libmpcodecs: remove redundant audio and video decoders

Probably all of these are supported by libavcodec. Missing things can
be added back.

Also remove qtpalette.h. It was used by demux_mov.c, and should have
been deleted with commit 1fde09db6f4ce.

21 files changed, 0 insertions, 4351 deletions

diff --git a/libmpcodecs/ad.c b/libmpcodecs/ad.c
index 0c56f0d546..aea9badadf 100644
--- a/libmpcodecs/ad.c
+++ b/libmpcodecs/ad.c
@@ -33,42 +33,14 @@
 
 extern const ad_functions_t mpcodecs_ad_mpg123;
 extern const ad_functions_t mpcodecs_ad_ffmpeg;
-extern const ad_functions_t mpcodecs_ad_liba52;
-extern const ad_functions_t mpcodecs_ad_hwac3;
-extern const ad_functions_t mpcodecs_ad_hwmpa;
-extern const ad_functions_t mpcodecs_ad_pcm;
-extern const ad_functions_t mpcodecs_ad_dvdpcm;
-extern const ad_functions_t mpcodecs_ad_alaw;
-extern const ad_functions_t mpcodecs_ad_imaadpcm;
-extern const ad_functions_t mpcodecs_ad_faad;
-extern const ad_functions_t mpcodecs_ad_libmad;
 extern const ad_functions_t mpcodecs_ad_spdif;
-extern const ad_functions_t mpcodecs_ad_libdca;
 
 const ad_functions_t * const mpcodecs_ad_drivers[] =
 {
 #ifdef CONFIG_MPG123
   &mpcodecs_ad_mpg123,
 #endif
-#ifdef CONFIG_LIBA52
-  &mpcodecs_ad_liba52,
-#endif
-  &mpcodecs_ad_hwac3,
-  &mpcodecs_ad_hwmpa,
   &mpcodecs_ad_ffmpeg,
   &mpcodecs_ad_spdif,
-  &mpcodecs_ad_pcm,
-  &mpcodecs_ad_dvdpcm,
-  &mpcodecs_ad_alaw,
-  &mpcodecs_ad_imaadpcm,
-#ifdef CONFIG_FAAD
-  &mpcodecs_ad_faad,
-#endif
-#ifdef CONFIG_LIBMAD
-  &mpcodecs_ad_libmad,
-#endif
-#ifdef CONFIG_LIBDCA
-  &mpcodecs_ad_libdca,
-#endif
   NULL
 };
diff --git a/libmpcodecs/ad_alaw.c b/libmpcodecs/ad_alaw.c
deleted file mode 100644
index 4a8f3849cf..0000000000
--- a/libmpcodecs/ad_alaw.c
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "config.h"
-#include "ad_internal.h"
-
-static const ad_info_t info =
-{
-	"aLaw/uLaw audio decoder",
-	"alaw",
-	"Nick Kurshev",
-	"A'rpi",
-	""
-};
-
-LIBAD_EXTERN(alaw)
-
-#include "native/alaw.h"
-
-static int init(sh_audio_t *sh_audio)
-{
-  /* aLaw audio codec:*/
-  if(!sh_audio->wf) return 0;
-  sh_audio->channels=sh_audio->wf->nChannels;
-  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-  sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate;
-  sh_audio->samplesize=2;
-  return 1;
-}
-
-static int preinit(sh_audio_t *sh)
-{
-  sh->audio_out_minsize=2048;
-  sh->ds->ss_div = 1; // 1 samples/packet
-  sh->ds->ss_mul = sh->wf->nChannels; // bytes/packet
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-  int skip;
-    switch(cmd)
-    {
-      case ADCTRL_SKIP_FRAME:
-	skip=sh->i_bps/16;
-	skip=skip&(~3);
-	demux_read_data(sh->ds,NULL,skip);
-	return CONTROL_TRUE;
-      default:
-	return CONTROL_UNKNOWN;
-    }
-  return CONTROL_UNKNOWN;
-}
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
- int len;
- int l=demux_read_data(sh_audio->ds,buf,minlen/2);
- unsigned short *d=(unsigned short *) buf;
- unsigned char *s=buf;
- len=2*l;
- if(sh_audio->format==6 || sh_audio->format==0x77616C61){
- /* aLaw */
-   while(l>0){ --l; d[l]=alaw2short[s[l]]; }
- } else {
- /* uLaw */
-    while(l>0){ --l; d[l]=ulaw2short[s[l]]; }
- }
- return len;
-}
diff --git a/libmpcodecs/ad_dvdpcm.c b/libmpcodecs/ad_dvdpcm.c
deleted file mode 100644
index 41f6a1426d..0000000000
--- a/libmpcodecs/ad_dvdpcm.c
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "ad_internal.h"
-
-static const ad_info_t info =
-{
-	"Uncompressed DVD/VOB LPCM audio decoder",
-	"dvdpcm",
-	"Nick Kurshev",
-	"A'rpi",
-	""
-};
-
-LIBAD_EXTERN(dvdpcm)
-
-static int init(sh_audio_t *sh)
-{
-/* DVD PCM Audio:*/
-    sh->i_bps = 0;
-    if(sh->codecdata_len==3){
-	// we have LPCM header:
-	unsigned char h=sh->codecdata[1];
-	sh->channels=1+(h&7);
-	switch((h>>4)&3){
-	case 0: sh->samplerate=48000;break;
-	case 1: sh->samplerate=96000;break;
-	case 2: sh->samplerate=44100;break;
-	case 3: sh->samplerate=32000;break;
-	}
-	switch ((h >> 6) & 3) {
-	  case 0:
-	    sh->sample_format = AF_FORMAT_S16_BE;
-	    sh->samplesize = 2;
-	    break;
-	  case 1:
-	    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
-	    sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
-	  case 2:
-	    sh->sample_format = AF_FORMAT_S24_BE;
-	    sh->samplesize = 3;
-	    break;
-	  default:
-	    sh->sample_format = AF_FORMAT_S16_BE;
-	    sh->samplesize = 2;
-	}
-    } else {
-	// use defaults:
-	sh->channels=2;
-	sh->samplerate=48000;
-	sh->sample_format = AF_FORMAT_S16_BE;
-	sh->samplesize = 2;
-    }
-    if (!sh->i_bps)
-    sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
-    return 1;
-}
-
-static int preinit(sh_audio_t *sh)
-{
-  sh->audio_out_minsize=2048;
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-  int skip;
-    switch(cmd)
-    {
-      case ADCTRL_SKIP_FRAME:
-	skip=sh->i_bps/16;
-	skip=skip&(~3);
-	demux_read_data(sh->ds,NULL,skip);
-	return CONTROL_TRUE;
-    }
-  return CONTROL_UNKNOWN;
-}
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
-  int j,len;
-  if (sh_audio->samplesize == 3) {
-    if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
-      // 20 bit
-      // not sure if the "& 0xf0" and "<< 4" are the right way around
-      // can somebody clarify?
-      for (j = 0; j < minlen; j += 12) {
-        char tmp[10];
-        len = demux_read_data(sh_audio->ds, tmp, 10);
-        if (len < 10) break;
-        // first sample
-        buf[j + 0] = tmp[0];
-        buf[j + 1] = tmp[1];
-        buf[j + 2] = tmp[8] & 0xf0;
-        // second sample
-        buf[j + 3] = tmp[2];
-        buf[j + 4] = tmp[3];
-        buf[j + 5] = tmp[8] << 4;
-        // third sample
-        buf[j + 6] = tmp[4];
-        buf[j + 7] = tmp[5];
-        buf[j + 8] = tmp[9] & 0xf0;
-        // fourth sample
-        buf[j + 9] = tmp[6];
-        buf[j + 10] = tmp[7];
-        buf[j + 11] = tmp[9] << 4;
-      }
-      len = j;
-    } else {
-      // 24 bit
-      for (j = 0; j < minlen; j += 12) {
-        char tmp[12];
-        len = demux_read_data(sh_audio->ds, tmp, 12);
-        if (len < 12) break;
-        // first sample
-        buf[j + 0] = tmp[0];
-        buf[j + 1] = tmp[1];
-        buf[j + 2] = tmp[8];
-        // second sample
-        buf[j + 3] = tmp[2];
-        buf[j + 4] = tmp[3];
-        buf[j + 5] = tmp[9];
-        // third sample
-        buf[j + 6] = tmp[4];
-        buf[j + 7] = tmp[5];
-        buf[j + 8] = tmp[10];
-        // fourth sample
-        buf[j + 9] = tmp[6];
-        buf[j + 10] = tmp[7];
-        buf[j + 11] = tmp[11];
-      }
-      len = j;
-    }
-  } else
-  len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
-  return len;
-}
diff --git a/libmpcodecs/ad_faad.c b/libmpcodecs/ad_faad.c
deleted file mode 100644
index d750f2ff30..0000000000
--- a/libmpcodecs/ad_faad.c
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
- * MPlayer AAC decoder using libfaad2
- *
- * Copyright (C) 2002 Felix Buenemann <atmosfear at users.sourceforge.net>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <neaacdec.h>
-
-#include "config.h"
-#include "options.h"
-#include "ad_internal.h"
-#include "libaf/reorder_ch.h"
-
-static const ad_info_t info =
-{
-	"AAC (MPEG2/4 Advanced Audio Coding)",
-	"faad",
-	"Felix Buenemann",
-	"faad2",
-	"uses libfaad2"
-};
-
-LIBAD_EXTERN(faad)
-
-/* configure maximum supported channels, *
- * this is theoretically max. 64 chans   */
-#define FAAD_MAX_CHANNELS 8
-#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)
-
-//#define AAC_DUMP_COMPRESSED
-
-static faacDecHandle faac_hdec;
-static faacDecFrameInfo faac_finfo;
-
-static int preinit(sh_audio_t *sh)
-{
-  sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS;
-  sh->audio_in_minsize=FAAD_BUFFLEN;
-  return 1;
-}
-
-static int aac_probe(unsigned char *buffer, int len)
-{
-  int i = 0, pos = 0;
-  mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len);
-  while(i <= len-4) {
-    if(
-       ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) ||
-       (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F')
-    ) {
-      pos = i;
-      break;
-    }
-    mp_msg(MSGT_DECAUDIO,MSGL_V, "AUDIO PAYLOAD: %x %x %x %x\n", buffer[i], buffer[i+1], buffer[i+2], buffer[i+3]);
-    i++;
-  }
-  mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos);
-  return pos;
-}
-
-static int init(sh_audio_t *sh)
-{
-  struct MPOpts *opts = sh->opts;
-  unsigned long faac_samplerate;
-  unsigned char faac_channels;
-  int faac_init, pos = 0;
-  faac_hdec = faacDecOpen();
-
-  // If we don't get the ES descriptor, try manual config
-  if(!sh->codecdata_len && sh->wf) {
-    sh->codecdata_len = sh->wf->cbSize;
-    sh->codecdata = malloc(sh->codecdata_len);
-    memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len);
-    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
-  }
-  if(!sh->codecdata_len || sh->format == mmioFOURCC('M', 'P', '4', 'L')) {
-    faacDecConfigurationPtr faac_conf;
-    /* Set the default object type and samplerate */
-    /* This is useful for RAW AAC files */
-    faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
-    if(sh->samplerate)
-      faac_conf->defSampleRate = sh->samplerate;
-    /* XXX: FAAD support FLOAT output, how do we handle
-      * that (FAAD_FMT_FLOAT)? ::atmos
-      */
-    if (opts->audio_output_channels <= 2)
-        faac_conf->downMatrix = 1;
-      switch(sh->samplesize){
-	case 1: // 8Bit
-	  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
-	default:
-	  sh->samplesize=2;
-	case 2: // 16Bit
-	  faac_conf->outputFormat = FAAD_FMT_16BIT;
-	  break;
-	case 3: // 24Bit
-	  faac_conf->outputFormat = FAAD_FMT_24BIT;
-	  break;
-	case 4: // 32Bit
-	  faac_conf->outputFormat = FAAD_FMT_32BIT;
-	  break;
-      }
-    //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.
-
-    faacDecSetConfiguration(faac_hdec, faac_conf);
-
-    sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
-    if (!sh->a_in_buffer_len) {
-      // faad init will crash with 0 buffer length
-      mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Could not get audio data!\n");
-      return 0;
-    }
-    /* external faad does not have latm lookup support */
-    faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
-                            sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
-
-    if (faac_init < 0) {
-    pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
-    if(pos) {
-      sh->a_in_buffer_len -= pos;
-      memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
-      sh->a_in_buffer_len +=
-	demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
-	sh->a_in_buffer_size - sh->a_in_buffer_len);
-      pos = 0;
-    }
-
-    /* init the codec */
-    faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
-          sh->a_in_buffer_len, &faac_samplerate, &faac_channels);
-    }
-
-    sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
-    // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi
-
-  } else { // We have ES DS in codecdata
-    faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
-    if (opts->audio_output_channels <= 2) {
-        faac_conf->downMatrix = 1;
-        faacDecSetConfiguration(faac_hdec, faac_conf);
-    }
-
-    /*int i;
-    for(i = 0; i < sh_audio->codecdata_len; i++)
-      printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
-
-    faac_init = faacDecInit2(faac_hdec, sh->codecdata,
-       sh->codecdata_len,	&faac_samplerate, &faac_channels);
-  }
-  if(faac_init < 0) {
-    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
-    faacDecClose(faac_hdec);
-    // XXX: free a_in_buffer here or in uninit?
-    return 0;
-  } else {
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz  channels: %d\n", faac_samplerate, faac_channels);
-    // 8 channels is aac channel order #7.
-    sh->channels = faac_channels == 7 ? 8 : faac_channels;
-    if (opts->audio_output_channels <= 2)
-        sh->channels = faac_channels > 1 ? 2 : 1;
-    sh->samplerate = faac_samplerate;
-    sh->samplesize=2;
-    //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
-    if(!sh->i_bps) {
-      mp_msg(MSGT_DECAUDIO, MSGL_V, "FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
-      sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
-    } else
-      mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
-  }
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-  mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n");
-  faacDecClose(faac_hdec);
-}
-
-static int aac_sync(sh_audio_t *sh)
-{
-  int pos = 0;
-  // do not probe LATM, faad does that
-  if(!sh->codecdata_len && sh->format != mmioFOURCC('M', 'P', '4', 'L')) {
-    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
-      sh->a_in_buffer_len +=
-	demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
-	sh->a_in_buffer_size - sh->a_in_buffer_len);
-    }
-    pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
-    if(pos) {
-      sh->a_in_buffer_len -= pos;
-      memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
-      mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
-    }
-  }
-  return pos;
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-    switch(cmd)
-    {
-      case ADCTRL_RESYNC_STREAM:
-         aac_sync(sh);
-	 return CONTROL_TRUE;
-#if 0
-      case ADCTRL_SKIP_FRAME:
-	  return CONTROL_TRUE;
-#endif
-    }
-  return CONTROL_UNKNOWN;
-}
-
-#define MAX_FAAD_ERRORS 10
-static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
-{
-  int len = 0, last_dec_len = 1, errors = 0;
-  //  int j = 0;
-  void *faac_sample_buffer;
-
-  while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {
-
-    /* update buffer for raw aac streams: */
-  if(!sh->codecdata_len)
-    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
-      sh->a_in_buffer_len +=
-	demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
-	sh->a_in_buffer_size - sh->a_in_buffer_len);
-    }
-
-#ifdef DUMP_AAC_COMPRESSED
-    {int i;
-    for (i = 0; i < 16; i++)
-      printf ("%02X ", sh->a_in_buffer[i]);
-    printf ("\n");}
-#endif
-
-  if(!sh->codecdata_len){
-   // raw aac stream:
-   do {
-    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
-
-    /* update buffer index after faacDecDecode */
-    if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
-      sh->a_in_buffer_len=0;
-    } else {
-      sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
-      memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
-    }
-
-    if(faac_finfo.error > 0) {
-      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
-              faacDecGetErrorMessage(faac_finfo.error));
-      if (sh->a_in_buffer_len <= 0) {
-        errors = MAX_FAAD_ERRORS;
-        break;
-      }
-      sh->a_in_buffer_len--;
-      memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
-      aac_sync(sh);
-      errors++;
-    } else
-      break;
-   } while(errors < MAX_FAAD_ERRORS);
-  } else {
-   // packetized (.mp4) aac stream:
-    unsigned char* bufptr=NULL;
-    double pts;
-    int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
-    if(buflen<=0) break;
-    if (pts != MP_NOPTS_VALUE) {
-	sh->pts = pts;
-	sh->pts_bytes = 0;
-    }
-    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
-  }
-  //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
-
-    if(faac_finfo.error > 0) {
-      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
-      faacDecGetErrorMessage(faac_finfo.error));
-    } else if (faac_finfo.samples == 0) {
-      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
-    } else {
-      /* XXX: samples already multiplied by channels! */
-      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
-      sh->samplesize*faac_finfo.samples);
-
-      if (sh->channels >= 5)
-        reorder_channel_copy_nch(faac_sample_buffer,
-                                 AF_CHANNEL_LAYOUT_AAC_DEFAULT,
-                                 buf+len, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
-                                 sh->channels,
-                                 faac_finfo.samples, sh->samplesize);
-      else
-      memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
-      last_dec_len = sh->samplesize*faac_finfo.samples;
-      len += last_dec_len;
-      sh->pts_bytes += last_dec_len;
-    //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
-    }
-  }
-  return len;
-}
diff --git a/libmpcodecs/ad_hwac3.c b/libmpcodecs/ad_hwac3.c
deleted file mode 100644
index 44e7657107..0000000000
--- a/libmpcodecs/ad_hwac3.c
+++ /dev/null
@@ -1,577 +0,0 @@
-/*
- * DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
- * (see http://www.dtek.chalmers.se/~dvd/)
- * Reference: DOCS/tech/hwac3.txt !!!!!
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#define _GNU_SOURCE
-#define _XOPEN_SOURCE 600
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <unistd.h>
-#include <sys/types.h>
-
-#include <libavutil/intreadwrite.h>
-#include <libavutil/common.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "mpbswap.h"
-
-#include "ad_internal.h"
-
-
-static int isdts = -1;
-
-static const ad_info_t info =
-{
-  "AC3/DTS pass-through S/PDIF",
-  "hwac3",
-  "Nick Kurshev/Peter Schüller",
-  "???",
-  ""
-};
-
-LIBAD_EXTERN(hwac3)
-
-
-static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
-static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);
-
-
-static int a52_syncinfo (uint8_t *buf, int *sample_rate, int *bit_rate)
-{
-    static const uint16_t rate[] = { 32,  40,  48,  56,  64,  80,  96, 112,
-                                    128, 160, 192, 224, 256, 320, 384, 448,
-                                    512, 576, 640};
-    int frmsizecod;
-    int bitrate;
-    int half;
-
-    if (buf[0] != 0x0b || buf[1] != 0x77)    /* syncword */
-        return 0;
-
-    if (buf[5] >= 0x60)                      /* bsid >= 12 */
-        return 0;
-    half = buf[5] >> 3;
-    half = FFMAX(half - 8, 0);
-
-    frmsizecod = buf[4] & 63;
-    if (frmsizecod >= 38)
-        return 0;
-    bitrate = rate[frmsizecod >> 1];
-    *bit_rate = (bitrate * 1000) >> half;
-
-    switch (buf[4] & 0xc0) {
-    case 0:
-        *sample_rate = 48000 >> half;
-        return 4 * bitrate;
-    case 0x40:
-        *sample_rate = 44100 >> half;
-        return 2 * (320 * bitrate / 147 + (frmsizecod & 1));
-    case 0x80:
-        *sample_rate = 32000 >> half;
-        return 6 * bitrate;
-    default:
-        return 0;
-    }
-}
-
-static int ac3dts_fillbuff(sh_audio_t *sh_audio)
-{
-  int length = 0;
-  int flags = 0;
-  int sample_rate = 0;
-  int bit_rate = 0;
-
-  sh_audio->a_in_buffer_len = 0;
-  /* sync frame:*/
-  while(1)
-  {
-    // Original code DTS has a 10 bytes header.
-    // Now max 12 bytes for 14 bits DTS header.
-    while(sh_audio->a_in_buffer_len < 12)
-    {
-      int c = demux_getc(sh_audio->ds);
-      if(c<0)
-        return -1; /* EOF*/
-      sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++] = c;
-    }
-
-    if (sh_audio->format == 0x2001)
-    {
-      length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
-      if(length >= 12)
-      {
-        if(isdts != 1)
-        {
-          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
-          isdts = 1;
-        }
-        break;
-      }
-    }
-    else
-    {
-      length = a52_syncinfo(sh_audio->a_in_buffer, &sample_rate, &bit_rate);
-      if(length >= 7 && length <= 3840)
-      {
-        if(isdts != 0)
-        {
-          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
-          isdts = 0;
-        }
-        break; /* we're done.*/
-      }
-    }
-    /* bad file => resync*/
-    memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
-    --sh_audio->a_in_buffer_len;
-  }
-  mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "ac3dts: %s len=%d  flags=0x%X  %d Hz %d bit/s\n", isdts == 1 ? "DTS" : isdts == 0 ? "AC3" : "unknown", length, flags, sample_rate, bit_rate);
-
-  sh_audio->samplerate = sample_rate;
-  sh_audio->i_bps = bit_rate / 8;
-  demux_read_data(sh_audio->ds, sh_audio->a_in_buffer + 12, length - 12);
-  sh_audio->a_in_buffer_len = length;
-
-  return length;
-}
-
-
-static int preinit(sh_audio_t *sh)
-{
-  /* Dolby AC3 audio: */
-  sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
-  sh->audio_in_minsize = 8192;
-  sh->channels = 2;
-  sh->samplesize = 2;
-  sh->sample_format = AF_FORMAT_AC3_BE;
-  // HACK for DTS where useless swapping can't easily be removed
-  if (sh->format == 0x2001)
-    sh->sample_format = AF_FORMAT_AC3_NE;
-  return 1;
-}
-
-static int init(sh_audio_t *sh_audio)
-{
-  /* Dolby AC3 passthrough:*/
-  if(ac3dts_fillbuff(sh_audio) < 0)
-  {
-    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
-    return 0;
-  }
-  return 1;
-}
-
-static void uninit(sh_audio_t *sh)
-{
-}
-
-static int control(sh_audio_t *sh,int cmd,void* arg, ...)
-{
-  switch(cmd)
-  {
-  case ADCTRL_RESYNC_STREAM:
-  case ADCTRL_SKIP_FRAME:
-      ac3dts_fillbuff(sh);
-      return CONTROL_TRUE;
-  }
-  return CONTROL_UNKNOWN;
-}
-
-
-static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
-{
-  int len = sh_audio->a_in_buffer_len;
-
-  if(len <= 0)
-    if((len = ac3dts_fillbuff(sh_audio)) <= 0)
-      return len; /*EOF*/
-  sh_audio->a_in_buffer_len = 0;
-
-  if(isdts == 1)
-  {
-    return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
-  }
-  else if(isdts == 0)
-  {
-    AV_WB16(buf,     0xF872);   // iec 61937 syncword 1
-    AV_WB16(buf + 2, 0x4E1F);   // iec 61937 syncword 2
-    buf[4] = sh_audio->a_in_buffer[5] & 0x7; // bsmod
-    buf[5] = 0x01;              // data-type ac3
-    AV_WB16(buf + 6, len << 3); // number of bits in payload
-    memcpy(buf + 8, sh_audio->a_in_buffer, len);
-    memset(buf + 8 + len, 0, 6144 - 8 - len);
-
-    return 6144;
-  }
-  else
-    return -1;
-}
-
-
-static const int DTS_SAMPLEFREQS[16] =
-{
-  0,
-  8000,
-  16000,
-  32000,
-  64000,
-  128000,
-  11025,
-  22050,
-  44100,
-  88200,
-  176400,
-  12000,
-  24000,
-  48000,
-  96000,
-  192000
-};
-
-static const int DTS_BITRATES[30] =
-{
-  32000,
-  56000,
-  64000,
-  96000,
-  112000,
-  128000,
-  192000,
-  224000,
-  256000,
-  320000,
-  384000,
-  448000,
-  512000,
-  576000,
-  640000,
-  768000,
-  896000,
-  1024000,
-  1152000,
-  1280000,
-  1344000,
-  1408000,
-  1411200,
-  1472000,
-  1536000,
-  1920000,
-  2048000,
-  3072000,
-  3840000,
-  4096000
-};
-
-static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
-{
-  int ftype;
-  int surp;
-  int unknown_bit av_unused;
-  int fsize;
-  int amode av_unused;
-
-  int word_mode;
-  int le_mode;
-
-  unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
-