imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5341 b3059339-0415-0410-9bf9-f77b7e298cf2

15 files changed, 1176 insertions, 0 deletions

diff --git a/libmpcodecs/ad_a52.c b/libmpcodecs/ad_a52.c
new file mode 100644
index 0000000000..ca6eea2398
--- /dev/null
+++ b/libmpcodecs/ad_a52.c
@@ -0,0 +1,194 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "help_mp.h"
+
+#include "ad_internal.h"
+
+#include "cpudetect.h"
+
+#include "../liba52/a52.h"
+#include "../liba52/mm_accel.h"
+
+sample_t * a52_samples;
+a52_state_t a52_state;
+uint32_t a52_accel=0;
+uint32_t a52_flags=0;
+
+#include "bswap.h"
+
+static ad_info_t info = 
+{
+	"AC3-liba52",
+	"liba52",
+	AFM_A52,
+	"Nick Kurshev",
+	"Michel LESPINASSE",
+	""
+};
+
+LIBAD_EXTERN(a52)
+
+extern int audio_output_channels;
+
+int a52_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){
+    while(sh_audio->a_in_buffer_len<7){
+	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;
+    }
+    length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
+    if(length>=7 && length<=3840) break; /* we're done.*/
+    /* bad file => resync*/
+    memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6);
+    --sh_audio->a_in_buffer_len;
+}
+    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d  flags=0x%X  %d Hz %d bit/s\n",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+7,length-7);
+    
+    if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
+	mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed!  \n");
+    
+    return length;
+}
+
+/* returns: number of available channels*/
+static int a52_printinfo(sh_audio_t *sh_audio){
+int flags, sample_rate, bit_rate;
+char* mode="unknown";
+int channels=0;
+  a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
+  switch(flags&A52_CHANNEL_MASK){
+    case A52_CHANNEL: mode="channel"; channels=2; break;
+    case A52_MONO: mode="mono"; channels=1; break;
+    case A52_STEREO: mode="stereo"; channels=2; break;
+    case A52_3F: mode="3f";channels=3;break;
+    case A52_2F1R: mode="2f+1r";channels=3;break;
+    case A52_3F1R: mode="3f+1r";channels=4;break;
+    case A52_2F2R: mode="2f+2r";channels=4;break;
+    case A52_3F2R: mode="3f+2r";channels=5;break;
+    case A52_CHANNEL1: mode="channel1"; channels=2; break;
+    case A52_CHANNEL2: mode="channel2"; channels=2; break;
+    case A52_DOLBY: mode="dolby"; channels=2; break;
+  }
+  mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s)  %d Hz  %3.1f kbit/s\n",
+	channels, (flags&A52_LFE)?1:0,
+	mode, (flags&A52_LFE)?"+lfe":"",
+	sample_rate, bit_rate*0.001f);
+  return (flags&A52_LFE) ? (channels+1) : channels;
+}
+
+
+static int preinit(sh_audio_t *sh)
+{
+  /* Dolby AC3 audio: */
+  /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
+  sh->audio_out_minsize=audio_output_channels*2*256*6;
+  return 1;
+}
+
+static int init(sh_audio_t *sh_audio)
+{
+  sample_t level=1, bias=384;
+  int flags=0;
+  /* Dolby AC3 audio:*/
+  if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
+  if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
+  if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
+  if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
+  if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
+  a52_samples=a52_init (a52_accel);
+  if (a52_samples == NULL) {
+	mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
+	return 0;
+  }
+   sh_audio->a_in_buffer_size=3840;
+   sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
+   sh_audio->a_in_buffer_len=0;
+  if(a52_fillbuff(sh_audio)<0){
+	mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
+	return 0;
+  }
+  /* 'a52 cannot upmix' hotfix:*/
+  a52_printinfo(sh_audio);
+  sh_audio->channels=audio_output_channels;
+while(sh_audio->channels>0){
+  switch(sh_audio->channels){
+	    case 1: a52_flags=A52_MONO; break;
+/*	    case 2: a52_flags=A52_STEREO; break;*/
+	    case 2: a52_flags=A52_DOLBY; break;
+/*	    case 3: a52_flags=A52_3F; break;*/
+	    case 3: a52_flags=A52_2F1R; break;
+	    case 4: a52_flags=A52_2F2R; break; /* 2+2*/
+	    case 5: a52_flags=A52_3F2R; break;
+	    case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
+  }
+  /* test:*/
+  flags=a52_flags|A52_ADJUST_LEVEL;
+  mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
+  if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
+    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
+    return 0;
+  }
+  mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
+  /* frame decoded, let's init resampler:*/
+  if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
+  --sh_audio->channels; /* try to decrease no. of channels*/
+}
+  if(sh_audio->channels<=0){
+    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\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_SKIP_FRAME:
+	  a52_fillbuff(sh); break; // skip AC3 frame
+	  return CONTROL_TRUE;
+    }
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+    sample_t level=1, bias=384;
+    int flags=a52_flags|A52_ADJUST_LEVEL;
+    int i,len=-1;
+	if(!sh_audio->a_in_buffer_len) 
+	    if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
+	sh_audio->a_in_buffer_len=0;
+	if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
+	    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
+	    return len;
+	}
+	len=0;
+	for (i = 0; i < 6; i++) {
+	    if (a52_block (&a52_state, a52_samples)){
+		mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
+		break;
+	    }
+	    len+=2*a52_resample(a52_samples,&buf[len]);
+	}
+  return len;
+}
diff --git a/libmpcodecs/ad_acm.c b/libmpcodecs/ad_acm.c
new file mode 100644
index 0000000000..5a74071489
--- /dev/null
+++ b/libmpcodecs/ad_acm.c
@@ -0,0 +1,76 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "help_mp.h"
+
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"Win32 ACM audio decoder",
+	"msacm",
+	AFM_ACM,
+	"Nick Kurshev",
+	"avifile.sf.net",
+	""
+};
+
+LIBAD_EXTERN(msacm)
+
+#include "dll_init.h"
+
+static int init(sh_audio_t *sh_audio)
+{
+    int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size);
+    if(ret<0){
+        mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret);
+        return 0;
+    }
+    sh_audio->a_buffer_len=ret;
+  return 1;
+}
+
+static int preinit(sh_audio_t *sh_audio)
+{
+  /* Win32 ACM audio codec: */
+  if(init_acm_audio_codec(sh_audio)){
+    sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
+    sh_audio->channels=sh_audio->o_wf.nChannels;
+    sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec;
+  } else {
+    mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror);
+    return 0;
+  }
+  mp_msg(MSGT_DECVIDEO,MSGL_V,"INFO: Win32/ACM audio codec init OK!\n");
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    // TODO!
+}
+
+static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
+{
+  int skip;
+    switch(cmd)
+    {
+      case ADCTRL_SKIP_FRAME:
+		    skip=sh_audio->wf->nBlockAlign;
+		    if(skip<16){
+		      skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7);
+		      if(skip<16) skip=16;
+		    }
+		    demux_read_data(sh_audio->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)
+{
+  return acm_decode_audio(sh_audio,buf,minlen,maxlen);
+}
diff --git a/libmpcodecs/ad_alaw.c b/libmpcodecs/ad_alaw.c
new file mode 100644
index 0000000000..d1a7e4cfe4
--- /dev/null
+++ b/libmpcodecs/ad_alaw.c
@@ -0,0 +1,73 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"aLaw/uLaw audio decoder",
+	"alaw",
+	AFM_ALAW,
+	"Nick Kurshev",
+	"A'rpi",
+	""
+};
+
+LIBAD_EXTERN(alaw)
+
+#include "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;
+  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;
+      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){
+ /* 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_dk3adpcm.c b/libmpcodecs/ad_dk3adpcm.c
new file mode 100644
index 0000000000..a6de3d96e1
--- /dev/null
+++ b/libmpcodecs/ad_dk3adpcm.c
@@ -0,0 +1,60 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"Duck DK3 ADPCM decoder",
+	"dk3adpcm",
+	AFM_DK3ADPCM,
+	"Nick Kurshev",
+	"Mike Melanson",
+	"This format number was used by Duck Corp. but not officially registered with Microsoft"
+};
+
+LIBAD_EXTERN(dk3adpcm)
+
+#include "adpcm.h"
+
+static int init(sh_audio_t *sh_audio)
+{
+  sh_audio->channels=sh_audio->wf->nChannels;
+  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
+  sh_audio->i_bps=DK3_ADPCM_BLOCK_SIZE*
+    (sh_audio->channels*sh_audio->samplerate) / DK3_ADPCM_SAMPLES_PER_BLOCK;
+  return 1;
+}
+
+static int preinit(sh_audio_t *sh_audio)
+{
+  sh_audio->audio_out_minsize=DK3_ADPCM_SAMPLES_PER_BLOCK * 4;
+  sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK;
+  sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE;
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    // TODO!
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+  int len=-1;
+   unsigned char ibuf[DK3_ADPCM_BLOCK_SIZE * 2]; /* bytes / stereo frame */
+    if (demux_read_data(sh_audio->ds, ibuf,
+          DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != 
+          DK3_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) 
+      return len; /* EOF */
+    len = 2 * dk3_adpcm_decode_block(
+          (unsigned short*)buf,ibuf);
+    return len;
+}
diff --git a/libmpcodecs/ad_dk4adpcm.c b/libmpcodecs/ad_dk4adpcm.c
new file mode 100644
index 0000000000..c647f984fa
--- /dev/null
+++ b/libmpcodecs/ad_dk4adpcm.c
@@ -0,0 +1,60 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"Duck DK4 ADPCM (rogue format number) audio decoder",
+	"dk4adpcm",
+	AFM_DK4ADPCM,
+	"Nick Kurshev",
+	"This format number was used by Duck Corp. but not officially registered with Microsoft"
+};
+
+LIBAD_EXTERN(dk4adpcm)
+
+#include "adpcm.h"
+
+static int init(sh_audio_t *sh_audio)
+{
+  sh_audio->channels=sh_audio->wf->nChannels;
+  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
+  sh_audio->i_bps = sh_audio->wf->nBlockAlign *
+    (sh_audio->channels*sh_audio->samplerate) / DK4_ADPCM_SAMPLES_PER_BLOCK;
+  return 1;
+}
+
+static int preinit(sh_audio_t *sh_audio)
+{
+  sh_audio->audio_out_minsize=DK4_ADPCM_SAMPLES_PER_BLOCK * 4;
+  sh_audio->ds->ss_div=DK4_ADPCM_SAMPLES_PER_BLOCK;
+  sh_audio->ds->ss_mul=sh_audio->wf->nBlockAlign;
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    // TODO!
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+  int len=-1;
+  static unsigned char *ibuf = NULL;
+  if (!ibuf)
+    ibuf = (unsigned char *)malloc(sh_audio->wf->nBlockAlign);
+  if (demux_read_data(sh_audio->ds, ibuf, sh_audio->wf->nBlockAlign) != 
+      sh_audio->wf->nBlockAlign)
+          return len; /* EOF */
+  len=2*dk4_adpcm_decode_block((unsigned short*)buf,ibuf,
+          sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
+  return len;
+}
diff --git a/libmpcodecs/ad_dshow.c b/libmpcodecs/ad_dshow.c
new file mode 100644
index 0000000000..369368e7e2
--- /dev/null
+++ b/libmpcodecs/ad_dshow.c
@@ -0,0 +1,99 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "help_mp.h"
+
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"Win32/DirectShow decoders",
+	"dshow",
+	AFM_DSHOW,
+	"Nick Kurshev",
+	"avifile.sf.net",
+	""
+};
+
+LIBAD_EXTERN(dshow)
+
+#include "dshow/DS_AudioDecoder.h"
+
+static DS_AudioDecoder* ds_adec=NULL;
+
+static int init(sh_audio_t *sh)
+{
+  return 1;
+}
+
+static int preinit(sh_audio_t *sh_audio)
+{
+  if(!(ds_adec=DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf)))
+  {
+    mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll);
+    return 0;
+  } else {
+    sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
+    sh_audio->channels=sh_audio->wf->nChannels;
+    sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
+    sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign;
+    if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192;
+    sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
+    sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
+    sh_audio->a_in_buffer_len=0;
+    sh_audio->audio_out_minsize=16384;
+  }
+  mp_msg(MSGT_DECVIDEO,MSGL_V,"INFO: Win32/DShow audio codec init OK!\n");
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    // TODO!!!
+}
+
+static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
+{
+  int skip;
+    switch(cmd)
+    {
+      case ADCTRL_SKIP_FRAME:
+		    skip=sh_audio->wf->nBlockAlign;
+		    if(skip<16){
+		      skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7);
+		      if(skip<16) skip=16;
+		    }
+		    demux_read_data(sh_audio->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 len=-1;
+        int size_in=0;
+        int size_out=0;
+        int srcsize=DS_AudioDecoder_GetSrcSize(ds_adec, maxlen);
+        mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DShow says: srcsize=%d  (buffsize=%d)  out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen);
+        if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!!
+        if(sh_audio->a_in_buffer_len<srcsize){
+          sh_audio->a_in_buffer_len+=
+            demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len],
+            srcsize-sh_audio->a_in_buffer_len);
+        }
+        DS_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len,
+            buf,maxlen, &size_in,&size_out);
+        mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DShow: audio %d -> %d converted  (in_buf_len=%d of %d)  %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds));
+        if(size_in>=sh_audio->a_in_buffer_len){
+          sh_audio->a_in_buffer_len=0;
+        } else {
+          sh_audio->a_in_buffer_len-=size_in;
+          memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len);
+        }
+//        len=size_out;
+  return size_out;
+}
diff --git a/libmpcodecs/ad_dvdpcm.c b/libmpcodecs/ad_dvdpcm.c
new file mode 100644
index 0000000000..3a08db8588
--- /dev/null
+++ b/libmpcodecs/ad_dvdpcm.c
@@ -0,0 +1,64 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"Uncompressed DVD PCM audio decoder",
+	"dvdpcm",
+	AFM_DVDPCM,
+	"Nick Kurshev",
+	"A'rpi",
+	""
+};
+
+LIBAD_EXTERN(dvdpcm)
+
+static int init(sh_audio_t *sh)
+{
+/* DVD PCM Audio:*/
+    sh->channels=2;
+    sh->samplerate=48000;
+    sh->i_bps=2*2*48000;
+/*    sh_audio->pcm_bswap=1;*/
+  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;
+  len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
+  for(j=0;j<len;j+=2){
+    char x=buf[j];
+    buf[j]=buf[j+1];
+    buf[j+1]=x;
+  }
+  return len;
+}
diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
new file mode 100644
index 0000000000..9929ca2802
--- /dev/null
+++ b/libmpcodecs/ad_ffmpeg.c
@@ -0,0 +1,106 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "help_mp.h"
+
+#include "ad_internal.h"
+
+#include "bswap.h"
+
+static ad_info_t info = 
+{
+	"FFmpeg audio decoders",
+	"ffmpeg",
+	AFM_FFMPEG,
+	"Nick Kurshev",
+	"ffmpeg.sf.net",
+	""
+};
+
+LIBAD_EXTERN(ffmpeg)
+
+#define assert(x)
+#include "../libavcodec/avcodec.h"
+
+static AVCodec *lavc_codec=NULL;
+static AVCodecContext lavc_context;
+extern int avcodec_inited;
+
+static int preinit(sh_audio_t *sh)
+{
+  sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
+  return 1;
+}
+
+static int init(sh_audio_t *sh_audio)
+{
+   int x;
+   mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
+    if(!avcodec_inited){
+      avcodec_init();
+      avcodec_register_all();
+      avcodec_inited=1;
+    }
+    lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
+    if(!lavc_codec){
+	mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
+	return 0;
+    }
+    memset(&lavc_context, 0, sizeof(lavc_context));
+    /* open it */
+    if (avcodec_open(&lavc_context, lavc_codec) < 0) {
+        mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
+        return 0;
+    }
+   mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
+
+   // Decode at least 1 byte:  (to get header filled)
+   x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
+   if(x>0) sh_audio->a_buffer_len=x;
+
+#if 1
+  sh_audio->channels=lavc_context.channels;
+  sh_audio->samplerate=lavc_context.sample_rate;
+  sh_audio->i_bps=lavc_context.bit_rate/8;
+#else
+  sh_audio->channels=sh_audio->wf->nChannels;
+  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
+  sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
+#endif
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+  avcodec_close(&lavc_context);
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    // TODO ???
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+    unsigned char *start=NULL;
+    int y,len=-1;
+    while(len<minlen){
+	int len2=0;
+	int x=ds_get_packet(sh_audio->ds,&start);
+	if(x<=0) break; // error
+	y=avcodec_decode_audio(&lavc_context,(INT16*)buf,&len2,start,x);
+	if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
+	if(y<x) sh_audio->ds->buffer_pos+=y-x;  // put back data (HACK!)
+	if(len2>0){
+	  //len=len2;break;
+	  if(len<0) len=len2; else len+=len2;
+	  buf+=len2;
+	}
+        mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d  \n",y,len2);
+    }
+  return len;
+}
diff --git a/libmpcodecs/ad_hwac3.c b/libmpcodecs/ad_hwac3.c
new file mode 100644
index 0000000000..fb92d5b593
--- /dev/null
+++ b/libmpcodecs/ad_hwac3.c
@@ -0,0 +1,88 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "help_mp.h"
+
+#include "ad_internal.h"
+
+#include "../liba52/a52.h"
+#include "../ac3-iec958.h"
+
+extern int a52_fillbuff(sh_audio_t *sh_audio);
+
+static ad_info_t info = 
+{
+	"AC3 through SPDIF",
+	"hwac3",
+	AFM_HWAC3,
+	"Nick Kurshev",
+	"???",
+	""
+};
+
+LIBAD_EXTERN(hwac3)
+
+static int preinit(sh_audio_t *sh)
+{
+  /* Dolby AC3 audio: */
+  sh->audio_out_minsize=4*256*6;
+  sh->channels=2;
+  return 1;
+}
+
+static int init(sh_audio_t *sh_audio)
+{
+  /* Dolby AC3 passthrough:*/
+  sample_t *a52_samples=a52_init(0);
+  if (a52_samples == NULL) {
+       mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
+       return 0;
+  }
+  sh_audio->a_in_buffer_size=3840;
+  sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size);
+  sh_audio->a_in_buffer_len=0;
+  if(a52_fillbuff(sh_audio)<0) {
+       mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
+       return 0;
+  }
+ /* 
+  sh_audio->samplerate=ai.samplerate;   // SET by a52_fillbuff()
+  sh_audio->samplesize=ai.framesize;
+  sh_audio->i_bps=ai.bitrate*(1000/8);  // SET by a52_fillbuff()
+  sh_audio->ac3_frame=malloc(6144);
+  sh_audio->o_bps=sh_audio->i_bps;  // XXX FIXME!!! XXX
+
+   o_bps is calculated from samplesize*channels*samplerate
+   a single ac3 frame is always translated to 6144 byte packet. (zero padding)*/
+  sh_audio->channels=2;
+  sh_audio->samplesize=2;   /* 2*2*(6*256) = 6144 (very TRICKY!)*/
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    switch(cmd)
+    {
+      case ADCTRL_SKIP_FRAME:
+	  a52_fillbuff(sh); break; // skip AC3 frame
+	  return CONTROL_TRUE;
+    }
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+  int len=-1;
+  if(!sh_audio->a_in_buffer_len)
+    if((len=a52_fillbuff(sh_audio))<0) return len; /*EOF*/
+  sh_audio->a_in_buffer_len=0;
+  len = ac3_iec958_build_burst(len, 0x01, 1, sh_audio->a_in_buffer, buf);
+  return len;
+}
diff --git a/libmpcodecs/ad_imaadpcm.c b/libmpcodecs/ad_imaadpcm.c
new file mode 100644
index 0000000000..eca064d6c5
--- /dev/null
+++ b/libmpcodecs/ad_imaadpcm.c
@@ -0,0 +1,59 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+#include "../adpcm.h"
+
+static ad_info_t info = 
+{
+	"IMA ADPCM audio decoder",
+	"imaadpcm",
+	AFM_IMAADPCM,
+	"Nick Kurshev",
+	"Mike Melanson",
+	"ima4 (MOV files)"
+};
+
+LIBAD_EXTERN(imaadpcm)
+
+static int init(sh_audio_t *sh_audio)
+{
+  /* IMA-ADPCM 4:1 audio codec:*/
+  sh_audio->channels=sh_audio->wf->nChannels;
+  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
+  /* decodes 34 byte -> 64 short*/
+  sh_audio->i_bps=IMA_ADPCM_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/
+		  IMA_ADPCM_SAMPLES_PER_BLOCK;  /* 1:4 */
+  return 1;
+}
+
+static int preinit(sh_audio_t *sh_audio)
+{
+  sh_audio->audio_out_minsize=4096;
+  sh_audio->ds->ss_div=IMA_ADPCM_SAMPLES_PER_BLOCK;
+  sh_audio->ds->ss_mul=IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels;
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+    // TODO!!!
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+    unsigned char ibuf[IMA_ADPCM_BLOCK_SIZE * 2]; /* bytes / stereo frame */
+    if (demux_read_data(sh_audio->ds, ibuf,
+      IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) != 
+      IMA_ADPCM_BLOCK_SIZE * sh_audio->wf->nChannels) 
+	return -1;
+    return 2*ima_adpcm_decode_block((unsigned short*)buf,ibuf, sh_audio->wf->nChannels);
+}
diff --git a/libmpcodecs/ad_internal.h b/libmpcodecs/ad_internal.h
new file mode 100644
index 0000000000..e5b67f5844
--- /dev/null
+++ b/libmpcodecs/ad_internal.h
@@ -0,0 +1,25 @@
+
+#include "codec-cfg.h"
+#include "../libao2/afmt.h"
+
+#include "stream.h"
+#include "demuxer.h"
+#include "stheader.h"
+
+#include "ad.h"
+
+static int init(sh_audio_t *sh);
+static int preinit(sh_audio_t *sh);
+static void uninit(sh_audio_t *sh);
+static int control(sh_audio_t *sh,int cmd,void* arg, ...);
+static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
+
+#define LIBAD_EXTERN(x) ad_functions_t mpcodecs_ad_##x = {\
+	&info,\
+	preinit,\
+	init,\
+        uninit,\
+	control,\
+	decode_audio\
+};
+
diff --git a/libmpcodecs/ad_mp3.c b/libmpcodecs/ad_mp3.c
new file mode 100644
index 0000000000..683ce09771
--- /dev/null
+++ b/libmpcodecs/ad_mp3.c
@@ -0,0 +1,79 @@
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "ad_internal.h"
+
+static ad_info_t info = 
+{
+	"MPEG layer-2, layer-3",
+	"mp3lib",
+	AFM_MPEG,
+	"Nick Kurshev",
+	"mpg123",
+	"Optimized to MMX/SSE/3Dnow!"
+};
+
+LIBAD_EXTERN(mp3)
+
+#include "../mp3lib/mp3.h"
+
+extern int fakemono;
+
+static sh_audio_t* dec_audio_sh=NULL;
+
+// MP3 decoder buffer callback:
+int mplayer_audio_read(char *buf,int size){
+  return demux_read_data(dec_audio_sh->ds,buf,size);
+}
+
+static int preinit(sh_audio_t *sh)
+{
+  sh->audio_out_minsize=32*36*2*2; //4608;
+  return 1;
+}
+
+static int init(sh_audio_t *sh)
+{
+  // MPEG Audio:
+  dec_audio_sh=sh; // save sh_audio for the callback:
+//  MP3_Init(fakemono,mplayer_accel,&mplayer_audio_read); // TODO!!!
+#ifdef USE_FAKE_MONO
+  MP3_Init(fakemono);
+#else
+  MP3_Init();
+#endif
+  MP3_samplerate=MP3_channels=0;
+  sh->a_buffer_len=MP3_DecodeFrame(sh->a_buffer,-1);
+  sh->channels=2; // hack
+  sh->samplerate=MP3_samplerate;
+  sh->i_bps=MP3_bitrate*(1000/8);
+  MP3_PrintHeader();
+  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:
+          MP3_DecodeFrame(NULL,-2); // resync
+          MP3_DecodeFrame(NULL,-2); // resync
+          MP3_DecodeFrame(NULL,-2); // resync
+	  return CONTROL_TRUE;
+      case ADCTRL_SKIP_FRAME:
+	  MP3_DecodeFrame(NULL,-2); // skip MPEG frame
+	  return CONTROL_TRUE;
+    }
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned