view libmpcodecs/ad_faad.c @ 31246:cc6ee3017097

Limit buffered PTS only when we actually got a frame from the decoder. This avoids some issues with H.264 PAFF due to dropping PTS values too early because only every second packet actually produced output. Just keeping up to one additional pts value would have avoided this particular issue as well, but this is more generic.
author reimar
date Thu, 03 Jun 2010 20:59:40 +0000
parents ac4bcd2064ce
children 86888a4c406e
line wrap: on
line source

/*
 * MPlayer AAC decoder using libfaad2
 *
 * Copyright (C) 2002 Felix Buenemann <atmosfear at users.sourceforge.net>
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

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

#include "config.h"
#include "ad_internal.h"
#include "libaf/reorder_ch.h"

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

LIBAD_EXTERN(faad)

#ifndef CONFIG_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 8
#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;
}

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 = malloc(sh->codecdata_len);
    memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len);
    mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n");
  }
  if(!sh->codecdata_len) {
    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);

    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);
    // 8 channels is aac channel order #7.
    sh->channels = faac_channels == 7 ? 8 : 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 len = 0, last_dec_len = 1, errors = 0;
  //  int j = 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);

      if (sh->channels >= 5)
        reorder_channel_copy_nch(faac_sample_buffer,
                                 AF_CHANNEL_LAYOUT_AAC_DEFAULT,
                                 buf+len, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
                                 sh->channels,
                                 faac_finfo.samples, sh->samplesize);
      else
      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;
}