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[mplayer/glamo.git] / libmpcodecs / ad_msadpcm.c
blob6ef4bb35bbb37e4b94b022d4ad27600910664250
1 /*
2 MS ADPCM Decoder for MPlayer
3 by Mike Melanson
5 This file is responsible for decoding Microsoft ADPCM data.
6 Details about the data format can be found here:
7 http://www.pcisys.net/~melanson/codecs/
8 */
10 #include <stdio.h>
11 #include <stdlib.h>
12 #include <unistd.h>
14 #include "config.h"
15 #include "libavutil/common.h"
16 #include "mpbswap.h"
17 #include "ad_internal.h"
19 static ad_info_t info =
21 "MS ADPCM audio decoder",
22 "msadpcm",
23 "Nick Kurshev",
24 "Mike Melanson",
28 LIBAD_EXTERN(msadpcm)
30 static int ms_adapt_table[] =
32 230, 230, 230, 230, 307, 409, 512, 614,
33 768, 614, 512, 409, 307, 230, 230, 230
36 static int ms_adapt_coeff1[] =
38 256, 512, 0, 192, 240, 460, 392
41 static int ms_adapt_coeff2[] =
43 0, -256, 0, 64, 0, -208, -232
46 #define MS_ADPCM_PREAMBLE_SIZE 6
48 #define LE_16(x) ((x)[0]+(256*((x)[1])))
49 //#define LE_16(x) (le2me_16((x)[1]+(256*((x)[0]))))
50 //#define LE_16(x) (le2me_16(*(unsigned short *)(x)))
51 //#define LE_32(x) (le2me_32(*(unsigned int *)(x)))
53 // useful macros
54 // clamp a number between 0 and 88
55 #define CLAMP_0_TO_88(x) if (x < 0) x = 0; else if (x > 88) x = 88;
56 // clamp a number within a signed 16-bit range
57 #define CLAMP_S16(x) if (x < -32768) x = -32768; \
58 else if (x > 32767) x = 32767;
59 // clamp a number above 16
60 #define CLAMP_ABOVE_16(x) if (x < 16) x = 16;
61 // sign extend a 16-bit value
62 #define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
63 // sign extend a 4-bit value
64 #define SE_4BIT(x) if (x & 0x8) x -= 0x10;
66 static int preinit(sh_audio_t *sh_audio)
68 sh_audio->audio_out_minsize = sh_audio->wf->nBlockAlign * 4;
69 sh_audio->ds->ss_div =
70 (sh_audio->wf->nBlockAlign - MS_ADPCM_PREAMBLE_SIZE) * 2;
71 sh_audio->audio_in_minsize =
72 sh_audio->ds->ss_mul = sh_audio->wf->nBlockAlign;
73 return 1;
76 static int init(sh_audio_t *sh_audio)
78 sh_audio->channels=sh_audio->wf->nChannels;
79 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
80 sh_audio->i_bps = sh_audio->wf->nBlockAlign *
81 (sh_audio->channels*sh_audio->samplerate) / sh_audio->ds->ss_div;
82 sh_audio->samplesize=2;
84 return 1;
87 static void uninit(sh_audio_t *sh_audio)
91 static int control(sh_audio_t *sh_audio,int cmd,void* arg, ...)
93 if(cmd==ADCTRL_SKIP_FRAME){
94 demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,sh_audio->ds->ss_mul);
95 return CONTROL_TRUE;
97 return CONTROL_UNKNOWN;
100 static int ms_adpcm_decode_block(unsigned short *output, unsigned char *input,
101 int channels, int block_size)
103 int current_channel = 0;
104 int idelta[2];
105 int sample1[2];
106 int sample2[2];
107 int coeff1[2];
108 int coeff2[2];
109 int stream_ptr = 0;
110 int out_ptr = 0;
111 int upper_nibble = 1;
112 int nibble;
113 int snibble; // signed nibble
114 int predictor;
116 // fetch the header information, in stereo if both channels are present
117 if (input[stream_ptr] > 6)
118 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
119 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
120 input[stream_ptr]);
121 coeff1[0] = ms_adapt_coeff1[input[stream_ptr]];
122 coeff2[0] = ms_adapt_coeff2[input[stream_ptr]];
123 stream_ptr++;
124 if (channels == 2)
126 if (input[stream_ptr] > 6)
127 mp_msg(MSGT_DECAUDIO, MSGL_WARN,
128 "MS ADPCM: coefficient (%d) out of range (should be [0..6])\n",
129 input[stream_ptr]);
130 coeff1[1] = ms_adapt_coeff1[input[stream_ptr]];
131 coeff2[1] = ms_adapt_coeff2[input[stream_ptr]];
132 stream_ptr++;
135 idelta[0] = LE_16(&input[stream_ptr]);
136 stream_ptr += 2;
137 SE_16BIT(idelta[0]);
138 if (channels == 2)
140 idelta[1] = LE_16(&input[stream_ptr]);
141 stream_ptr += 2;
142 SE_16BIT(idelta[1]);
145 sample1[0] = LE_16(&input[stream_ptr]);
146 stream_ptr += 2;
147 SE_16BIT(sample1[0]);
148 if (channels == 2)
150 sample1[1] = LE_16(&input[stream_ptr]);
151 stream_ptr += 2;
152 SE_16BIT(sample1[1]);
155 sample2[0] = LE_16(&input[stream_ptr]);
156 stream_ptr += 2;
157 SE_16BIT(sample2[0]);
158 if (channels == 2)
160 sample2[1] = LE_16(&input[stream_ptr]);
161 stream_ptr += 2;
162 SE_16BIT(sample2[1]);
165 if (channels == 1)
167 output[out_ptr++] = sample2[0];
168 output[out_ptr++] = sample1[0];
169 } else {
170 output[out_ptr++] = sample2[0];
171 output[out_ptr++] = sample2[1];
172 output[out_ptr++] = sample1[0];
173 output[out_ptr++] = sample1[1];
176 while (stream_ptr < block_size)
178 // get the next nibble
179 if (upper_nibble)
180 nibble = snibble = input[stream_ptr] >> 4;
181 else
182 nibble = snibble = input[stream_ptr++] & 0x0F;
183 upper_nibble ^= 1;
184 SE_4BIT(snibble);
186 predictor = (
187 ((sample1[current_channel] * coeff1[current_channel]) +
188 (sample2[current_channel] * coeff2[current_channel])) / 256) +
189 (snibble * idelta[current_channel]);
190 CLAMP_S16(predictor);
191 sample2[current_channel] = sample1[current_channel];
192 sample1[current_channel] = predictor;
193 output[out_ptr++] = predictor;
195 // compute the next adaptive scale factor (a.k.a. the variable idelta)
196 idelta[current_channel] =
197 (ms_adapt_table[nibble] * idelta[current_channel]) / 256;
198 CLAMP_ABOVE_16(idelta[current_channel]);
200 // toggle the channel
201 current_channel ^= channels - 1;
204 return (block_size - (MS_ADPCM_PREAMBLE_SIZE * channels)) * 2;
207 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
209 if (demux_read_data(sh_audio->ds, sh_audio->a_in_buffer,
210 sh_audio->ds->ss_mul) !=
211 sh_audio->ds->ss_mul)
212 return -1; /* EOF */
214 return 2 * ms_adpcm_decode_block(
215 (unsigned short*)buf, sh_audio->a_in_buffer,
216 sh_audio->wf->nChannels, sh_audio->wf->nBlockAlign);