view libmpcodecs/ad_msadpcm.c @ 27815:f92271dc5f17

Remove X11 backing store: this is now a useless flag. Also, it is mandatory for Xserver 1.5.x (part of Xorg 7.4, shipped on all Linux distributions starting from Oct. 08) and will be removed from Xserver 1.6 anyhow ... Patch by Stephane Marchesin (marchesin at icps dot u dash strasbg dot fr). For more info, see long flame thread at: http://lists.mplayerhq.hu/pipermail/mplayer-dev-eng/2008-August/058323.html

/*
    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 "libavutil/intreadwrite.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 const int ms_adapt_table[] =
{
  230, 230, 230, 230, 307, 409, 512, 614,
  768, 614, 512, 409, 307, 230, 230, 230
};

static const uint8_t ms_adapt_coeff1[] =
{
  64, 128, 0, 48, 60, 115, 98
};

static const int8_t ms_adapt_coeff2[] =
{
  0, -64, 0, 16, 0, -52, -58
};

#define MS_ADPCM_PREAMBLE_SIZE 6

#define LE_16(x) ((int16_t)AV_RL16(x))

// clamp a number between 0 and 88
#define CLAMP_0_TO_88(x) x = av_clip(x, 0, 88);
// clamp a number within a signed 16-bit range
#define CLAMP_S16(x) x = av_clip_int16(x);
// clamp a number above 16
#define CLAMP_ABOVE_16(x)  if (x < 16) x = 16;
// 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 inline int check_coeff(uint8_t c) {
  if (c > 6) {
    mp_msg(MSGT_DECAUDIO, MSGL_WARN,
      "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
      c);
    c = 6;
  }
  return c;
}

static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
  int channels, int block_size)
{
  int current_channel = 0;
  int coeff_idx;
  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;

  if (channels != 1) channels = 2;
  if (block_size < 7 * channels)
    return -1;

  // fetch the header information, in stereo if both channels are present
  coeff_idx = check_coeff(input[stream_ptr]);
  coeff1[0] = ms_adapt_coeff1[coeff_idx];
  coeff2[0] = ms_adapt_coeff2[coeff_idx];
  stream_ptr++;
  if (channels == 2)
  {
    coeff_idx = check_coeff(input[stream_ptr]);
    coeff1[1] = ms_adapt_coeff1[coeff_idx];
    coeff2[1] = ms_adapt_coeff2[coeff_idx];
    stream_ptr++;
  }

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

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

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

  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);

    // should this really be a division and not a shift?
    // coefficients were originally scaled by for, which might have
    // been an optimization for 8-bit CPUs _if_ a shift is correct
    predictor = (
      ((sample1[current_channel] * coeff1[current_channel]) +
       (sample2[current_channel] * coeff2[current_channel])) / 64) +
      (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)
{
  int res;
  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 */

  res = ms_adpcm_decode_block(
    (unsigned short*)buf, sh_audio->a_in_buffer,
    sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
  return res < 0 ? res : 2 * res;
}