First audio codec playback attempt by Miikka Pekkarinen

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6574 a1c6a512-1295-4272-9138-f99709370657

8 files changed, 661 insertions, 2 deletions

diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES
index 64f85b1..6ed0e7c 100644
--- a/apps/plugins/SOURCES
+++ b/apps/plugins/SOURCES
@@ -72,6 +72,10 @@ mpa2wav.c
 a52towav.c
 flac2wav.c
 vorbis2wav.c
+#ifdef IRIVER_H100
+codecvorbis.c
+codecmpa.c
+#endif
 wv2wav.c
 mpc2wav.c
 midi2wav.c
diff --git a/apps/plugins/codecmpa.c b/apps/plugins/codecmpa.c
new file mode 100644
index 0000000..1125b4b
--- /dev/null
+++ b/apps/plugins/codecmpa.c
@@ -0,0 +1,404 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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;
+
+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
+
+#undef DEBUG_GAPLESS
+
+/* 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;
+  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;
+  
+  /* 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 */
+
+  next_track:
+  
+  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;
+  
+#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
+
+  samplesdone = 0;
+  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);
+  
+  samplecount = ci->id3->length * (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); */
+  
+  /* 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)) {
+            ci->seek_time = 0;
+            if (sample_loc >= samplecount + samplesdone)
+                break ;
+            samplecount += samplesdone - sample_loc;
+            samplesdone = sample_loc;
+        }
+    }
+
+    /* 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 */
+    for(i=0;i<Synth.pcm.length;i++)
+    {
+      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(Synth.pcm.samples[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(Synth.pcm.samples[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;
+      }
+  
+      /* 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/codecvorbis.c b/apps/plugins/codecvorbis.c
new file mode 100644
index 0000000..41db223
--- /dev/null
+++ b/apps/plugins/codecvorbis.c
@@ -0,0 +1,160 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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
+    
+  /* This function sets up the buffers and reads the file into RAM */
+    
+  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;
+
+  next_track:
+  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 ;
+      while (!ci->audiobuffer_insert(pcmbuf, n))
+        rb->yield();
+	    
+#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/lib/SOURCES b/apps/plugins/lib/SOURCES
index 0e8e14c..58356af 100644
--- a/apps/plugins/lib/SOURCES
+++ b/apps/plugins/lib/SOURCES
@@ -35,4 +35,7 @@ playergfx.c
 #endif
 #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
new file mode 100644
index 0000000..d9866ef
--- /dev/null
+++ b/apps/plugins/lib/codeclib.c
@@ -0,0 +1,36 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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"
+
+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
new file mode 100644
index 0000000..876e69b
--- /dev/null
+++ b/apps/plugins/lib/codeclib.h
@@ -0,0 +1,46 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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
index d864527..ed67184 100644
--- a/apps/plugins/lib/xxx2wav.c
+++ b/apps/plugins/lib/xxx2wav.c
@@ -40,14 +40,14 @@ void* codec_malloc(size_t size) {
 
   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)) {
       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);
 }
 
@@ -162,6 +162,11 @@ static unsigned char wav_header[44]={'R','I','F','F',    //  0 - ChunkID
                              };
 
 
+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;
diff --git a/apps/plugins/lib/xxx2wav.h b/apps/plugins/lib/xxx2wav.h
index 7e3afce..7e1b942 100644
--- a/apps/plugins/lib/xxx2wav.h
+++ b/apps/plugins/lib/xxx2wav.h
@@ -64,3 +64,4 @@ 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);