *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
- *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
- *                     \/            \/     \/    \/            \/
- *
- * Copyright (C) 2005 Stepan Moskovchenko
- *
- * 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.
- *
- ****************************************************************************/
-
-#define SAMPLE_RATE 22050
-#define MAX_VOICES 100
-
-
-/* Only define LOCAL_DSP on Simulator or else we're asking for trouble  */
-#if defined(SIMULATOR)
-    /*Enable this to write to the soundcard via a /dsv/dsp symlink in */
-//    #define LOCAL_DSP
-#endif
-
-
-#if defined(LOCAL_DSP)
-/*  This is for writing to the DSP directly from the Simulator */
-#include <stdio.h>
-#include <stdlib.h>
-#include <linux/soundcard.h>
-#include <sys/ioctl.h>
-#endif
-
-#include "../../firmware/export/system.h"
-
-#include "../../plugin.h"
-
-#include "lib/xxx2wav.h"
-
-int numberOfSamples IDATA_ATTR;
-long bpm;
-
-#include "midi/midiutil.c"
-#include "midi/guspat.h"
-#include "midi/guspat.c"
-#include "midi/sequencer.c"
-#include "midi/midifile.c"
-#include "midi/synth.c"
-
-
-
-
-int fd=-1; /* File descriptor where the output is written */
-
-extern long tempo; /* The sequencer keeps track of this */
-
-
-struct plugin_api * rb;
-
-
-
-
-
-enum plugin_status plugin_start(struct plugin_api* api, void* parameter)
-{
-    TEST_PLUGIN_API(api);
-    rb = api;
-    TEST_PLUGIN_API(api);
-    (void)parameter;
-    rb = api;
-
-    if(parameter == NULL)
-    {
-        rb->splash(HZ*2, true, " Play .MID file ");
-        return PLUGIN_OK;
-    }
-
-    rb->splash(HZ, true, parameter);
-    if(midimain(parameter) == -1)
-    {
-        return PLUGIN_ERROR;
-    }
-    rb->splash(HZ*3, true, "FINISHED PLAYING");
-    return PLUGIN_OK;
-}
-
-signed char outputBuffer[3000] IDATA_ATTR; /* signed char.. gonna run out of iram ... ! */
-
-
-int currentSample IDATA_ATTR;
-int outputBufferPosition IDATA_ATTR;
-int outputSampleOne IDATA_ATTR;
-int outputSampleTwo IDATA_ATTR;
-
-
-int midimain(void * filename)
-{
-
-    printf("\nHello.\n");
-
-    rb->splash(HZ/5, true, "LOADING MIDI");
-
-    struct MIDIfile * mf = loadFile(filename);
-
-    rb->splash(HZ/5, true, "LOADING PATCHES");
-    if (initSynth(mf, "/.rockbox/patchset/patchset.cfg", "/.rockbox/patchset/drums.cfg") == -1)
-    {
-        return -1;
-    }
-
-/*
- *  This lets you hear the music through the sound card if you are on Simulator
- *  Make a symlink, archos/dsp.raw and make it point to /dev/dsp or whatever
- *  your sound device is.
- */
-
-#if defined(LOCAL_DSP)
-    fd=rb->open("/dsp.raw", O_WRONLY);
-    int arg, status;
-    int bit, samp, ch;
-
-    arg = 16;          /* sample size */
-    status = ioctl(fd, SOUND_PCM_WRITE_BITS, &arg);
-    status = ioctl(fd, SOUND_PCM_READ_BITS, &arg);
-    bit=arg;
-
-
-    arg = 2;        /* Number of channels, 1=mono */
-    status = ioctl(fd, SOUND_PCM_WRITE_CHANNELS, &arg);
-    status = ioctl(fd, SOUND_PCM_READ_CHANNELS, &arg);
-    ch=arg;
-
-    arg = SAMPLE_RATE; /* Yeah. sampling rate */
-    status = ioctl(fd, SOUND_PCM_WRITE_RATE, &arg);
-    status = ioctl(fd, SOUND_PCM_READ_RATE, &arg);
-    samp=arg;
-#else
-    file_info_struct file_info;
-    file_info.samplerate = SAMPLE_RATE;
-    file_info.infile = fd;
-    file_info.channels = 2;
-    file_info.bitspersample = 16;
-    local_init("/miditest.tmp", "/miditest.wav", &file_info, rb);
-    fd = file_info.outfile;
-#endif
-
-
-    rb->splash(HZ/5, true, "  I hope this works...  ");
-
-
-
-
-    /*
-     * tick() will do one MIDI clock tick. Then, there's a loop here that
-     * will generate the right number of samples per MIDI tick. The whole
-     * MIDI playback is timed in terms of this value.. there are no forced
-     * delays or anything. It just produces enough samples for each tick, and
-     * the playback of these samples is what makes the timings right.
-     *
-     * This seems to work quite well.
-     */
-
-    printf("\nOkay, starting sequencing");
-
-
-    currentSample=0;            /* Sample counting variable */
-    outputBufferPosition = 0;
-
-
-    bpm=mf->div*1000000/tempo;
-    numberOfSamples=SAMPLE_RATE/bpm;
-
-
-
-    /* Tick() will return 0 if there are no more events left to play */
-    while(tick(mf))
-    {
-        /*
-         * Tempo recalculation moved to sequencer.c to be done on a tempo event only
-         *
-         */
-        for(currentSample=0; currentSample<numberOfSamples; currentSample++)
-        {
-
-            synthSample(&outputSampleOne, &outputSampleTwo);
-
-
-            /*
-             * 16-bit audio because, well, it's better
-             * But really because ALSA's OSS emulation sounds extremely
-             * noisy and distorted when in 8-bit mode. I still do not know
-             * why this happens.
-             */
-
-            outputBuffer[outputBufferPosition]=outputSampleOne&0XFF; // Low byte first
-            outputBufferPosition++;
-            outputBuffer[outputBufferPosition]=outputSampleOne>>8;   //High byte second
-            outputBufferPosition++;
-
-            outputBuffer[outputBufferPosition]=outputSampleTwo&0XFF; // Low byte first
-            outputBufferPosition++;
-            outputBuffer[outputBufferPosition]=outputSampleTwo>>8;   //High byte second
-            outputBufferPosition++;
-
-
-            /*
-             * As soon as we produce 2000 bytes of sound,
-             * write it to the sound card. Why 2000? I have
-             * no idea. It's 1 AM and I am dead tired.
-             */
-            if(outputBufferPosition>=2000)
-            {
-                rb->write(fd, outputBuffer, 2000);
-                outputBufferPosition=0;
-            }
-        }
-    }
-
-    printf("\n");
-
-#if !defined(LOCAL_DSP)
-
-    close_wav(&file_info);
-#else
-    rb->close(fd);
-#endif
-    return 0;
-}
diff --git a/apps/plugins/mpa2wav.c b/apps/plugins/mpa2wav.c
deleted file mode 100644
index bf40fa3b81..0000000000
--- a/apps/plugins/mpa2wav.c
+++ /dev/null
@@ -1,269 +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"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include <codecs/libmad/mad.h>
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-
-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;
-}
-
-#define SHRT_MAX 32767
-
-#define INPUT_BUFFER_SIZE	(10*8192)
-#define OUTPUT_BUFFER_SIZE	65536 /* Must be an integer multiple of 4. */
-
-unsigned char InputBuffer[INPUT_BUFFER_SIZE+MAD_BUFFER_GUARD];
-unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
-unsigned char *OutputPtr=OutputBuffer;
-unsigned char *GuardPtr=NULL;
-const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
-
-mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
-unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
-
-#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* file)
-{
-  file_info_struct file_info;
-  int Status=0;
-  unsigned short Sample;
-  int i;
-
-  /* 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 (local_init(file,"/libmadtest.wav",&file_info,api)) {
-    return PLUGIN_ERROR;
-  }
-
-  /* Create a decoder instance */
-
-  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;
-  
-  GuardPtr = &filebuf[file_info.filesize];
-  memset(GuardPtr,0,MAD_BUFFER_GUARD);
-  mad_stream_buffer(&Stream, filebuf,file_info.filesize);
-
-  file_info.curpos=0;
-  file_info.start_tick=*(rb->current_tick);
-
-  rb->button_clear_queue();
-
-  /* This is the decoding loop. */
-  while (file_info.curpos < file_info.filesize &&
-         Stream.this_frame != GuardPtr &&
-      Stream.error != MAD_ERROR_BUFLEN) {
-      file_info.curpos += (int)Stream.next_frame - (int)Stream.this_frame;
-
-    if(mad_frame_decode(&Frame,&Stream))
-    {
-      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)
-          continue;
-        else
-        {
-          rb->splash(HZ*1, true, "Recoverable...!");
-          //fprintf(stderr,"%s: unrecoverable frame level error.\n",ProgName);
-          Status=1;
-          break;
-        }
-    }
-
-    /* We assume all frames have same samplerate as the first */
-    if(file_info.frames_decoded==0) {
-      file_info.samplerate=Frame.header.samplerate;
-    }
-
-    file_info.frames_decoded++;
-
-    /* ?? 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);
-
-    /* Convert MAD's numbers to an array of 16-bit LE signed integers */
-    for(i=0;i<Synth.pcm.length;i++)
-    {
-      /* Left channel */
-      Sample=scale(Synth.pcm.samples[0][i],&d0);
-      *(OutputPtr++)=Sample&0xff;
-      *(OutputPtr++)=Sample>>8;
-
-      /* 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(Synth.pcm.samples[1][i],&d1);
-      *(OutputPtr++)=Sample&0xff;
-      *(OutputPtr++)=Sample>>8;
-
-      /* Flush the buffer if it is full. */
-      if(OutputPtr==OutputBufferEnd)
-      {
-        rb->write(file_info.outfile,OutputBuffer,OUTPUT_BUFFER_SIZE);
-        OutputPtr=OutputBuffer;
-      }
-    }
-
-    file_info.current_sample+=Synth.pcm.length;
-
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) { 
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }
-
-  close_wav(&file_info);
-  rb->splash(HZ*2, true, "FINISHED!");
-
-  return PLUGIN_OK;
-}
-#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/mpc2wav.c b/apps/plugins/mpc2wav.c
deleted file mode 100644
index b1478bac31..0000000000
--- a/apps/plugins/mpc2wav.c
+++ /dev/null
@@ -1,208 +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.
- *
- ****************************************************************************/
-
-/* This is a lovely mishmash of sample.c from libmusepack and mpa2wav.c,
- * but happens to work, so no whining!
- */
-
-#include "plugin.h"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include <codecs/libmusepack/musepack.h>
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-
-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)
-{
-  file_info_struct *f = (file_info_struct *)data;
-  mpc_int32_t num = f->filesize - f->curpos;
-  if (num > size)
-    num = size;
-  rb->memcpy(ptr, filebuf + f->curpos, num); 
-  f->curpos += num;
-  return num;
-}
-
-bool
-seek_impl(void *data, mpc_int32_t offset)
-{  
-  file_info_struct *f = (file_info_struct *)data;
-  if (offset > f->filesize) {
-    return 0;
-  } else {
-    f->curpos = offset;
-    return 1;
-  }
-}
-
-mpc_int32_t
-tell_impl(void *data)
-{
-  file_info_struct *f = (file_info_struct *)data;
-  return f->curpos;
-}
-
-mpc_int32_t
-get_size_impl(void *data)
-{
-  file_info_struct *f = (file_info_struct *)data;
-  return f->filesize;
-}
-
-bool
-canseek_impl(void *data)
-{
-    (void)data;
-    return true;
-}
-
-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];
-MPC_SAMPLE_FORMAT sample_buffer[MPC_DECODER_BUFFER_LENGTH];
-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* file)
-{
-  file_info_struct file_info;
-  unsigned short Sample;
-  unsigned status = 1;
-  unsigned int i;
-  mpc_reader reader;
-
-  /* Generic plugin inititialisation */
-
-  TEST_PLUGIN_API(api);
-  rb = api;
-
-#ifdef USE_IRAM
-  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
-#endif
-
-  reader.read = read_impl;
-  reader.seek = seek_impl;
-  reader.tell = tell_impl;
-  reader.get_size = get_size_impl;
-  reader.canseek = canseek_impl;
-  reader.data = &file_info;
-
-  /* This function sets up the buffers and reads the file into RAM */
-
-  if (local_init(file, "/libmusepacktest.wav", &file_info, api)) {
-    return PLUGIN_ERROR;
-  }
-  
-  /* read file's streaminfo data */
-  mpc_streaminfo info;
-  mpc_streaminfo_init(&info);
-  if (mpc_streaminfo_read(&info, &reader) != ERROR_CODE_OK) {
-    rb->splash(HZ, true, "Not an MPC file.");
-    return PLUGIN_ERROR;  
-  }
-  file_info.samplerate=info.sample_freq;
-  /* instantiate a decoder with our file reader */
-  mpc_decoder_setup(&decoder, &reader);
-  if (!mpc_decoder_initialize(&decoder, &info)) {
-    rb->splash(HZ, true, "Error in init.");
-    return PLUGIN_ERROR;
-  }
-  file_info.frames_decoded = 0;
-  file_info.start_tick = *(rb->current_tick);
-
-  rb->button_clear_queue();
-
-  /* This is the decoding loop. */
-  while (status != 0) {
-    status = mpc_decoder_decode(&decoder, sample_buffer, 0, 0);
-    if (status == (unsigned)(-1)) {
-        //decode error
-        rb->splash(HZ, true, "Error decoding file.");
-        break;
-    }
-    else                    //status>0
-    {
-      file_info.current_sample += status;
-      file_info.frames_decoded++;
-      /* Convert musepack's numbers to an array of 16-bit LE signed integers */
-#if 1 /* uncomment to time without byte swapping and disk writing */
-      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&0xff;
-        *(OutputPtr++)=Sample>>8;
-
-        /* 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&0xff;
-        *(OutputPtr++)=Sample>>8;
-
-        /* Flush the buffer if it is full. */
-        if(OutputPtr==OutputBufferEnd)
-        {
-          rb->write(file_info.outfile,OutputBuffer,OUTPUT_BUFFER_SIZE);
-          OutputPtr=OutputBuffer;
-        }
-      }
-#endif
-
-    }
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) { 
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }
-  close_wav(&file_info);
-  rb->splash(HZ*2, true, "FINISHED!");
-
-  return PLUGIN_OK;
-}
-#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/vorbis2wav.c b/apps/plugins/vorbis2wav.c
deleted file mode 100644
index 01815ab1ca..0000000000
--- a/apps/plugins/vorbis2wav.c
+++ /dev/null
@@ -1,180 +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"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include <codecs/Tremor/ivorbisfile.h>
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-
-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) {
-    size_t len;
-    file_info_struct *p = (file_info_struct *) datasource;
-
-    if (p->curpos >= p->filesize) {
-      return 0; /* EOF */
-    }
-
-    len=nmemb*size;
-    if ((long)(p->curpos+len) > (long)p->filesize) { len=p->filesize-p->curpos; }
-
-    rb->memcpy(ptr,&filebuf[p->curpos],len);
-    p->curpos+=len;
-
-    return(len);  
-}
-
-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) {
-  file_info_struct *p = (file_info_struct *) 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* file)
-{
-  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
-
-  file_info_struct file_info;
-
-  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
-    
-  /* This function sets up the buffers and reads the file into RAM */
-
-  if (local_init(file,"/vorbistest.wav",&file_info,api)) {
-    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;
-
-  file_info.frames_decoded=0;
-  file_info.start_tick=*(rb->current_tick);
-  rb->button_clear_queue();
-
-  error=ov_open_callbacks(&file_info,&vf,NULL,0,callbacks);
-
-  vi=ov_info(&vf,-1);
-
-  if (vi==NULL) {
-    rb->splash(HZ*2, true, "Error");
-  }
-  file_info.samplerate=vi->rate;
-  
-  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 {
-      file_info.frames_decoded++;
-#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
-      rb->write(file_info.outfile,pcmbuf,n);
-      file_info.current_sample+=(n/4);
-    }
-
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) {
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }
-
-  close_wav(&file_info);
-  rb->splash(HZ*2, true, "FINISHED!");
-  return PLUGIN_OK;
-}
-#endif /* CONFIG_HWCODEC == MASNONE */
diff --git a/apps/plugins/wv2wav.c b/apps/plugins/wv2wav.c
deleted file mode 100644
index 909a0c3c63..0000000000
--- a/apps/plugins/wv2wav.c
+++ /dev/null
@@ -1,217 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
- *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
- *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
- *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
- *                     \/            \/     \/    \/            \/
- * $Id$
- *
- * Copyright (C) 2005 Christian Gmeiner
- *
- * 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"
-
-#if (CONFIG_HWCODEC == MASNONE)
-/* software codec platforms */
-
-#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
-#include <codecs/libwavpack/wavpack.h>
-
-#define BUFFER_SIZE 4096
-
-static struct plugin_api* rb;
-static file_info_struct file_info;
-static long temp_buffer [BUFFER_SIZE] IDATA_ATTR;
-
-/* Reformat samples from longs in processor's native endian mode to
- little-endian data with 2 bytes / sample. */
-uchar* format_samples (int bps, uchar *dst, long *src, ulong samcnt)
-{
-  long temp;
-
-  switch (bps) 
-  {
-    case 1:
-      while (samcnt--)
-      {
-        *dst++ = (uchar)(temp = (*src++ << 8));
-        *dst++ = (uchar)(temp >> 8);
-      }
-
-      break;
-
-    case 2:
-      while (samcnt--) 
-      {
-        *dst++ = (uchar)(temp = *src++);
-        *dst++ = (uchar)(temp >> 8);
-      }
-
-      break;
-
-    case 3:
-      while (samcnt--) 
-      {
-        *dst++ = (uchar)(temp = (*src++ >> 8));
-        *dst++ = (uchar)(temp >> 8);
-      }
-
-      break;
-
-    case 4:
-      while (samcnt--) 
-      {
-        *dst++ = (uchar)(temp = (*src++ >> 16));
-        *dst++ = (uchar)(temp >> 8);
-      }
-
-      break;
-  }
-
-  return dst;
-}
-
-/* this is our function to decode a memory block from a file */
-void wvpack_decode_data(file_info_struct* file_info, int samples_to_decode, WavpackContext **wpc)
-{
-  int bps = WavpackGetBytesPerSample(*wpc);
-  /* nothing to decode */
-  if (!samples_to_decode)
-  {
-    return;
-  }
- 
-  /* decode now */
-  ulong samples_unpacked = WavpackUnpackSamples(*wpc, temp_buffer, samples_to_decode);  
-
-  if (samples_unpacked)
-  {
-    /* update some infos */
-    file_info->current_sample += samples_unpacked;    
- 
-    /* for now, convert mono to stereo here, in place */
-    if (WavpackGetReducedChannels (*wpc) == 1) {
-      long *dst = temp_buffer + (samples_unpacked * 2);
-      long *src = temp_buffer + samples_unpacked;
-      long count = samples_unpacked;
-
-      while (count--) {
-        *--dst = *--src;
-        *--dst = *src;
-      }
-    }
- 
-    format_samples (bps, (uchar *) temp_buffer, temp_buffer, samples_unpacked * file_info->channels);
-    rb->write(file_info->outfile, temp_buffer, samples_unpacked * 4);
-  }   
-}
-
-/* callback function for wavpack 
-Maybe we do this at a lower level, but the
-first thing is to get all working */
-long Read(void* buffer, long size)
-{
-  long oldpos = file_info.curpos;
-  
-  if ((file_info.curpos + size) < file_info.filesize) 
-  {   
-    memcpy(buffer, &filebuf[file_info.curpos], size);
-    file_info.curpos += size;
-  } 
-  else 
-  {
-    memcpy(buffer, &filebuf[file_info.curpos], file_info.filesize-file_info.curpos);
-    file_info.curpos = file_info.filesize;
-  }
-    
-  return (file_info.curpos - oldpos);
-}
-
-#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* file)
-{
-  WavpackContext *wpc;
-  char error[80];
-  
-  /* generic plugin initialisation */
-  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 (local_init(file,"/wvtest.wav",&file_info,api)) 
-  {
-    return PLUGIN_ERROR;
-  }
-  
-  /* setup wavpack */
-  wpc = WavpackOpenFileInput(Read, error);
-
-  /* was there an error? */
-  if (!wpc) 
-  {
-    rb->splash(HZ*2, true, error);
-    return PLUGIN_ERROR;
-  }  
-
-  /* grap/set some infos (forcing some to temp values) */
-  file_info.channels = 2;
-  file_info.total_samples = WavpackGetNumSamples(wpc);
-  file_info.bitspersample = 16;
-  file_info.samplerate = WavpackGetSampleRate(wpc);
-  file_info.current_sample = 0;
-  
-  /* deciding loop */  
-  file_info.start_tick=*(rb->current_tick);
-  rb->button_clear_queue();
-
-  while (file_info.current_sample < file_info.total_samples) 
-  {
-    wvpack_decode_data(&file_info, BUFFER_SIZE / file_info.channels, &wpc);
-
-    display_status(&file_info);
-
-    if (rb->button_get(false)!=BUTTON_NONE) 
-    {
-      close_wav(&file_info);
-      return PLUGIN_OK;
-    }
-  }  
-          
-  close_wav(&file_info);    
-
-  /* do some last checks */
-  if ((WavpackGetNumSamples (wpc) != (ulong) -1) && (file_info.current_sample != WavpackGetNumSamples (wpc))) 
-  {
-    rb->splash(HZ*2, true, "incorrect number of samples!");
-    return PLUGIN_ERROR;
-  }  
-  
-  if (WavpackGetNumErrors (wpc)) {
-    rb->splash(HZ*2, true, "crc errors detected!");
-    return PLUGIN_ERROR;
-  }
-  
-  rb->splash(HZ*2, true, "FINISHED!");
-
-  return PLUGIN_OK;
-}
-
-#endif /* CONFIG_HWCODEC == MASNONE */