[Mplayer-cvslog] CVS: main/libmpcodecs ad_a52.c,NONE,1.1 ad_acm.c,NONE,1.1 ad_alaw.c,NONE,1.1 ad_dk3adpcm.c,NONE,1.1 ad_dk4adpcm.c,NONE,1.1 ad_dshow.c,NONE,1.1 ad_dvdpcm.c,NONE,1.1 ad_ffmpeg.c,NONE,1.1 ad_hwac3.c,NONE,1.1 ad_imaadpcm.c,NONE,1.1 ad_internal.h,NONE,1.1 ad_mp3.c,NONE,1.1 ad_msadpcm.c,NONE,1.1 ad_pcm.c,NONE,1.1 ad_roqaudio.c,NONE,1.1

[Arpi of Ize]

Update of /cvsroot/mplayer/main/libmpcodecs
In directory mplayer:/var/tmp.root/cvs-serv17531

Added Files:
	ad_a52.c ad_acm.c ad_alaw.c ad_dk3adpcm.c ad_dk4adpcm.c 
	ad_dshow.c ad_dvdpcm.c ad_ffmpeg.c ad_hwac3.c ad_imaadpcm.c 
	ad_internal.h ad_mp3.c ad_msadpcm.c ad_pcm.c ad_roqaudio.c 
Log Message:
imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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);
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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;
}

--- NEW FILE ---
#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);
}

--- NEW FILE ---

#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\
};


--- NEW FILE ---
#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 char *buf,int minlen,int maxlen)
{
   return MP3_DecodeFrame(buf,-1);
}

--- NEW FILE ---
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"
#include "ad_internal.h"

static ad_info_t info = 
{
	"MS ADPCM audio decoder",
	"msadpcm",
	AFM_MSADPCM,
	"Nick Kurshev",
	"Mike Melanson",
	""
};

LIBAD_EXTERN(msadpcm)

#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) / MS_ADPCM_SAMPLES_PER_BLOCK;
  return 1;
}

static int preinit(sh_audio_t *sh_audio)
{
  sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign * 8;
  sh_audio->ds->ss_div = MS_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)
{
  static unsigned char *ibuf = NULL;	// FIXME!!! TODO!!! use sh->a_in_buffer!
  if (!ibuf)
   ibuf = (unsigned char *)malloc
        (sh_audio->wf->nBlockAlign * sh_audio->wf->nChannels);
  if (demux_read_data(sh_audio->ds, ibuf,
      sh_audio->wf->nBlockAlign) != 
      sh_audio->wf->nBlockAlign) 
         return -1; /* EOF */
  return 2 * ms_adpcm_decode_block(
          (unsigned short*)buf,ibuf, sh_audio->wf->nChannels,
          sh_audio->wf->nBlockAlign);
}

--- NEW FILE ---
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"
#include "ad_internal.h"

static ad_info_t info = 
{
	"Uncompressed PCM audio decoder",
	"pcm",
	AFM_PCM,
	"Nick Kurshev",
	"A'rpi",
	""
};

LIBAD_EXTERN(pcm)

static int init(sh_audio_t *sh_audio)
{
  WAVEFORMATEX *h=sh_audio->wf;
  sh_audio->i_bps=h->nAvgBytesPerSec;
  sh_audio->channels=h->nChannels;
  sh_audio->samplerate=h->nSamplesPerSec;
  sh_audio->samplesize=(h->wBitsPerSample+7)/8;
  switch(sh_audio->format){ /* hardware formats: */
    case 0x6:  sh_audio->sample_format=AFMT_A_LAW;break;
    case 0x7:  sh_audio->sample_format=AFMT_MU_LAW;break;
    case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break;
    case 0x50: sh_audio->sample_format=AFMT_MPEG;break;
    case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break;
/*    case 0x2000: sh_audio->sample_format=AFMT_AC3; */
    default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8;
  }
  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)
{
  return demux_read_data(sh_audio->ds,buf,minlen);
}

--- NEW FILE ---
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

#include "config.h"
#include "ad_internal.h"
#include "../roqav.h"
//#include "../adpcm.h"

static ad_info_t info = 
{
	"Id RoQ File Audio Decoder",
	"roqaudio",
	AFM_ROQAUDIO,
	"Nick Kurshev",
	"Mike Melanson ???"
	"RoQA is an internal MPlayer FOURCC"
};

LIBAD_EXTERN(roqaudio)

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->channels * 22050) / 2;
  return 1;
}

static int preinit(sh_audio_t *sh_audio)
{
  /* minsize was stored in wf->nBlockAlign by the RoQ demuxer */
  sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign;
  sh_audio->context = roq_decode_audio_init();
  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)
{
  static unsigned char *ibuf = NULL;
  unsigned char header_data[6];
  int read_len;

// TODO!! FIXME!!!
  if (!ibuf) ibuf = (unsigned char *)malloc(sh_audio->audio_out_minsize / 2);
  
  /* figure out how much data to read */
  if (demux_read_data(sh_audio->ds, header_data, 6) != 6) return -1; /* EOF */
  read_len = (header_data[5] << 24) | (header_data[4] << 16) |
	     (header_data[3] << 8) | header_data[2];
  read_len += 2;  /* 16-bit arguments */
  if (demux_read_data(sh_audio->ds, ibuf, read_len) != read_len) return -1;
  return 2 * roq_decode_audio((unsigned short *)buf, ibuf,
			     read_len, sh_audio->channels, sh_audio->context);          
}