annotate libmpcodecs/ad_liba52.c @ 21548:bf65ffcf0cdb

Set AVFMT_FLAG_GENPTS if -correct-pts is used. This should allow using -correct-pts (and thus filters which adjust pts or add frames) with dvd or other mpeg container files by specifying "-correct-pts -demuxer lavf -vc ffmpeg12". Might work with libmpeg2 decoder too but certainly not with internal demuxer. Using this flag isn't quite optimal as it can cause extra buffering of demuxed frames, but at least it's better than just failing until a more complex solution is implemented.

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

#include "config.h"

#include "mp_msg.h"
#include "help_mp.h"

#include "ad_internal.h"

#include "cpudetect.h"

#include "libaf/af_format.h"

#include "liba52/a52.h"
#include "liba52/mm_accel.h"

static a52_state_t *a52_state;
static uint32_t a52_flags=0;
/** Used by a52_resample_float, it defines the mapping between liba52
 * channels and output channels.  The ith nibble from the right in the
 * hex representation of channel_map is the index of the source
 * channel corresponding to the ith output channel.  Source channels are
 * indexed 1-6.  Silent output channels are marked by 0xf. */
static uint32_t channel_map;

#define DRC_NO_ACTION      0
#define DRC_NO_COMPRESSION 1
#define DRC_CALLBACK       2

/** The output is multiplied by this var.  Used for volume control */
static sample_t a52_level = 1;
/** The value of the -a52drc switch. */
float a52_drc_level = 1.0;
static int a52_drc_action = DRC_NO_ACTION;

#include "mpbswap.h"

static ad_info_t info = 
{
	"AC3 decoding with liba52",
	"liba52",
	"Nick Kurshev",
	"Michel LESPINASSE",
	""
};

LIBAD_EXTERN(liba52)

extern int audio_output_channels;

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<8){
	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;
    }
    if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
    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*/
    if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
    memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
    --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;
    sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
    demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
    if(sh_audio->format!=0x2000)
	swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
    
    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_V,"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;
}

sample_t dynrng_call (sample_t c, void *data) {
//	fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
	return pow((double)c, a52_drc_level);
}


static int preinit(sh_audio_t *sh)
{
  /* Dolby AC3 audio: */
  /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
  if (sh->samplesize < 2) sh->samplesize = 2;
  sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
  sh->audio_in_minsize=3840;
  a52_level = 1.0;
  return 1;
}

/**
 * \brief Function to convert the "planar" float format used by liba52
 * into the interleaved float format used by libaf/libao2.
 * \param in the input buffer containing the planar samples.
 * \param out the output buffer where the interleaved result is stored.
 */
static int a52_resample_float(float *in, int16_t *out)
{
    unsigned long i;
    float *p = (float*) out;
    for (i = 0; i != 256; i++) {
	unsigned long map = channel_map;
	do {
	    unsigned long ch = map & 15;
	    if (ch == 15)
		*p = 0;
	    else
		*p = in[i + ((ch-1)<<8)];
	    p++;
	} while ((map >>= 4));
    }
    return (int16_t*) p - out;
}

static int init(sh_audio_t *sh_audio)
{
  uint32_t a52_accel=0;
  sample_t level=a52_level, bias=384;
  int flags=0;
  /* Dolby AC3 audio:*/
  if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
  if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
  if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
  if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
  if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
  if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
  a52_state=a52_init (a52_accel);
  if (a52_state == NULL) {
	mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
	return 0;
  }
  if(a52_fillbuff(sh_audio)<0){
	mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
	return 0;
  }


  /* Init a52 dynrng */
  if (a52_drc_level < 0.001) {
	  /* level == 0 --> no compression, init library without callback */
	  a52_drc_action = DRC_NO_COMPRESSION;
  } else if (a52_drc_level > 0.999) {
	  /* level == 1 --> full compression, do nothing at all (library default = full compression) */
	  a52_drc_action = DRC_NO_ACTION;
  } else {
	  a52_drc_action = DRC_CALLBACK;
  }
  /* Library init for dynrng has to be done for each frame, see decode_audio() */


  /* 'a52 cannot upmix' hotfix:*/
  a52_printinfo(sh_audio);
  sh_audio->channels=audio_output_channels;
while(sh_audio->channels>0){
  switch(sh_audio->channels){
	    case 1: a52_flags=A52_MONO; break;
/*	    case 2: a52_flags=A52_STEREO; break;*/
	    case 2: a52_flags=A52_DOLBY; break;
/*	    case 3: a52_flags=A52_3F; break;*/
	    case 3: a52_flags=A52_2F1R; break;
	    case 4: a52_flags=A52_2F2R; break; /* 2+2*/
	    case 5: a52_flags=A52_3F2R; break;
	    case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
  }
  /* test:*/
  flags=a52_flags|A52_ADJUST_LEVEL;
  mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
  if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
    return 0;
  }
  mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
  /* frame decoded, let's init resampler:*/
  channel_map = 0;
  if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
      if (!(flags & A52_LFE)) {
	  switch ((flags<<3) | sh_audio->channels) {
	    case (A52_MONO    << 3) | 1: channel_map = 0x1; break;
	    case (A52_CHANNEL << 3) | 2:
	    case (A52_STEREO  << 3) | 2:
	    case (A52_DOLBY   << 3) | 2: channel_map =    0x21; break;
	    case (A52_2F1R    << 3) | 3: channel_map =   0x321; break;
	    case (A52_2F2R    << 3) | 4: channel_map =  0x4321; break;
	    case (A52_3F      << 3) | 5: channel_map = 0x2ff31; break;
	    case (A52_3F2R    << 3) | 5: channel_map = 0x25431; break;
	  }
      } else if (sh_audio->channels == 6) {
	  switch (flags & ~A52_LFE) {
	    case A52_MONO   : channel_map = 0x12ffff; break;
	    case A52_CHANNEL:
	    case A52_STEREO :
	    case A52_DOLBY  : channel_map = 0x1fff32; break;
	    case A52_3F     : channel_map = 0x13ff42; break;
	    case A52_2F1R   : channel_map = 0x1f4432; break;
	    case A52_2F2R   : channel_map = 0x1f5432; break;
	    case A52_3F2R   : channel_map = 0x136542; break;
	  }
      }
      if (channel_map) {
	  a52_resample = a52_resample_float;
	  break;
      }
  } else
  if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
  --sh_audio->channels; /* try to decrease no. of channels*/
}
  if(sh_audio->channels<=0){
    mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
    return 0;
  }
  return 1;
}

static void uninit(sh_audio_t *sh)
{
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
    switch(cmd)
    {
      case ADCTRL_RESYNC_STREAM:
      case ADCTRL_SKIP_FRAME:
	  a52_fillbuff(sh);
	  return CONTROL_TRUE;
      case ADCTRL_SET_VOLUME: {
	  float vol = *(float*)arg;
	  if (vol > 60.0) vol = 60.0;
	  a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
	  return CONTROL_TRUE;
      }
      case ADCTRL_QUERY_FORMAT:
	  if (*(int*)arg == AF_FORMAT_S16_NE ||
	      *(int*)arg == AF_FORMAT_FLOAT_NE)
	      return CONTROL_TRUE;
	  return CONTROL_FALSE;
    }
  return CONTROL_UNKNOWN;
}

static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
    sample_t level=a52_level, bias=384;
    int flags=a52_flags|A52_ADJUST_LEVEL;
    int i,len=-1;
	if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
	    bias = 0;
	if(!sh_audio->a_in_buffer_len) 
	    if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
	sh_audio->a_in_buffer_len=0;
	if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
	    mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
	    return len;
	}

	/* handle dynrng */
	if (a52_drc_action != DRC_NO_ACTION) {
	    if (a52_drc_action == DRC_NO_COMPRESSION)
		a52_dynrng(a52_state, NULL, NULL);
	    else
		a52_dynrng(a52_state, dynrng_call, NULL);
	}

	len=0;
	for (i = 0; i < 6; i++) {
	    if (a52_block (a52_state)){
		mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
		break;
	    }
	    len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]);
	}
	assert(len <= maxlen);
  return len;
}