view libmpcodecs/ad_msadpcm.c @ 7492:5465cbd5c5ef

Modern versions of OpenSSH listen on localhost to forward the X11 connection (X11UseLocalhost defaults to yes). The following patch permits to consider as non local any DISPLAY environment which port is greater or equal to 10 else mplayer tries a local optimization on a distant X server which doesn't work. patch by [e-mail]

/*
    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 "bswap.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) (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;

  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);
}