diff --git a/panda/src/audiotraits/audio_load_midi.cxx b/panda/src/audiotraits/audio_load_midi.cxx
index 61f3196ab1..c317e40336 100644
--- a/panda/src/audiotraits/audio_load_midi.cxx
+++ b/panda/src/audiotraits/audio_load_midi.cxx
@@ -16,63 +16,3 @@
 //
 ////////////////////////////////////////////////////////////////////
 
-#include <dconfig.h>
-#include "audio_pool.h"
-#include "config_audio.h"
-#include "audio_trait.h"
-
-Configure(audio_load_midi);
-
-#ifdef AUDIO_USE_MIKMOD
-
-#include "audio_mikmod_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMidi(Filename filename) {
-  return MikModMidi::load_midi(filename);
-}
-
-#elif defined(AUDIO_USE_RAD_MSS)
-
-#include "audio_rad_mss_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadMidi(Filename filename) {
-  return MilesSound::load(filename);
-}
-
-#elif defined(AUDIO_USE_WIN32)
-
-#include "audio_win_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadMidi(Filename filename) {
-  return WinMusic::load_midi(filename);
-}
-
-#elif defined(AUDIO_USE_LINUX)
-
-#include "audio_linux_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMidi(Filename) {
-  audio_cat->warning() << "linux doesn't support reading midi data yet"
-                       << endl;
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_NULL)
-
-// Null driver
-#include "audio_null_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMidi(Filename) {
-  return new NullSound();
-}
-
-#else /* AUDIO_USE_NULL */
-
-#error "unknown driver type"
-
-#endif /* AUDIO_USE_NULL */
-
-ConfigureFn(audio_load_midi) {
-  AudioPool::register_sound_loader("midi", AudioLoadMidi);
-  AudioPool::register_sound_loader("mid", AudioLoadMidi);
-}
diff --git a/panda/src/audiotraits/audio_load_mp3.cxx b/panda/src/audiotraits/audio_load_mp3.cxx
index f2917c5684..32d5b94d36 100644
--- a/panda/src/audiotraits/audio_load_mp3.cxx
+++ b/panda/src/audiotraits/audio_load_mp3.cxx
@@ -15,574 +15,3 @@
 // panda3d@yahoogroups.com .
 //
 ////////////////////////////////////////////////////////////////////
-
-#include <dconfig.h>
-#include "audio_pool.h"
-#include "config_audio.h"
-#include "audio_trait.h"
-#include "config_util.h"
-
-Configure(audio_load_mp3);
-
-#if !(defined(WIN32) && defined(AUDIO_USE_RAD_MSS))
-
-#include <math.h>
-
-extern "C" {
-#include <mpg123.h>
-}
-
-static bool initialized = false;
-static struct audio_info_struct ai;
-static struct frame fr;
-struct parameter param = {
-  FALSE, /* aggressive */
-  FALSE, /* shuffle */
-  FALSE, /* remote */
-  DECODE_AUDIO, /* write samples to audio device */
-  FALSE, /* silent operation */
-  FALSE, /* xterm title on/off */
-  0, /* second level buffer size */
-  TRUE, /* resync after stream error */
-  0, /* verbose level */
-#ifdef TERM_CONTROL
-  FALSE, /* term control */
-#endif /* TERM_CONTROL */
-  -1, /* force mono */
-  0, /* force stereo */
-  0, /* force 8-bit */
-  0, /* force rate */
-  0, /* down sample */
-  FALSE, /* check range */
-  0, /* double speed */
-  0, /* half speed */
-  0, /* force re-open. always (re)opens audio device for next song */
-  0, /* 3Dnow: autodetect from CPUFLAGS */
-  FALSE, /* 3Dnow: normal operation */
-  FALSE, /* try to run process in 'realtime mode' */
-  { 0, }, /* wav, cdr, au filename */
-  NULL, /* esdserver */
-  NULL, /* equalfile */
-  0, /* enable_equalizer */
-  32768, /* outscale */
-  0, /* startFrame */
-};
-static long numframes = -1;
-static int intflag = FALSE;
-static struct mpstr mp;
-// stuff I have to have to make the linkage happy
-int OutputDescriptor;
-int buffer_fd[2];
-struct reader *rd;
-txfermem* buffermem;
-
-static void set_synth_functions(struct frame* fr) {
-  typedef int (*func)(real*, int, unsigned char*, int*);
-  typedef int (*func_mono)(real*, unsigned char*, int*);
-  typedef void (*func_dct36)(real*, real*, real*, real*, real*);
-
-  int ds = fr->down_sample;
-  int p8=0;
-
-  static func funcs[][4] = {
-    { synth_1to1,
-      synth_2to1,
-      synth_4to1,
-      synth_ntom } ,
-    { synth_1to1_8bit,
-      synth_2to1_8bit,
-      synth_4to1_8bit,
-      synth_ntom_8bit }
-  };
-  static func_mono funcs_mono[2][2][4] = {
-    { { synth_1to1_mono2stereo,
-    synth_2to1_mono2stereo,
-    synth_4to1_mono2stereo,
-    synth_ntom_mono2stereo } ,
-      { synth_1to1_8bit_mono2stereo,
-    synth_2to1_8bit_mono2stereo,
-    synth_4to1_8bit_mono2stereo,
-    synth_ntom_8bit_mono2stereo } } ,
-    { { synth_1to1_mono,
-    synth_2to1_mono,
-    synth_4to1_mono,
-    synth_ntom_mono } ,
-      { synth_1to1_8bit_mono,
-    synth_2to1_8bit_mono,
-    synth_4to1_8bit_mono,
-    synth_ntom_8bit_mono } } ,
-  };
-
-  if ((ai.format & AUDIO_FORMAT_MASK) == AUDIO_FORMAT_8)
-    p8 = 1;
-  fr->synth = funcs[p8][ds];
-  fr->synth_mono = funcs_mono[param.force_stereo?0:1][p8][ds];
-  if (p8)
-    make_conv16to8_table(ai.format);
-}
-
-static void initialize(void) {
-  // make sure params say what we want
-  param.quiet = TRUE;
-  param.force_stereo = 1;
-  param.force_rate = audio_mix_freq;
-
-  memset(&mp, 0, sizeof(struct mpstr));
-  audio_info_struct_init(&ai);
-  audio_capabilities(&ai);
-
-  if (initialized)
-    return;
-
-  set_synth_functions(&fr);
-  make_decode_tables(param.outscale);
-  init_layer2(); /* inits also shared tables with layer1 */
-  init_layer3(fr.down_sample);
-  equalizer_cnt = 0;
-  for (int i=0; i<32; ++i) {
-    equalizer[0][i] = equalizer[1][i] = 1.0;
-    equalizer_sum[0][i] = equalizer_sum[1][i] = 0.0;
-  }
-  initialized = true;
-}
-
-class BufferStuff {
-private:
-  typedef pvector<unsigned char> Buffer;
-  typedef pvector<Buffer> Buffers;
-  Buffers _bufs;
-public:
-  BufferStuff(void) {}
-  ~BufferStuff(void) {
-  }
-  void add(unsigned char* b, unsigned long l) {
-    _bufs.push_back(Buffer(b, b+l));
-  }
-  unsigned long length(void) const {
-    unsigned long ret = 0;
-    for (Buffers::const_iterator i=_bufs.begin(); i!=_bufs.end(); ++i)
-      ret += (*i).size();
-    return ret;
-  }
-  void output(unsigned char* b) {
-    for (Buffers::const_iterator i=_bufs.begin(); i!=_bufs.end(); ++i)
-      for (Buffer::const_iterator j=(*i).begin(); j!=(*i).end(); ++j)
-    *(b++) = (*j);
-  }
-};
-
-static BufferStuff* my_buf;
-
-/*
-class BufferPart {
-private:
-  unsigned char* _ptr;
-  unsigned long _len;
-  BufferPart* _next;
-
-  BufferPart(void) : _ptr((unsigned char*)0L), _len(0), _next((BufferPart*)0L)
-     {}
-public:
-  BufferPart(unsigned char* b, unsigned long l) : _next((BufferPart*)0L),
-                                                  _len(l) {
-    _ptr = new unsigned char[l];
-    memcpy(_ptr, b, l);
-  }
-  ~BufferPart(void) {
-    delete _next;
-    delete [] _ptr;
-  }
-  BufferPart* add(unsigned char* b, unsigned long l) {
-    _next = new BufferPart(b, l);
-    return _next;
-  }
-  unsigned long length(void) const {
-    unsigned long ret = _len;
-    if (_next != (BufferPart*)0L)
-      ret += _next->length();
-    return ret;
-  }
-  void output(unsigned char* b) {
-     memcpy(b, _ptr, _len);
-     if (_next != (BufferPart*)0L)
-    _next->output(b+_len);
-  }
-};
-
-static BufferPart* my_buf_head;
-static BufferPart* my_buf_curr;
-*/
-
-/*
-string my_buf;
-*/
-
-extern "C" {
-int audio_open(struct audio_info_struct* ai) {
-  return 0;
-}
-
-int audio_reset_parameters(struct audio_info_struct* ai) {
-  audio_set_format(ai);
-  audio_set_channels(ai);
-  audio_set_rate(ai);
-  return 0;
-}
-
-int audio_rate_best_match(struct audio_info_struct* ai) {
-  if (!ai || ai->rate < 0)
-    return -1;
-  ai->rate = audio_mix_freq;
-  return 0;
-}
-
-int audio_set_rate(struct audio_info_struct* ai) {
-  if (ai->rate != audio_mix_freq)
-    audio_cat->warning()
-      << "trying to decode mp3 to rate other then mix rate (" << ai->rate
-      << " != " << audio_mix_freq << ")" << endl;
-  return 0;
-}
-
-int audio_set_channels(struct audio_info_struct* ai) {
-  if (ai->channels != 2)
-    audio_cat->warning() << "trying to decode mp3 to non-stereo ("
-             << ai->channels << " != 2)" << endl;
-  return 0;
-}
-
-int audio_set_format(struct audio_info_struct* ai) {
-  if (ai->format != AUDIO_FORMAT_SIGNED_16)
-    audio_cat->warning()
-      << "trying to decode mp3 to format other then signed 16-bit" << endl;
-  return 0;
-}
-
-int audio_get_formats(struct audio_info_struct* ai) {
-  return AUDIO_FORMAT_SIGNED_16;
-}
-
-int audio_play_samples(struct audio_info_struct* ai, unsigned char* buf,
-               int len) {
-  /*
-  if (my_buf_head == (BufferPart*)0L) {
-    my_buf_head = my_buf_curr = new BufferPart(buf, len);
-  } else {
-    my_buf_curr = my_buf_curr->add(buf, len);
-  }
-  */
-
-  if (my_buf == (BufferStuff*)0L)
-    my_buf = new BufferStuff;
-  my_buf->add(buf, len);
-
-  /*
-  string tmp;
-  for (int i=0; i<len; ++i)
-    tmp += buf[i];
-  my_buf += tmp;
-  */
-
-  return len;
-}
-
-int audio_close(struct audio_info_struct* ai) {
-  return 0;
-}
-
-// we won't use these functions, but they have to exist
-int cdr_open(struct audio_info_struct *ai, char *ame) { return 0; }
-int au_open(struct audio_info_struct *ai, char *name) { return 0; }
-int wav_open(struct audio_info_struct *ai, char *wavfilename) { return 0; }
-int wav_write(unsigned char *buf,int len) { return 0; }
-int cdr_close(void) { return 0; }
-int au_close(void) { return 0; }
-int wav_close(void) { return 0; }
-int xfermem_get_usedspace(txfermem*) { return 0; }
-}
-
-// static unsigned char* real_sample_buf;
-
-static void init_output(void) {
-  //  static int init_done = FALSE;
-  //  if (init_done)
-  //    return;
-  //  init_done = TRUE;
-  // + 1024 for NtoM rate converter
-  //  if (!(real_sample_buf=(unsigned char*)malloc(2*(audiobufsize*2 + 2*1024)))) {
-  if (!(pcm_sample=(unsigned char*)malloc(audiobufsize*2 + 2*1024))) {
-    audio_cat->fatal() << "cannot allocate sample buffer" << endl;
-    exit(1);
-  }
-  //  pcm_sample = &(real_sample_buf[1024]);
-  switch (param.outmode) {
-  case DECODE_AUDIO:
-    if (audio_open(&ai) < 0) {
-      audio_cat->fatal() << "could not open output stream" << endl;
-      exit(1);
-    }
-    break;
-  case DECODE_WAV:
-    wav_open(&ai, param.filename);
-    break;
-  case DECODE_AU:
-    au_open(&ai, param.filename);
-    break;
-  case DECODE_CDR:
-    cdr_open(&ai, param.filename);
-    break;
-  }
-}
-
-static void reset_audio(void) {
-  if (param.outmode == DECODE_AUDIO) {
-    audio_close(&ai);
-    if (audio_open(&ai) < 0) {
-      audio_cat->fatal() << "couldn't reopen" << endl;
-      exit(1);
-    }
-  }
-}
-
-int play_frame(struct mpstr* mp, int init, struct frame* fr) {
-  int clip;
-  long newrate;
-  long old_rate, old_format, old_channels;
-
-  if (fr->header_change || init) {
-    if (fr->header_change > 1 || init) {
-      old_rate = ai.rate;
-      old_format = ai.format;
-      old_channels = ai.channels;
-      newrate = freqs[fr->sampling_frequency]>>(param.down_sample);
-      fr->down_sample = param.down_sample;
-      audio_fit_capabilities(&ai, fr->stereo, newrate);
-      // check whether the fitter set our proposed rate
-      if (ai.rate != newrate) {
-    if (ai.rate == (newrate >> 1))
-      fr->down_sample++;
-    else if (ai.rate == (newrate >> 2))
-      fr->down_sample += 2;
-    else {
-      fr->down_sample = 3;
-      audio_cat->warning() << "flexable rate not heavily tested!" << endl;
-    }
-    if (fr->down_sample > 3)
-      fr->down_sample = 3;
-      }
-      switch (fr->down_sample) {
-      case 0:
-      case 1:
-      case 2:
-    fr->down_sample_sblimit = SBLIMIT >> (fr->down_sample);
-    break;
-      case 3:
-    {
-      long n = freqs[fr->sampling_frequency];
-      long m = ai.rate;
-      synth_ntom_set_step(n, m);
-      if (n>m) {
-        fr->down_sample_sblimit = SBLIMIT * m;
-        fr->down_sample_sblimit /= n;
-      } else
-        fr->down_sample_sblimit = SBLIMIT;
-    }
-    break;
-      }
-      set_synth_functions(fr);
-      init_output();
-      if (ai.rate != old_rate || ai.channels != old_channels ||
-      ai.format != old_format || param.force_reopen) {
-    if (param.force_mono < 0) {
-      if (ai.channels == 1)
-        fr->single = 3;
-      else
-        fr->single = -1;
-    } else
-      fr->single = param.force_mono;
-    param.force_stereo &= ~0x2;
-    if (fr->single >= 0 && ai.channels == 2)
-      param.force_stereo |= 0x2;
-    set_synth_functions(fr);
-    init_layer3(fr->down_sample_sblimit);
-    reset_audio();
-      }
-      if (intflag)
-    return !0;
-    }
-  }
-  if (fr->error_protection)
-    bsi.wordpointer += 2;
-  // do the decoding
-  switch (fr->lay) {
-  case 1:
-    if ((clip=do_layer1(mp, fr, param.outmode, &ai)) < 0)
-      return 0;
-    break;
-  case 2:
-    if ((clip=do_layer2(mp, fr, param.outmode, &ai)) < 0)
-      return 0;
-    break;
-  case 3:
-    if ((clip=do_layer3(mp, fr, param.outmode, &ai)) < 0)
-      return 0;
-    break;
-  default:
-    clip = 0;
-  }
-  if (clip > 0 && param.checkrange)
-    audio_cat->warning() << clip << " samples clipped" << endl;
-  return !0;
-}
-
-static void read_file(Filename filename, unsigned char** buf,
-              unsigned long& slen) {
-  int init;
-  unsigned long frameNum = 0;
-
-  initialize();
-  //  my_buf_head = my_buf_curr = (BufferPart*)0L;
-  my_buf = (BufferStuff*)0L;
-  //  my_buf = "";
-  if (open_stream((char*)(filename.to_os_specific().c_str()), -1)) {
-    long leftFrames, newFrame;
-
-    read_frame_init();
-    init = 1;
-    newFrame = param.startFrame;
-    leftFrames = numframes;
-    for (frameNum=0; read_frame(&fr) && leftFrames && !intflag; ++frameNum) {
-      if ((frameNum % 100) == 0)
-        if (audio_cat.is_debug())
-          audio_cat->debug(false) << ".";
-      if (frameNum < param.startFrame || (param.doublespeed &&
-                      (frameNum % param.doublespeed))) {
-    if (fr.lay == 3)
-      set_pointer(512);
-    continue;
-      }
-      if (leftFrames > 0)
-    --leftFrames;
-      if (!play_frame(&mp, init, &fr)) {
-    audio_cat->error() << "Error in frame #" << frameNum << endl;
-    break;
-      }
-      init = 0;
-    }
-    rd->close(rd);
-    if (intflag) {
-      intflag = FALSE;
-    }
-  }
-  if (audio_cat.is_debug())
-    audio_cat->debug(false) << endl;
-  audio_flush(param.outmode, &ai);
-  switch (param.outmode) {
-  case DECODE_AUDIO:
-    audio_close(&ai);
-    break;
-  case DECODE_WAV:
-    wav_close();
-    break;
-  case DECODE_AU:
-    au_close();
-    break;
-  case DECODE_CDR:
-    cdr_close();
-    break;
-  }
-  /*
-  if (real_sample_buf != (unsigned char*)0L) {
-    free(real_sample_buf);
-    pcm_sample = (unsigned char*)0L;
-  }
-  */
-  if (pcm_sample != (unsigned char*)0L) {
-    free(pcm_sample);
-    pcm_sample = (unsigned char*)0L;
-  }
-  // generate output
-  /*
-  slen = my_buf_head->length();
-  *buf = new byte[slen];
-  my_buf_head->output(*buf);
-  delete my_buf_head;
-  */
-
-  slen = my_buf->length();
-  *buf = new byte[slen];
-  my_buf->output(*buf);
-  delete my_buf;
-  my_buf = (BufferStuff*)0L;
-
-  /*
-  slen = my_buf.size();
-  *buf = new byte[slen];
-  memcpy(*buf, my_buf.data(), slen);
-  */
-}
-#endif
-
-#ifdef AUDIO_USE_MIKMOD
-
-#include "audio_mikmod_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMp3(Filename) {
-  audio_cat->warning() << "Mikmod doesn't support reading mp3 data yet"
-               << endl;
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_RAD_MSS)
-
-#include "audio_rad_mss_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadMp3(Filename filename) {
-  return MilesSound::load(filename);
-}
-
-#elif defined(AUDIO_USE_WIN32)
-
-#include "audio_win_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadMp3(Filename filename) {
-  unsigned char* buf;
-  unsigned long len;
-  read_file(filename, &buf, len);
-  if (buf != (unsigned char*)0L) {
-    return WinSample::load_raw(buf, len);
-  }
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_LINUX)
-
-#include "audio_linux_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMp3(Filename filename) {
-  unsigned char* buf;
-  unsigned long len;
-  read_file(filename, &buf, len);
-  if (buf != (unsigned char*)0L) {
-    return LinuxSample::load_raw(buf, len);
-  }
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_NULL)
-
-#include "audio_null_traits.h"
-
-AudioTraits::SoundClass* AudioLoadMp3(Filename) {
-  return new NullSound();
-}
-
-#else
-
-#error "unknown audio driver type"
-
-#endif
-
-ConfigureFn(audio_load_mp3) {
-  AudioPool::register_sound_loader("mp3", AudioLoadMp3);
-}
diff --git a/panda/src/audiotraits/audio_load_st.cxx b/panda/src/audiotraits/audio_load_st.cxx
index 9b4ec16123..cff87b89f1 100644
--- a/panda/src/audiotraits/audio_load_st.cxx
+++ b/panda/src/audiotraits/audio_load_st.cxx
@@ -15,456 +15,3 @@
 // panda3d@yahoogroups.com .
 //
 ////////////////////////////////////////////////////////////////////
-
-#include <dconfig.h>
-#include "audio_pool.h"
-#include "config_audio.h"
-#include "audio_trait.h"
-
-#ifdef HAVE_SOXST
-
-extern "C" {
-#include <sox/st.h>
-#include <sox/patchlvl.h>
-}
-
-#if (PATCHLEVEL == 16)
-#define FORMATS formats
-#define ENCODEFIELD style
-#define EFFECTS_TYPE struct effect
-#define STREAM_TYPE struct soundstream
-#define GETTYPE gettype
-#define COMPAT_EOF (-1)
-#define CHECKFORMAT checkformat
-#define SIZES sizes
-#define ENCODING styles
-#define COPYFORMAT copyformat
-#define UPDATEEFFECT(a, b, c, d) \
-  (a)->ininfo.channels = (b)->info.channels; \
-  (a)->outinfo.channels = (c)->info.channels; \
-  (a)->ininfo.rate = (b)->info.rate; \
-  (a)->outinfo.rate = (c)->info.rate;
-#else /* PATCHLEVEL != 16 */
-#define FORMATS st_formats
-#define ENCODEFIELD encoding
-#define EFFECTS_TYPE struct st_effect
-#define STREAM_TYPE struct st_soundstream
-#define GETTYPE st_gettype
-#define COMPAT_EOF ST_EOF
-#define CHECKFORMAT st_checkformat
-#define SIZES st_sizes_str
-#define ENCODING st_encodings_str
-#define COPYFORMAT st_copyformat
-#define UPDATEEFFECT st_updateeffect
-#endif /* PATCHLEVEL */
-
-#endif /* HAVE_SOXST */
-
-Configure(audio_load_st);
-
-#ifdef HAVE_SOXST
-
-// the effects we will be using are to change the rate and number of channels
-static EFFECTS_TYPE efftab[5];  // left/mono channel effects
-static EFFECTS_TYPE efftabR[5]; // right channel effects
-static int neffects;            // how many effects are in action
-
-static STREAM_TYPE iformat;    // holder for the input
-static STREAM_TYPE oformat;   // holder for fake output;
-
-INLINE static void init_stream(void) {
-  iformat.info.rate = 0;
-  iformat.info.size = -1;
-  iformat.info.ENCODEFIELD = -1;
-  iformat.info.channels = -1;
-  iformat.comment = (char*)0L;
-  iformat.swap = 0;
-  iformat.filetype = (char*)0L;
-  iformat.fp = stdin;
-  iformat.filename = "input";
-
-  oformat.info.rate = audio_mix_freq;
-  oformat.info.size = -1;
-  oformat.info.ENCODEFIELD = -1;
-  oformat.info.channels = 2;
-  oformat.comment = (char*)0L;
-  oformat.swap = 0;
-  oformat.filetype = (char*)0L;
-  oformat.fp = stdout;
-  oformat.filename = "output";
-}
-
-INLINE static void compat_geteffect(EFFECTS_TYPE* eff, const char* name) {
-#if (PATCHLEVEL == 16)
-  eff->name = (char*)name;
-  geteffect(eff);
-#else /* PATCHLEVEL == 16 */
-  st_geteffect(eff, name);
-#endif /* PATCHLEVEL == 16 */
-}
-
-typedef void effOptFunc(EFFECTS_TYPE*, int, char**);
-#ifndef HAVE_DEFINED_BYTE
-typedef unsigned char byte;
-#define HAVE_DEFINED_BYTE
-#endif /* HAVE_DEFINED_BYTE */
-
-INLINE static void check_effects(void) {
-  bool needrate = (iformat.info.rate != audio_mix_freq);
-  bool needchan = (iformat.info.channels != 2);
-  effOptFunc* func;
-
-  // efftab[0] is always the input stream and always exists
-  neffects = 1;
-
-  // if reducing the number of samples, it is faster to run all effects
-  // after the resample effect
-  if (needrate) {
-    compat_geteffect(&efftab[neffects], "resample");
-    // setup and give default opts
-    func = (effOptFunc*)(efftab[neffects].h->getopts);
-    (*func)(&efftab[neffects],(int)0,(char**)0L);
-    // copy format info to effect table
-    UPDATEEFFECT(&efftab[neffects], &iformat, &oformat, 0);
-    // rate can't handle multiple channels so be sure and account for that
-    if (efftab[neffects].ininfo.channels > 1)
-      memcpy(&efftabR[neffects], &efftab[neffects], sizeof(EFFECTS_TYPE));
-    ++neffects;
-  }
-  // if we ever have more then 2 channels in an input file, we will need to
-  // deal with that somewhere here
-  if (needchan) {
-    compat_geteffect(&efftab[neffects], "avg");
-    //setup and give default opts
-    func = (effOptFunc*)(efftab[neffects].h->getopts);
-    (*func)(&efftab[neffects],(int)0,(char**)0L);
-    // copy format info to effect table
-    UPDATEEFFECT(&efftab[neffects], &iformat, &oformat, 0);
-    ++neffects;
-  }
-}
-
-typedef void effFlowFunc(EFFECTS_TYPE*, LONG*, LONG*, LONG*, LONG*);
-
-static LONG ibufl[BUFSIZ/2];
-static LONG ibufr[BUFSIZ/2];
-static LONG obufl[BUFSIZ/2];
-static LONG obufr[BUFSIZ/2];
-
-static int flow_effect(int e) {
-  LONG i, done, idone, odone, idonel, odonel, idoner, odoner;
-  LONG *ibuf, *obuf;
-  effFlowFunc* eflow;
-
-  // is there any input data?
-  if (efftab[e-1].odone == efftab[e-1].olen)
-    return 0;
-  if (!efftabR[e].name) {
-    // no stereo data, or effect can handle stereo data.  so run effect
-    // over the entire buffer
-    idone = efftab[e-1].olen - efftab[e-1].odone;
-    odone = BUFSIZ;
-    eflow = (effFlowFunc*)(efftab[e].h->flow);
-    (*eflow)(&efftab[e], &efftab[e-1].obuf[efftab[e-1].odone], efftab[e].obuf,
-         &idone, &odone);
-    efftab[e-1].odone += idone;
-    efftab[e].odone = 0;
-    efftab[e].olen = odone;
-    done = idone + odone;
-  } else {
-    // put stereo data in two seperate buffers and run effect on each of them
-    idone = efftab[e-1].olen - efftab[e-1].odone;
-    odone = BUFSIZ;
-    ibuf = &efftab[e-1].obuf[efftab[e-1].odone];
-    for (i=0; i<idone; i+=2) {
-      ibufl[i/2] = *ibuf++;
-      ibufr[i/2] = *ibuf++;
-    }
-    // left
-    idonel = (idone + 1)/2; // odd-length logic
-    odonel = odone / 2;
-    eflow = (effFlowFunc*)(efftab[e].h->flow);
-    (*eflow)(&efftab[e], ibufl, obufl, &idonel, &odonel);
-    // right
-    idoner = idone/2;  // odd-length logic
-    odoner = odone/2;
-    eflow = (effFlowFunc*)(efftabR[e].h->flow);
-    (*eflow)(&efftabR[e], ibufr, obufr, &idoner, &odoner);
-    obuf = efftab[e].obuf;
-    // this loop implies that left and right effects will always output
-    // the same amount of data
-    for (i=0; i<odoner; i++) {
-      *obuf++ = obufl[i];
-      *obuf++ = obufr[i];
-    }
-    efftab[e-1].odone += idonel + idoner;
-    efftab[e].odone = 0;
-    efftab[e].olen = odonel + odoner;
-    done = idonel + idoner + odonel + odoner;
-  }
-  if (done == 0)
-    audio_cat->error() << "Effect took & gave no samples!" << endl;
-  return 1;
-}
-
-typedef void effDrainFunc(EFFECTS_TYPE*, LONG*, LONG*);
-
-static int drain_effect(int e) {
-  LONG i, olen, olenl, olenr;
-  LONG *obuf;
-  effDrainFunc* edrain;
-
-  if (!efftabR[e].name) {
-    efftab[e].olen = BUFSIZ;
-    edrain = (effDrainFunc*)(efftab[e].h->drain);
-    (*edrain)(&efftab[e], efftab[e].obuf, &efftab[e].olen);
-  } else {
-    olen = BUFSIZ;
-    // left
-    olenl = olen / 2;
-    edrain = (effDrainFunc*)(efftab[e].h->drain);
-    (*edrain)(&efftab[e], obufl, &olenl);
-    // right
-    olenr = olen / 2;
-    edrain = (effDrainFunc*)(efftab[e].h->drain);
-    (*edrain)(&efftabR[e], obufr, &olenr);
-    obuf = efftab[e].obuf;
-    // this loop implies left and right effect will always output the same
-    // amount of data
-    for (i=0; i<olenr; ++i) {
-      *obuf++ = obufl[i];
-      *obuf++ = obufr[i];
-    }
-    efftab[e].olen = olenl + olenr;
-  }
-  return (efftab[e].olen);
-}
-
-typedef int formatSReadFunc(STREAM_TYPE*);
-typedef LONG formatReadFunc(STREAM_TYPE*, LONG*, LONG);
-typedef int formatStopReadFunc(STREAM_TYPE*);
-typedef void effStartFunc(EFFECTS_TYPE*);
-typedef void effStopFunc(EFFECTS_TYPE*);
-
-static void read_file(Filename filename, byte** buf, unsigned long& slen) {
-  int e, havedata;
-  ostringstream out;
-  formatSReadFunc* srfunc;
-  formatReadFunc* rfunc;
-  formatStopReadFunc* strfunc;
-  effStartFunc* esfunc;
-  effStopFunc* estfunc;
-
-  init_stream();
-  if ((iformat.fp = fopen(filename.c_str(), READBINARY)) == NULL) {
-    audio_cat->error() << "could not open '" << filename << "'" << endl;
-    *buf = (byte*)0L;
-    slen = 0;
-    return;
-  }
-  iformat.filename = (char*)filename.c_str();
-  iformat.filetype = (char*)filename.get_extension().c_str();
-  iformat.comment = (char*)filename.c_str();  // for lack of anything better
-  // now we start some more real work
-  GETTYPE(&iformat);
-  // read and write starters can change their formats
-  srfunc = (formatSReadFunc*)(iformat.h->startread);
-  if ((*srfunc)(&iformat) == COMPAT_EOF) {
-    audio_cat->error() << "failed to start read" << endl;
-    *buf = (byte*)0L;
-    slen = 0;
-    return;
-  }
-  CHECKFORMAT(&iformat);
-  if (audio_cat.is_debug())
-    audio_cat->debug() << "Input file '" << iformat.filename
-               << "': sample rate = " << iformat.info.rate
-               << "  size = " << SIZES[iformat.info.size]
-               << "  encoding = " << ENCODING[iformat.info.ENCODEFIELD]
-               << "  " << iformat.info.channels
-               << ((iformat.info.channels > 1)?"channels":"channel")
-               << endl;
-  if (audio_cat.is_debug())
-    audio_cat->debug() << "Input file comment: '" << iformat.comment << "'"
-               << endl;
-  COPYFORMAT(&iformat, &oformat);
-  check_effects();
-  // start all effects
-  for (e=1; e<neffects; ++e) {
-    esfunc = (effStartFunc*)(efftab[e].h->start);
-    (*esfunc)(&efftab[e]);
-    if (efftabR[e].name) {
-      esfunc = (effStartFunc*)(efftabR[e].h->start);
-      (*esfunc)(&efftabR[e]);
-    }
-  }
-  // reserve output buffers for all effects
-  for (e=0; e<neffects; e++) {
-    efftab[e].obuf = new LONG[BUFSIZ*sizeof(LONG)];
-    if (efftabR[e].name)
-      efftabR[e].obuf = new LONG[BUFSIZ*sizeof(LONG)];
-  }
-  // get the main while loop ready, have some data for it to start on
-  rfunc = (formatReadFunc*)(iformat.h->read);
-  efftab[0].olen = (*rfunc)(&iformat, efftab[0].obuf, (LONG)BUFSIZ);
-  efftab[0].odone = 0;
-  // run input data thru effects and get more until olen == 0
-  while (efftab[0].olen > 0) {
-    // mark chain as empty
-    for (e=1; e<neffects; ++e)
-      efftab[e].odone = efftab[e].olen = 0;
-    do {
-      ULONG w;
-      // run entire chain backwards: pull, don't push
-      // this is because buffering system isn't a nice queueing system
-      for (e=neffects-1; e>0; --e)
-    if (flow_effect(e))
-      break;
-      // add to output data
-      if (efftab[neffects-1].olen>efftab[neffects-1].odone) {
-    for (LONG i=0; i<efftab[neffects-1].olen; ++i) {
-      LONG foo = efftab[neffects-1].obuf[i];
-      signed short bar = (foo >> 16);
-      unsigned char *b = (unsigned char*)&bar;
-      out << *b << *(++b);
-    }
-    efftab[neffects-1].odone = efftab[neffects-1].olen;
-      }
-      // if there is still stuff in the pipeline, setup to flow effects again
-      havedata = 0;
-      for (e=0; e<neffects-1; ++e)
-    if (efftab[e].odone < efftab[e].olen) {
-      havedata = 1;
-      break;
-    }
-    } while (havedata);
-    // read another chunk
-    rfunc = (formatReadFunc*)(iformat.h->read);
-    efftab[0].olen = (*rfunc)(&iformat, efftab[0].obuf, (LONG)BUFSIZ);
-    efftab[0].odone = 0;
-  }
-  // drain the effects out first to last.  push the residue thru subsequent
-  // effects.  suck.
-  for (e=1; e<neffects; ++e)
-    while (1) {
-      if (drain_effect(e) == 0)
-    break;  // get out of while loop
-      if (efftab[neffects-1].olen > 0) {
-    for (LONG i=0; i<efftab[neffects-1].olen; ++i) {
-      LONG foo = efftab[neffects-1].obuf[i];
-      signed short bar = (foo >> 16);
-      unsigned char *b = (unsigned char*)&bar;
-      out << *b << *(++b);
-    }
-      }
-      if (efftab[e].olen != BUFSIZ)
-    break;
-    }
-  // stop all effects.  In so doing, some may generate more data
-  for (e=1; e<neffects; ++e) {
-    estfunc = (effStopFunc*)(efftab[e].h->stop);
-    (*estfunc)(&efftab[e]);
-    if (efftabR[e].name) {
-      estfunc = (effStopFunc*)(efftabR[e].h->stop);
-      (*estfunc)(&efftabR[e]);
-    }
-  }
-  // stop reading the file
-  strfunc = (formatStopReadFunc*)(iformat.h->stopread);
-  if ((*strfunc)(&iformat) == COMPAT_EOF) {
-    audio_cat->error() << "error stoping input file" << endl;
-  }
-  fclose(iformat.fp);
-  // generate output
-  string s = out.str();
-  slen = s.length();
-  *buf = new byte[slen];
-  memcpy(*buf, s.data(), slen);
-}
-
-extern "C" {
-void cleanup(void) {
-  // make sure everything is shut down
-  if (iformat.fp)
-    fclose(iformat.fp);
-}
-}
-
-#endif /* HAVE_SOXST */
-
-#ifdef AUDIO_USE_MIKMOD
-
-AudioTraits::SoundClass* AudioLoadSt(Filename) {
-  audio_cat->warning() << "MikMod doesn't support reading raw data yet"
-               << endl;
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_RAD_MSS)
-
-#include "audio_rad_mss_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadSt(Filename filename) {
-  return MilesSound::load(filename);
-}
-
-#elif defined(AUDIO_USE_WIN32)
-
-#include "audio_win_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadSt(Filename filename) {
-#ifdef HAVE_SOXST
-  unsigned char* buf;
-  unsigned long len;
-  read_file(filename, &buf, len);
-  if (buf != (unsigned char*)0L) {
-    return WinSample::load_raw(buf, len);
-  }
-#endif /* HAVE_SOXST */
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_LINUX)
-
-#include "audio_linux_traits.h"
-
-AudioTraits::SoundClass* AudioLoadSt(Filename filename) {
-#ifdef HAVE_SOXST
-  byte* buf;
-  unsigned long len;
-  read_file(filename, &buf, len);
-  if (buf != (byte*)0L) {
-    return LinuxSample::load_raw(buf, len);
-  }
-#endif /* HAVE_SOXST */
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_NULL)
-
-// Null driver
-#include "audio_null_traits.h"
-
-AudioTraits::SoundClass* AudioLoadSt(Filename) {
-  return new NullSound();
-}
-
-#else /* AUDIO_USE_NULL */
-
-#error "unknown audio driver type"
-
-#endif /* AUDIO_USE_NULL */
-
-ConfigureFn(audio_load_st) {
-#ifdef HAVE_SOXST
-  for (int i=0; FORMATS[i].names != (char**)0L; ++i)
-    for (int j=0; FORMATS[i].names[j] != (char*)0L; ++j) {
-      if (audio_cat.is_debug())
-    audio_cat->debug() << "adding reader for '." << FORMATS[i].names[j]
-               << "'" << endl;
-      AudioPool::register_sound_loader(FORMATS[i].names[j], AudioLoadSt);
-    }
-#else /* HAVE_SOXST */
-  audio_cat->info() << "HAVE_SOXST is not defined" << endl;
-#endif /* HAVE_SOXST */
-}
diff --git a/panda/src/audiotraits/audio_load_wav.cxx b/panda/src/audiotraits/audio_load_wav.cxx
index 7f7cc51fde..7355633a1c 100644
--- a/panda/src/audiotraits/audio_load_wav.cxx
+++ b/panda/src/audiotraits/audio_load_wav.cxx
@@ -15,64 +15,3 @@
 // panda3d@yahoogroups.com .
 //
 ////////////////////////////////////////////////////////////////////
-
-#include <dconfig.h>
-#include "audio_pool.h"
-#include "config_audio.h"
-
-Configure(audio_load_wav);
-
-#include "audio_trait.h"
-
-#ifdef AUDIO_USE_MIKMOD
-
-#include "audio_mikmod_traits.h"
-
-AudioTraits::SoundClass* AudioLoadWav(Filename filename) {
-  return MikModSample::load_wav(filename);
-}
-
-#elif defined(AUDIO_USE_RAD_MSS)
-
-#include "audio_rad_mss_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadWav(Filename filename) {
-  return MilesSound::load(filename);
-}
-
-#elif defined(AUDIO_USE_WIN32)
-
-#include "audio_win_traits.h"
-
-EXPCL_MISC AudioTraits::SoundClass* AudioLoadWav(Filename filename) {
-  return WinSample::load_wav(filename);
-}
-
-#elif defined(AUDIO_USE_LINUX)
-
-#include "audio_linux_traits.h"
-
-AudioTraits::SoundClass* AudioLoadWav(Filename) {
-  audio_cat->error() << "Linux driver does not natively support WAV."
-                     << "  Try the 'st' loader." << endl;
-  return (AudioTraits::SoundClass*)0L;
-}
-
-#elif defined(AUDIO_USE_NULL)
-
-// Null driver
-#include "audio_null_traits.h"
-
-AudioTraits::SoundClass* AudioLoadWav(Filename) {
-  return new NullSound();
-}
-
-#else /* AUDIO_USE_NULL */
-
-#error "unknown implementation driver"
-
-#endif /* AUDIO_USE_NULL */
-
-ConfigureFn(audio_load_wav) {
-  AudioPool::register_sound_loader("wav", AudioLoadWav);
-}