reworked ADPCM decoders; changes include:

 * fixed MS IMA ADPCM
 * dissolved adpcm.c/.h into appropriate ad_* decoders
 * DK4 audio is handled directly by IMA ADPCM decoder (this obsoletes
   ad_dk4adpcm.c)


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@5409 b3059339-0415-0410-9bf9-f77b7e298cf2

1 files changed, 143 insertions, 5 deletions

diff --git a/libmpcodecs/ad_msadpcm.c b/libmpcodecs/ad_msadpcm.c
index 8d1b5cebcd..4a56c4c1fc 100644
--- a/libmpcodecs/ad_msadpcm.c
+++ b/libmpcodecs/ad_msadpcm.c
@@ -1,8 +1,18 @@
+/*
+    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 = 
@@ -17,8 +27,40 @@ static ad_info_t info =
 
 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 7
 
+#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;
@@ -54,14 +96,110 @@ static int control(sh_audio_t *sh,int cmd,void* arg, ...)
   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]);
+  }
+
+  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 */
+    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);
+    (unsigned short*)buf, sh_audio->a_in_buffer,
+    sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);
 }