view libmpcodecs/ad_msadpcm.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.

/*
    MS ADPCM Decoder for MPlayer
      by Mike Melanson

    This file is responsible for decoding Microsoft ADPCM data.
    Details about the data format can be found here:
      http://www.pcisys.net/~melanson/codecs/
*/

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

#include "config.h"
#include "libavutil/common.h"
#include "mpbswap.h"
#include "ad_internal.h"

static ad_info_t info = 
{
	"MS ADPCM audio decoder",
	"msadpcm",
	"Nick Kurshev",
	"Mike Melanson",
	""
};

LIBAD_EXTERN(msadpcm)

static int ms_adapt_table[] =
{
  230, 230, 230, 230, 307, 409, 512, 614,
  768, 614, 512, 409, 307, 230, 230, 230
};

static int ms_adapt_coeff1[] =
{
  256, 512, 0, 192, 240, 460, 392
};

static int ms_adapt_coeff2[] =
{
  0, -256, 0, 64, 0, -208, -232
};

#define MS_ADPCM_PREAMBLE_SIZE 6

#define LE_16(x) ((x)[0]+(256*((x)[1])))
//#define LE_16(x) (le2me_16((x)[1]+(256*((x)[0]))))
//#define LE_16(x) (le2me_16(*(unsigned short *)(x)))
//#define LE_32(x) (le2me_32(*(unsigned int *)(x)))

// useful macros
// clamp a number between 0 and 88
#define CLAMP_0_TO_88(x)  if (x < 0) x = 0; else if (x > 88) x = 88;
// clamp a number within a signed 16-bit range
#define CLAMP_S16(x)  if (x < -32768) x = -32768; \
  else if (x > 32767) x = 32767;
// clamp a number above 16
#define CLAMP_ABOVE_16(x)  if (x < 16) x = 16;
// sign extend a 16-bit value
#define SE_16BIT(x)  if (x & 0x8000) x -= 0x10000;
// sign extend a 4-bit value
#define SE_4BIT(x)  if (x & 0x8) x -= 0x10;

static int preinit(sh_audio_t *sh_audio)
{
  sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
  sh_audio->ds->ss_div = 
    (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
  sh_audio->audio_in_minsize =
  sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
  return 1;
}

static int init(sh_audio_t *sh_audio)
{
  sh_audio->channels=sh_audio->wf->nChannels;
  sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
  sh_audio->i_bps = sh_audio->wf->nBlockAlign *
    (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
  sh_audio->samplesize=2;

  return 1;
}

static void uninit(sh_audio_t *sh_audio)
{
}

static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
{
  if(cmd==ADCTRL_SKIP_FRAME){
    demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
    return CONTROL_TRUE;
  }
  return CONTROL_UNKNOWN;
}

static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
  int channels, int block_size)
{
  int current_channel = 0;
  int idelta[2];
  int sample1[2];
  int sample2[2];
  int coeff1[2];
  int coeff2[2];
  int stream_ptr = 0;
  int out_ptr = 0;
  int upper_nibble = 1;
  int nibble;
  int snibble;  // signed nibble
  int predictor;

  // fetch the header information, in stereo if both channels are present
  if (input[stream_ptr] > 6)
    mp_msg(MSGT_DECAUDIO, MSGL_WARN,
      "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
      input[stream_ptr]);
  coeff1[0] = ms_adapt_coeff1[input[stream_ptr]];
  coeff2[0] = ms_adapt_coeff2[input[stream_ptr]];
  stream_ptr++;
  if (channels == 2)
  {
    if (input[stream_ptr] > 6)
     mp_msg(MSGT_DECAUDIO, MSGL_WARN,
       "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
       input[stream_ptr]);
    coeff1[1] = ms_adapt_coeff1[input[stream_ptr]];
    coeff2[1] = ms_adapt_coeff2[input[stream_ptr]];
    stream_ptr++;
  }

  idelta[0] = LE_16(&input[stream_ptr]);
  stream_ptr += 2;
  SE_16BIT(idelta[0]);
  if (channels == 2)
  {
    idelta[1] = LE_16(&input[stream_ptr]);
    stream_ptr += 2;
    SE_16BIT(idelta[1]);
  }

  sample1[0] = LE_16(&input[stream_ptr]);
  stream_ptr += 2;
  SE_16BIT(sample1[0]);
  if (channels == 2)
  {
    sample1[1] = LE_16(&input[stream_ptr]);
    stream_ptr += 2;
    SE_16BIT(sample1[1]);
  }

  sample2[0] = LE_16(&input[stream_ptr]);
  stream_ptr += 2;
  SE_16BIT(sample2[0]);
  if (channels == 2)
  {
    sample2[1] = LE_16(&input[stream_ptr]);
    stream_ptr += 2;
    SE_16BIT(sample2[1]);
  }

  if (channels == 1)
  {
    output[out_ptr++] = sample2[0];
    output[out_ptr++] = sample1[0];
  } else {
    output[out_ptr++] = sample2[0];
    output[out_ptr++] = sample2[1];
    output[out_ptr++] = sample1[0];
    output[out_ptr++] = sample1[1];
  }

  while (stream_ptr < block_size)
  {
    // get the next nibble
    if (upper_nibble)
      nibble = snibble = input[stream_ptr] >> 4;
    else
      nibble = snibble = input[stream_ptr++] & 0x0F;
    upper_nibble ^= 1;
    SE_4BIT(snibble);

    predictor = (
      ((sample1[current_channel] * coeff1[current_channel]) +
       (sample2[current_channel] * coeff2[current_channel])) / 256) +
      (snibble * idelta[current_channel]);
    CLAMP_S16(predictor);
    sample2[current_channel] = sample1[current_channel];
    sample1[current_channel] = predictor;
    output[out_ptr++] = predictor;

    // compute the next adaptive scale factor (a.k.a. the variable idelta)
    idelta[current_channel] =
      (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
    CLAMP_ABOVE_16(idelta[current_channel]);

    // toggle the channel
    current_channel ^= channels - 1;
  }

  return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
}

static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
  if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
    sh_audio->ds->ss_mul) != 
    sh_audio->ds->ss_mul) 
      return -1; /* EOF */

  return 2 * ms_adpcm_decode_block(
    (unsigned short*)buf, sh_audio->a_in_buffer,
    sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
}