view libmpcodecs/ad_faad.c @ 9278:caea8ed36b48

The reason why mplayer crashes (in some cases) when using x11 output and -wid (>0) parameter is this: Mplayer by default creates a colormap using DirectColor visual. If the window given to mplayer uses TrueColor visual there will be an error when mplayer sets the colormap for the window. This patch modifies mplayer to use TrueColor visual if the window given to mplayer uses TrueColor. Another solution is to make sure that the window given to mplayer is created using DirectColor visual if it is supported by the display. Jouni Tulkki <jitulkki@cc.hut.fi>
author arpi
date Tue, 04 Feb 2003 18:31:44 +0000
parents 7ee8239bfcc0
children 6fa743f3094b
line wrap: on
line source

/* 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"

#ifdef HAVE_FAAD

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

LIBAD_EXTERN(faad)

#include <faad.h>

/* 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=2048*FAAD_MAX_CHANNELS;
  sh->audio_in_minsize=FAAD_BUFFLEN;
  return 1;
}

static int init(sh_audio_t *sh)
{
  unsigned long faac_samplerate;
  unsigned char faac_channels;
  int faac_init;
  faac_hdec = faacDecOpen();

  // If we don't get the ES descriptor, try manual config
  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(sh->samplesize)
      switch(sh->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

    sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size);

    /* init the codec */
    faac_init = faacDecInit(faac_hdec, sh->a_in_buffer,
       &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
    /*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: %dHz  channels: %d\n", faac_samplerate, faac_channels);
    sh->channels = faac_channels;
    sh->samplerate = faac_samplerate;
    //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 control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
#if 0      
      case ADCTRL_RESYNC_STREAM:
	  return CONTROL_TRUE;
      case ADCTRL_SKIP_FRAME:
	  return CONTROL_TRUE;
#endif
    }
  return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen)
{
  int j = 0, len = 0;	      
  void *faac_sample_buffer;

  while(len < minlen) {

    /* 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+j);
    /* 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;
      memcpy(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: Trying to resync!\n");
      j++;
    } else
      break;
   } while(j < FAAD_BUFFLEN);	  
  } else {
   // packetized (.mp4) aac stream:
    unsigned char* bufptr=NULL;
    int buflen=ds_get_packet(sh->ds, &bufptr);
    if(buflen<=0) break;
    faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr);
//    printf("FAAC decoded %d of %d  (err: %d)  \n",faac_finfo.bytesconsumed,buflen,faac_finfo.error);
  }
  
    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 (%d Bytes)!\n",
      sh->samplesize*faac_finfo.samples);
      memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples);
      len += sh->samplesize*faac_finfo.samples;
    //printf("FAAD: buffer: %d bytes  consumed: %d \n", k, faac_finfo.bytesconsumed);
    }
  }
  return len;
}

#endif /* !HAVE_FAAD */