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/* 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 */
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