view libmpcodecs/ad_faad.c @ 20874:b6d87b58754f

Partial fix for semitransparent glyph outlines. This fix removes semitransparent area (less then pixel width) between glyph and it's outline. Instead, it makes them overlap a little. It usually looks much better this way. Complete fix seems impossible with the current output format (single color alpha bitmaps). The right way is to blend both glyph and outline into one bitmap so that 2 pixels with 50% transparency produce a fully solid one. This requires RGBA bitmap output from libass.

/* ad_faad.c - MPlayer AAC decoder using libfaad2
 * This file is part of MPlayer, see http://mplayerhq.hu/ for info.  
 * (c)2002 by Felix Buenemann <atmosfear at users.sourceforge.net>
 * File licensed under the GPL, see http://www.fsf.org/ for more info.
 */

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>

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

static ad_info_t info = 
{
	"AAC (MPEG2/4 Advanced Audio Coding)",
	"faad",
	"Felix Buenemann",
	"faad2",
	"uses libfaad2"
};

LIBAD_EXTERN(faad)

#ifndef USE_FAAD_INTERNAL
#include <faad.h>
#else
#include "libfaad2/faad.h"
#endif

/* configure maximum supported channels, *
 * this is theoretically max. 64 chans   */
#define FAAD_MAX_CHANNELS 6
#define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS)		       

//#define AAC_DUMP_COMPRESSED  

static faacDecHandle faac_hdec;
static faacDecFrameInfo faac_finfo;

static int preinit(sh_audio_t *sh)
{
  sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS;
  sh->audio_in_minsize=FAAD_BUFFLEN;
  return 1;
}

static int aac_probe(unsigned char *buffer, int len)
{
  int i = 0, pos = 0;
  mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len);
  while(i <= len-4) {
    if(
       ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) ||
       (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F')
    ) {
      pos = i;
      break;
    }
    mp_msg(MSGT_DECAUDIO,MSGL_V, "AUDIO PAYLOAD: %x %x %x %x\n", buffer[i], buffer[i+1], buffer[i+2], buffer[i+3]);
    i++;
  }
  mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos);
  return pos;
}
	
extern int audio_output_channels;
static int init(sh_audio_t *sh)
{
  unsigned long faac_samplerate;
  unsigned char faac_channels;
  int faac_init, pos = 0;
  faac_hdec = faacDecOpen();

  // If we don't get the ES descriptor, try manual config
  if(!sh->codecdata_len && sh->wf) {
    sh->codecdata_len = sh->wf->cbSize;
    sh->codecdata = (char*)(sh->wf+1);
    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
  }
  if(!sh->codecdata_len) {
#if 1
    faacDecConfigurationPtr faac_conf;
    /* Set the default object type and samplerate */
    /* This is useful for RAW AAC files */
    faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
    if(sh->samplerate)
      faac_conf->defSampleRate = sh->samplerate;
    /* XXX: FAAD support FLOAT output, how do we handle
      * that (FAAD_FMT_FLOAT)? ::atmos
      */
    if (audio_output_channels <= 2) faac_conf->downMatrix = 1;
      switch(sh->samplesize){
	case 1: // 8Bit
	  mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n");
	default:
	  sh->samplesize=2;
	case 2: // 16Bit
	  faac_conf->outputFormat = FAAD_FMT_16BIT;
	  break;
	case 3: // 24Bit
	  faac_conf->outputFormat = FAAD_FMT_24BIT;
	  break;
	case 4: // 32Bit
	  faac_conf->outputFormat = FAAD_FMT_32BIT;
	  break;
      }
    //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available.

    faacDecSetConfiguration(faac_hdec, faac_conf);
#endif

    sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);
    pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
    if(pos) {
      sh->a_in_buffer_len -= pos;
      memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
      sh->a_in_buffer_len +=
	demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]),
	sh->a_in_buffer_size - sh->a_in_buffer_len);
      pos = 0;
    }

    /* init the codec */
    faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
       sh->a_in_buffer_len, &faac_samplerate, &faac_channels);

    sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed
    // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi

  } else { // We have ES DS in codecdata
    faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec);
    if (audio_output_channels <= 2) {
        faac_conf->downMatrix = 1;
        faacDecSetConfiguration(faac_hdec, faac_conf);
    }
    
    /*int i;
    for(i = 0; i < sh_audio->codecdata_len; i++)
      printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/

    faac_init = faacDecInit2(faac_hdec, sh->codecdata,
       sh->codecdata_len,	&faac_samplerate, &faac_channels);
  }
  if(faac_init < 0) {
    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup!
    faacDecClose(faac_hdec);
    // XXX: free a_in_buffer here or in uninit?
    return 0;
  } else {
    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug!
    mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz  channels: %d\n", faac_samplerate, faac_channels);
    sh->channels = faac_channels;
    if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1;
    sh->samplerate = faac_samplerate;
    sh->samplesize=2;
    //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate;
    if(!sh->i_bps) {
      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n");
      sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos
    } else 
      mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000);
  }  
  return 1;
}

static void uninit(sh_audio_t *sh)
{
  mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n");
  faacDecClose(faac_hdec);
}

static int aac_sync(sh_audio_t *sh)
{
  int pos = 0;
  if(!sh->codecdata_len) {
    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
      sh->a_in_buffer_len +=
	demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
	sh->a_in_buffer_size - sh->a_in_buffer_len);
    }
    pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len);
    if(pos) {
      sh->a_in_buffer_len -= pos;
      memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len);
      mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos);
    }
  }
  return pos;
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
      case ADCTRL_RESYNC_STREAM:
         aac_sync(sh);
	 return CONTROL_TRUE;
#if 0      
      case ADCTRL_SKIP_FRAME:
	  return CONTROL_TRUE;
#endif
    }
  return CONTROL_UNKNOWN;
}

#define MAX_FAAD_ERRORS 10
static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
  int j = 0, len = 0, last_dec_len = 1, errors = 0;	      
  void *faac_sample_buffer;

  while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) {

    /* update buffer for raw aac streams: */
  if(!sh->codecdata_len)
    if(sh->a_in_buffer_len < sh->a_in_buffer_size){
      sh->a_in_buffer_len +=
	demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len],
	sh->a_in_buffer_size - sh->a_in_buffer_len);
    }
	  
#ifdef DUMP_AAC_COMPRESSED
    {int i;
    for (i = 0; i < 16; i++)
      printf ("%02X ", sh->a_in_buffer[i]);
    printf ("\n");}
#endif

  if(!sh->codecdata_len){
   // raw aac stream:
   do {
    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len);
	
    /* update buffer index after faacDecDecode */
    if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) {
      sh->a_in_buffer_len=0;
    } else {
      sh->a_in_buffer_len-=faac_finfo.bytesconsumed;
      memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len);
    }

    if(faac_finfo.error > 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n",
              faacDecGetErrorMessage(faac_finfo.error));
      if (sh->a_in_buffer_len <= 0) {
        errors = MAX_FAAD_ERRORS;
        break;
      }
      sh->a_in_buffer_len--;
      memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len);
      aac_sync(sh);
      errors++;
    } else
      break;
   } while(errors < MAX_FAAD_ERRORS);	  
  } else {
   // packetized (.mp4) aac stream:
    unsigned char* bufptr=NULL;
    double pts;
    int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts);
    if(buflen<=0) break;
    if (pts != MP_NOPTS_VALUE) {
	sh->pts = pts;
	sh->pts_bytes = 0;
    }
    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen);
  }
  //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]);
  
    if(faac_finfo.error > 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n",
      faacDecGetErrorMessage(faac_finfo.error));
    } else if (faac_finfo.samples == 0) {
      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n");
    } else {
      /* XXX: samples already multiplied by channels! */
      mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n",
      sh->samplesize*faac_finfo.samples);
      memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
      last_dec_len = sh->samplesize*faac_finfo.samples;
      len += last_dec_len;
      sh->pts_bytes += last_dec_len;
    //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
    }
  }
  return len;
}