unused files

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5601 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 0 insertions, 1444 deletions

diff --git a/dec_audio.c b/dec_audio.c
deleted file mode 100644
index 8cf576a8ca..0000000000
--- a/dec_audio.c
+++ /dev/null
@@ -1,1444 +0,0 @@
-
-#define USE_G72X
-//#define USE_LIBAC3
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-
-#include "config.h"
-#include "mp_msg.h"
-#include "help_mp.h"
-
-extern int verbose; // defined in mplayer.c
-
-#include "stream.h"
-#include "demuxer.h"
-
-#include "codec-cfg.h"
-#include "stheader.h"
-
-#include "dec_audio.h"
-
-#include "roqav.h"
-
-//==========================================================================
-
-#include "libao2/afmt.h"
-
-#include "dll_init.h"
-
-#include "mp3lib/mp3.h"
-
-#ifdef USE_LIBAC3
-#include "libac3/ac3.h"
-#endif
-
-#include "liba52/a52.h"
-#include "liba52/mm_accel.h"
-static sample_t * a52_samples;
-static a52_state_t a52_state;
-static uint32_t a52_accel=0;
-static uint32_t a52_flags=0;
-
-#ifdef USE_G72X
-#include "g72x/g72x.h"
-static G72x_DATA g72x_data;
-#endif
-
-#include "alaw.h"
-
-#include "xa/xa_gsm.h"
-
-#include "ac3-iec958.h"
-
-#include "adpcm.h"
-
-#include "cpudetect.h"
-
-/* used for ac3surround decoder - set using -channels option */
-int audio_output_channels = 2;
-
-#ifdef USE_FAKE_MONO
-int fakemono=0;
-#endif
-
-#ifdef USE_DIRECTSHOW
-#include "loader/dshow/DS_AudioDecoder.h"
-static DS_AudioDecoder* ds_adec=NULL;
-#endif
-
-#ifdef HAVE_OGGVORBIS
-/* XXX is math.h really needed? - atmos */
-#include <math.h>
-#include <vorbis/codec.h>
-
-// This struct is also defined in demux_ogg.c => common header ?
-typedef struct ov_struct_st {
-  vorbis_info      vi; /* struct that stores all the static vorbis bitstream
-			  settings */
-  vorbis_comment   vc; /* struct that stores all the bitstream user comments */
-  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
-  vorbis_block     vb; /* local working space for packet->PCM decode */
-} ov_struct_t;
-#endif
-
-#ifdef HAVE_FAAD
-#include <faad.h>
-static faacDecHandle faac_hdec;
-static faacDecFrameInfo faac_finfo;
-static int faac_bytesconsumed = 0;
-static unsigned char *faac_buffer;
-/* configure maximum supported channels, *
- * this is theoretically max. 64 chans   */
-#define FAAD_MAX_CHANNELS 6
-#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)		       
-#endif
-
-#ifdef USE_LIBAVCODEC
-#ifdef USE_LIBAVCODEC_SO
-#include <libffmpeg/avcodec.h>
-#else
-#include "libavcodec/avcodec.h"
-#endif
-    static AVCodec *lavc_codec=NULL;
-    static AVCodecContext lavc_context;
-    extern int avcodec_inited;
-#endif
-
-
-
-#ifdef USE_LIBMAD
-#include <mad.h>
-
-#define MAD_SINGLE_BUFFER_SIZE 8192
-#define MAD_TOTAL_BUFFER_SIZE  ((MAD_SINGLE_BUFFER_SIZE)*3)
-
-static struct mad_stream mad_stream;
-static struct mad_frame  mad_frame;
-static struct mad_synth  mad_synth;
-static char*  mad_in_buffer = 0; /* base pointer of buffer */
-
-// ensure buffer is filled with some data
-static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length)
-{
-  if(sh_audio->a_in_buffer_len < length) {
-    int len = demux_read_data(sh_audio->ds, sh_audio->a_in_buffer+sh_audio->a_in_buffer_len, length-sh_audio->a_in_buffer_len);
-    sh_audio->a_in_buffer_len += len;
-//    printf("mad_prepare_buffer: read %d bytes\n", len);
-  }
-}
-
-static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms)
-{
-  /* rotate buffer while possible, in order to reduce the overhead of endless memcpy */
-  int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer;
-  if((unsigned long)(sh_audio->a_in_buffer) - (unsigned long)mad_in_buffer < 
-     (MAD_TOTAL_BUFFER_SIZE - MAD_SINGLE_BUFFER_SIZE - delta)) {
-    sh_audio->a_in_buffer += delta;
-    sh_audio->a_in_buffer_len -= delta;
-  } else {
-    sh_audio->a_in_buffer = mad_in_buffer;
-    sh_audio->a_in_buffer_len -= delta;
-    memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len);
-  }
-}
-
-static inline
-signed short mad_scale(mad_fixed_t sample)
-{
-  /* round */
-  sample += (1L << (MAD_F_FRACBITS - 16));
-
-  /* clip */
-  if (sample >= MAD_F_ONE)
-    sample = MAD_F_ONE - 1;
-  else if (sample < -MAD_F_ONE)
-    sample = -MAD_F_ONE;
-
-  /* quantize */
-  return sample >> (MAD_F_FRACBITS + 1 - 16);
-
-}
-
-static void mad_sync(sh_audio_t* sh_audio, struct mad_stream* ms)
-{
-    int len;
-#if 1
-    int skipped = 0;
-
-//    printf("buffer len: %d\n", sh_audio->a_in_buffer_len);    
-    while(sh_audio->a_in_buffer_len - skipped)
-    {
-	len = mp_decode_mp3_header(sh_audio->a_in_buffer+skipped);
-	if (len != -1)
-	{
-//	    printf("Frame len=%d\n", len);
-	    break;
-	}
-	else
-	    skipped++;
-    }
-    if (skipped)
-    {
-	mp_msg(MSGT_DECAUDIO, MSGL_INFO, "mad: audio synced, skipped bytes: %d\n", skipped);
-//	ms->skiplen += skipped;
-//	printf("skiplen: %d (skipped: %d)\n", ms->skiplen, skipped);
-
-//	if (sh_audio->a_in_buffer_len - skipped < MAD_BUFFER_GUARD)
-//	    printf("Mad reports: too small buffer\n");
-
-//	mad_stream_buffer(ms, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len-skipped);
-//	mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_len-skipped);
-
-	/* move frame to the beginning of the buffer and fill up to a_in_buffer_size */
-	sh_audio->a_in_buffer_len -= skipped;
-	memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer+skipped, sh_audio->a_in_buffer_len);
-	mad_prepare_buffer(sh_audio, ms, sh_audio->a_in_buffer_size);
-	mad_stream_buffer(ms, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len);
-//	printf("bufflen: %d\n", sh_audio->a_in_buffer_len);
-	
-//	len = mp_decode_mp3_header(sh_audio->a_in_buffer);
-//	printf("len: %d\n", len);
-	ms->md_len = len;
-    }
-#else
-    len = mad_stream_sync(&ms);
-    if (len == -1)
-    {
-	mp_msg(MSGT_DECVIDEO, MSGL_ERR, "Mad sync failed\n");
-    }
-#endif
-}
-
-static void mad_print_error(struct mad_stream *mad_stream)
-{
-    printf("error (0x%x): ", mad_stream->error);
-    switch(mad_stream->error)
-    {
-	case MAD_ERROR_BUFLEN:	printf("buffer too small");		break;
-	case MAD_ERROR_BUFPTR:	printf("invalid buffer pointer"); 	break;
-	case MAD_ERROR_NOMEM:	printf("not enought memory");		break;
-	case MAD_ERROR_LOSTSYNC:	printf("lost sync");		break;
-	case MAD_ERROR_BADLAYER:	printf("bad layer");		break;
-	case MAD_ERROR_BADBITRATE:	printf("bad bitrate");		break;
-	case MAD_ERROR_BADSAMPLERATE:	printf("bad samplerate");	break;
-	case MAD_ERROR_BADEMPHASIS:	printf("bad emphasis");		break;
-	case MAD_ERROR_BADCRC:		printf("bad crc");		break;
-	case MAD_ERROR_BADBITALLOC:	printf("forbidden bit alloc val"); break;
-	case MAD_ERROR_BADSCALEFACTOR:	printf("bad scalefactor index"); break;
-	case MAD_ERROR_BADFRAMELEN:	printf("bad frame length");	break;
-	case MAD_ERROR_BADBIGVALUES:	printf("bad bigvalues count");	break;
-	case MAD_ERROR_BADBLOCKTYPE:	printf("reserved blocktype");	break;
-	case MAD_ERROR_BADSCFSI:	printf("bad scalefactor selinfo"); break;
-	case MAD_ERROR_BADDATAPTR:	printf("bad maindatabegin ptr"); break;
-	case MAD_ERROR_BADPART3LEN:	printf("bad audio data len");	break;
-	case MAD_ERROR_BADHUFFTABLE:	printf("bad huffman table sel"); break;
-	case MAD_ERROR_BADHUFFDATA:	printf("huffman data overrun");	break;
-	case MAD_ERROR_BADSTEREO:	printf("incomp. blocktype for JS"); break;
-	default:
-	    printf("unknown error");
-    }
-    printf("\n");
-}
-#endif
-
-
-static 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;
-}
-
-int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen);
-
-
-static sh_audio_t* dec_audio_sh=NULL;
-
-#ifdef USE_LIBAC3
-// AC3 decoder buffer callback:
-static void ac3_fill_buffer(uint8_t **start,uint8_t **end){
-    int len=ds_get_packet(dec_audio_sh->ds,start);
-    //printf("<ac3:%d>\n",len);
-    if(len<0)
-          *start = *end = NULL;
-    else
-          *end = *start + len;
-}
-#endif
-
-// MP3 decoder buffer callback:
-int mplayer_audio_read(char *buf,int size){
-  int len;
-  len=demux_read_data(dec_audio_sh->ds,buf,size);
-  return len;
-}
-
-int init_audio(sh_audio_t *sh_audio){
-int driver=sh_audio->codec->driver;
-
-if(!sh_audio->samplesize)
-  sh_audio->samplesize=2;
-if(!sh_audio->sample_format)
-#ifdef WORDS_BIGENDIAN
-  sh_audio->sample_format=AFMT_S16_BE;
-#else
-  sh_audio->sample_format=AFMT_S16_LE;
-#endif
-//sh_audio->samplerate=0;
-//sh_audio->pcm_bswap=0;
-//sh_audio->o_bps=0;
-
-sh_audio->a_buffer_size=0;
-sh_audio->a_buffer=NULL;
-
-sh_audio->a_in_buffer_len=0;
-
-// setup required min. in/out buffer size:
-sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM
-
-switch(driver){
-case AFM_ACM:
-#ifndef	USE_WIN32DLL
-  mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport);
-  driver=0;
-#else
-  // 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;
-//    if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384;
-//    sh_audio->a_buffer_size=sh_audio->audio_out_minsize;
-//    if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST)
-//        sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST;
-  } else {
-    mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror);
-    driver=0;
-  }
-#endif
-  break;
-case AFM_DSHOW:
-#ifndef USE_DIRECTSHOW
-  mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio);
-  driver=0;
-#else
-  // Win32 DShow audio codec:
-//  printf("DShow_audio: channs=%d  rate=%d\n",sh_audio->channels,sh_audio->samplerate);
-  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);
-    driver=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;
-  }
-#endif
-  break;
-case AFM_VORBIS:
-#ifndef	HAVE_OGGVORBIS
-  mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis);
-  driver=0;
-#else
-  /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
-  // Is there always 1024 samples/frame ? ***** Albeu
-  sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
-#endif
-  break;
-case AFM_AAC:
-  // AAC (MPEG2 Audio, MPEG4 Audio)
-#ifndef HAVE_FAAD
-  mp_msg(MSGT_DECAUDIO,MSGL_ERR,"Error: Cannot decode AAC data, because MPlayer was compiled without FAAD support\n"/*MSGTR_NoFAAD*/);
-  driver=0;
-#else  
-  mp_msg(MSGT_DECAUDIO,MSGL_V,"Using FAAD to decode AAC content!\n"/*MSGTR_UseFAAD*/);
-  // Samples per frame * channels per frame, this might not work with >2 chan AAC, need test samples! ::atmos
-  sh_audio->audio_out_minsize=2048*2;
-#endif  
-  break;
-case AFM_PCM:
-case AFM_DVDPCM:
-case AFM_ALAW:
-  // PCM, aLaw
-  sh_audio->audio_out_minsize=2048;
-  break;
-case AFM_AC3:
-case AFM_A52:
-  // Dolby AC3 audio:
-  // however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame
-  sh_audio->audio_out_minsize=audio_output_channels*2*256*6;
-  break;
-case AFM_HWAC3:
-  // Dolby AC3 audio:
-  sh_audio->audio_out_minsize=4*256*6;
-//  sh_audio->sample_format = AFMT_AC3;
-//  sh_audio->sample_format = AFMT_S16_LE;
-  sh_audio->channels=2;
-  break;
-case AFM_GSM:
-  // MS-GSM audio codec:
-  sh_audio->audio_out_minsize=4*320;
-  break;
-case AFM_IMAADPCM:
-  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;
-  break;
-case AFM_MSADPCM:
-  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;
-  break;
-case AFM_DK4ADPCM:
-  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;
-  break;
-case AFM_DK3ADPCM:
-  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;
-  break;
-case AFM_ROQAUDIO:
-  // minsize was stored in wf->nBlockAlign by the RoQ demuxer
-  sh_audio->audio_out_minsize=sh_audio->wf->nBlockAlign;
-  sh_audio->ds->ss_div=DK3_ADPCM_SAMPLES_PER_BLOCK;
-  sh_audio->ds->ss_mul=DK3_ADPCM_BLOCK_SIZE;
-  sh_audio->context = roq_decode_audio_init();
-  break;
-case AFM_MPEG:
-  // MPEG Audio:
-  sh_audio->audio_out_minsize=4608;
-  break;
-#ifdef USE_G72X
-case AFM_G72X:
-//  g72x_reader_init(&g72x_data,G723_16_BITS_PER_SAMPLE);
-  g72x_reader_init(&g72x_data,G723_24_BITS_PER_SAMPLE);
-//  g72x_reader_init(&g72x_data,G721_32_BITS_PER_SAMPLE);
-//  g72x_reader_init(&g72x_data,G721_40_BITS_PER_SAMPLE);
-  sh_audio->audio_out_minsize=g72x_data.samplesperblock*4;
-  break;
-#endif
-case AFM_FFMPEG:
-#ifndef USE_LIBAVCODEC
-   mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport);
-   return 0;
-#else
-  // FFmpeg Audio:
-  sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
-  break;
-#endif
-
-#ifdef USE_LIBMAD
- case AFM_MAD:
-   mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: setting minimum outputsize\n");
-   sh_audio->audio_out_minsize=4608;
-   if(sh_audio->audio_in_minsize<MAD_SINGLE_BUFFER_SIZE) sh_audio->audio_in_minsize=MAD_SINGLE_BUFFER_SIZE;
-   sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize;
-   mad_in_buffer = sh_audio->a_in_buffer = malloc(MAD_TOTAL_BUFFER_SIZE);
-   sh_audio->a_in_buffer_len=0;
-   break;
-#endif
-}
-
-if(!driver) return 0;
-
-// allocate audio out buffer:
-sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc.
-
-mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n",
-    sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size);
-
-sh_audio->a_buffer=malloc(sh_audio->a_buffer_size);
-if(!sh_audio->a_buffer){
-    mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf);
-    return 0;
-}
-memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size);
-sh_audio->a_buffer_len=0;
-
-switch(driver){
-#ifdef USE_WIN32DLL
-case AFM_ACM: {
-    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);
-        driver=0;
-    }
-    sh_audio->a_buffer_len=ret;
-    break;
-}
-#endif
-case AFM_PCM: {
-    // AVI PCM 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;
-    }
-    break;
-}
-case AFM_DVDPCM: {
-    // DVD PCM Audio:
-    sh_audio->channels=2;
-    sh_audio->samplerate=48000;
-    sh_audio->i_bps=2*2*48000;
-//    sh_audio->pcm_bswap=1;
-    break;
-}
-case AFM_AC3: {
-#ifndef USE_LIBAC3
-  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"WARNING: libac3 support is disabled. (hint: upgrade codecs.conf)\n");
-  driver=0;
-#else
-  // Dolby AC3 audio:
-  dec_audio_sh=sh_audio; // save sh_audio for the callback:
-  ac3_config.fill_buffer_callback = ac3_fill_buffer;
-  ac3_config.num_output_ch = audio_output_channels;
-  ac3_config.flags = 0;
-if(gCpuCaps.hasMMX){
-  ac3_config.flags |= AC3_MMX_ENABLE;
-}
-if(gCpuCaps.has3DNow){
-  ac3_config.flags |= AC3_3DNOW_ENABLE;
-}
-  ac3_init();
-  sh_audio->ac3_frame = ac3_decode_frame();
-  if(sh_audio->ac3_frame){
-    ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame;
-    sh_audio->samplerate=fr->sampling_rate;
-    sh_audio->channels=ac3_config.num_output_ch;
-    // 1 frame: 6*256 samples     1 sec: sh_audio->samplerate samples
-    //sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256);
-    sh_audio->i_bps=fr->bit_rate*(1000/8);
-  } else {
-    driver=0; // bad frame -> disable audio
-  }
-#endif
-  break;
-}
-case AFM_A52: {
-  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");
-	driver=0;break;
-  }
-   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");
-	driver=0;break;
-  }
-  // 'a52 cannot upmix' hotfix:
-  a52_printinfo(sh_audio);
-//  if(audio_output_channels<sh_audio->channels)
-//      sh_audio->channels=audio_output_channels;
-  // channels setup:
-  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");
-    driver=0;break;
-  }
-  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");
-    driver=0;break;
-  }
-  break;
-}
-case AFM_HWAC3: {
-  // Dolby AC3 passthrough:
-  a52_samples=a52_init (a52_accel);
-  if (a52_samples == NULL) {
-       mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
-       driver=0;break;
-  }
-  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");
-       driver=0;break;
-  }
-  
-  //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!)
-
-  break;
-}
-case AFM_ALAW: {
-  // aLaw audio codec:
-  sh_audio->channels=sh_audio->wf->nChannels;
-  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-  sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate;
-  break;
-}
-#ifdef USE_G72X
-case AFM_G72X: {
-  // GSM 723 audio codec:
-  sh_audio->channels=sh_audio->wf->nChannels;
-  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-  sh_audio->i_bps=(sh_audio->samplerate/g72x_data.samplesperblock)*g72x_data.blocksize;
-  break;
-}
-#endif
-#ifdef USE_LIBAVCODEC
-case AFM_FFMPEG: {
-   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
-  break;
-}
-#endif
-case AFM_GSM: {
-  // MS-GSM audio codec:
-  GSM_Init();
-  sh_audio->channels=sh_audio->wf->nChannels;
-  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-  // decodes 65 byte -> 320 short
-  // 1 sec: sh_audio->channels*sh_audio->samplerate  samples
-  // 1 frame: 320 samples
-  sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320;  // 1:10
-  break;
-}
-case AFM_IMAADPCM:
-  // 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
-  break;
-case AFM_MSADPCM:
-  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;
-  break;
-case AFM_DK4ADPCM:
-  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;
-  break;
-case AFM_DK3ADPCM:
-  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;
-  break;
-case AFM_ROQAUDIO:
-  sh_audio->channels=sh_audio->wf->nChannels;
-  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
-  sh_audio->i_bps = (sh_audio->channels * 22050) / 2;
-  break;
-case AFM_MPEG: {
-  // MPEG Audio:
-  dec_audio_sh=sh_audio; // save sh_audio for the callback:
-#ifdef USE_FAKE_MONO
-  MP3_Init(fakemono);
-#else
-  MP3_Init();
-#endif
-  MP3_samplerate=MP3_channels=0;
-  sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1);
-  sh_audio->channels=2; // hack
-  sh_audio->samplerate=MP3_samplerate;
-  sh_audio->i_bps=MP3_bitrate*(1000/8);
-  MP3_PrintHeader();
-  break;
-}
-#ifdef HAVE_OGGVORBIS
-case AFM_VORBIS: {
-  ogg_packet op;
-  vorbis_comment vc;
-  struct ov_struct_st *ov;
-
-  /// Init the decoder with the 3 header packets
-  ov = (struct ov_struct_st*)malloc(sizeof(struct ov_struct_st));
-  vorbis_info_init(&ov->vi);
-  vorbis_comment_init(&vc);
-  op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
-  op.b_o_s  = 1;
-  /// Header
-  if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
-    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: initial (identification) header broken!\n");
-    driver = 0;
-    free(ov);
-    break;
-  }
-  op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
-  op.b_o_s  = 0;
-  /// Comments
-  if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
-    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: comment header broken!\n");
-    driver = 0;
-    free(ov);
-    break;
-  }
-  op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
-  //// Codebook
-  if(vorbis_synthesis_headerin(&ov->vi,&vc,&op)<0) {
-    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n");
-    driver = 0;
-    free(ov);
-    break;
-  } else { /// Print the infos
-    char **ptr=vc.user_comments;
-    while(*ptr){
-      mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
-      ++ptr;
-    }
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",ov->vi.channels,ov->vi.rate,ov->vi.bitrate_nominal/1000, (ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C');
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor);
-  }
-
-  // Setup the decoder
-  sh_audio->channels=ov->vi.channels; 
-  sh_audio->samplerate=ov->vi.rate;
-  sh_audio->i_bps=ov->vi.bitrate_nominal/8;
-  sh_audio->context = ov;
-
-  /// Finish the decoder init
-  vorbis_synthesis_init(&ov->vd,&ov->vi);
-  vorbis_block_init(&ov->vd,&ov->vb);
-  mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
-} break;
-#endif
-
-#ifdef HAVE_FAAD
-case AFM_AAC: {
-  unsigned long faac_samplerate, faac_channels;
-  faacDecConfigurationPtr faac_conf;
-  faac_hdec = faacDecOpen();
-
-  if(faac_buffer == NULL)
-    faac_buffer = (unsigned char*)calloc(1,FAAD_BUFFLEN);
-  demux_read_data(sh_audio->ds, faac_buffer, FAAD_BUFFLEN);
-
-  // If we don't get the ES descriptor, try manual config
-  if(!sh_audio->codecdata_len) {
-#if 1
-    /* Set the default object type and samplerate */
-    /* This is useful for RAW AAC files */
-    faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
-    if(sh_audio->samplerate)
-      faac_conf->defSampleRate = sh_audio->samplerate;
-    /* XXX: FAAD support FLOAT output, how do we handle
-      * that (FAAD_FMT_FLOAT)? ::atmos
-      */
-    if(sh_audio->samplesize)
-      switch(sh_audio->samplesize){
-	case 1: // 8Bit
-	  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
-	default:
-	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);
-#endif
-
-    /* init the codec */
-    faac_bytesconsumed = faacDecInit(faac_hdec, faac_buffer,
-       &faac_samplerate, &faac_channels);
-
-  } else { // We have ES DS in codecdata
-    /*int i;
-    for(i = 0; i < sh_audio->codecdata_len; i++)
-      printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/
-
-    faac_bytesconsumed = faacDecInit2(faac_hdec, sh_audio->codecdata,
-       sh_audio->codecdata_len,	&faac_samplerate, &faac_channels);
-  }
-  if(faac_bytesconsumed < 0) {
-    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
-    faacDecClose(faac_hdec);
-    free(faac_buffer);
-    faac_buffer = NULL;
-    driver = 0;
-  } else {
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", faac_bytesconsumed); // XXX: remove or move to debug!
-    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz  channels: %d\n", faac_samplerate, faac_channels);
-    sh_audio->channels = faac_channels;
-    sh_audio->samplerate = faac_samplerate;
-    //sh_audio->o_bps = sh_audio->samplesize*faac_channels*faac_samplerate;
-    if(!sh_audio->i_bps) {
-      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
-      sh_audio->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_audio->i_bps*8/1000);
-  }  
-	    
-} break;		
-#endif
-
-#ifdef USE_LIBMAD
- case AFM_MAD:
-   {
-     printf("%s %s %s (%s)\n", mad_version, mad_copyright, mad_author, mad_build);
-
-     mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: initialising\n");
-     mad_frame_init(&mad_frame);
-     mad_stream_init(&mad_stream);
-
-     mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: preparing buffer\n");
-     mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size);
-     mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len);
-//     mad_stream_sync(&mad_stream);
-     mad_sync(sh_audio, &mad_stream);
-     mad_synth_init(&mad_synth);
-
-     if(mad_frame_decode(&mad_frame, &mad_stream) == 0)
-       {
-	 mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: post processing buffer\n");
-	 mad_postprocess_buffer(sh_audio, &mad_stream);
-       }
-     else
-       {
-	 mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: frame decoding failed\n");
-	 mad_print_error(&mad_stream);
-       }
-     
-     switch (mad_frame.header.mode)
-     {
-        case MAD_MODE_SINGLE_CHANNEL:
-	    sh_audio->channels=1;
-	    break;
-	case MAD_MODE_DUAL_CHANNEL:
-	case MAD_MODE_JOINT_STEREO:
-	case MAD_MODE_STEREO:
-	    sh_audio->channels=2;
-	    break;
-	default:
-	    mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n");
-     }
-     mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n",
-        sh_audio->channels, mad_frame.header.mode);
-/* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */
-#if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13)
-     sh_audio->samplerate=mad_frame.header.samplerate;
-#else
-     sh_audio->samplerate=mad_frame.header.sfreq;
-#endif
-     sh_audio->i_bps=mad_frame.header.bitrate;
-     mp_msg(MSGT_DECVIDEO, MSGL_V, "mad: continuing\n");
-     break;
-   }
-#endif
-}
-
-if(!sh_audio->channels || !sh_audio->samplerate){
-  mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio);
-  driver=0;
-}
-
-  if(!driver){
-      if(sh_audio->a_buffer) free(sh_audio->a_buffer);
-      sh_audio->a_buffer=NULL;
-      return 0;
-  }
-
-  if(!sh_audio->o_bps)
-  sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize;
-  return driver;
-}
-
-// Audio decoding:
-
-// Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc)
-// buffer length is 'maxlen' bytes, it shouldn't be exceeded...
-
-int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
-    int len=-1;
-    switch(sh_audio->codec->driver){
-#ifdef USE_LIBAVCODEC
-      case AFM_FFMPEG: {
-          unsigned char *start=NULL;