path: root/dec_audio.c

#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 "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/DirectShow/DS_AudioDec.h"
static void* ds_adec=NULL;
#endif

#ifdef HAVE_OGGVORBIS
/* XXX is math.h really needed? - atmos */
#include <math.h>
#include <vorbis/codec.h>

typedef struct ov_struct_st {
  ogg_sync_state   oy; /* sync and verify incoming physical bitstream */
  ogg_stream_state os; /* take physical pages, weld into a logical
			  stream of packets */
  ogg_page         og; /* one Ogg bitstream page.  Vorbis packets are inside */
  ogg_packet       op; /* one raw packet of data for decode */

  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 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>
static struct mad_stream mad_stream;
static struct mad_frame  mad_frame;
static struct mad_synth  mad_synth;

// 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)
{
  int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer;
  if(delta != 0) {
    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)
    {
	printf("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)
    {
	printf("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;

sh_audio->samplesize=2;
#if 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 */
  sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#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->channels=1;
  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;
  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_FOX62ADPCM:
  sh_audio->audio_out_minsize=FOX62_ADPCM_SAMPLES_PER_BLOCK * 4;
  sh_audio->ds->ss_div=FOX62_ADPCM_SAMPLES_PER_BLOCK;
  sh_audio->ds->ss_mul=FOX62_ADPCM_BLOCK_SIZE;
  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:
   printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__);
   sh_audio->audio_out_minsize=4608;
   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;
   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_OUTBUR