3 * Copyright (c) 2003 The ffmpeg Project
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * Assorted DPCM (differential pulse code modulation) audio codecs
25 * by Mike Melanson (melanson@pcisys.net)
26 * Xan DPCM decoder by Mario Brito (mbrito@student.dei.uc.pt)
27 * for more information on the specific data formats, visit:
28 * http://www.pcisys.net/~melanson/codecs/simpleaudio.html
29 * SOL DPCMs implemented by Konstantin Shishkov
31 * Note about using the Xan DPCM decoder: Xan DPCM is used in AVI files
32 * found in the Wing Commander IV computer game. These AVI files contain
33 * WAVEFORMAT headers which report the audio format as 0x01: raw PCM.
34 * Clearly incorrect. To detect Xan DPCM, you will probably have to
35 * special-case your AVI demuxer to use Xan DPCM if the file uses 'Xxan'
36 * (Xan video) for its video codec. Alternately, such AVI files also contain
37 * the fourcc 'Axan' in the 'auds' chunk of the AVI header.
40 #include "libavutil/intreadwrite.h"
42 #include "bytestream.h"
44 typedef struct DPCMContext {
46 int16_t roq_square_array[256];
47 int sample[2]; ///< previous sample (for SOL_DPCM)
48 const int8_t *sol_table; ///< delta table for SOL_DPCM
51 static const int16_t interplay_delta_table[] = {
52 0, 1, 2, 3, 4, 5, 6, 7,
53 8, 9, 10, 11, 12, 13, 14, 15,
54 16, 17, 18, 19, 20, 21, 22, 23,
55 24, 25, 26, 27, 28, 29, 30, 31,
56 32, 33, 34, 35, 36, 37, 38, 39,
57 40, 41, 42, 43, 47, 51, 56, 61,
58 66, 72, 79, 86, 94, 102, 112, 122,
59 133, 145, 158, 173, 189, 206, 225, 245,
60 267, 292, 318, 348, 379, 414, 452, 493,
61 538, 587, 640, 699, 763, 832, 908, 991,
62 1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
63 2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
64 4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
65 8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
66 17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
67 -29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
68 1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
69 29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
70 -17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
71 -8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
72 -4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
73 -2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
74 -1081, -991, -908, -832, -763, -699, -640, -587,
75 -538, -493, -452, -414, -379, -348, -318, -292,
76 -267, -245, -225, -206, -189, -173, -158, -145,
77 -133, -122, -112, -102, -94, -86, -79, -72,
78 -66, -61, -56, -51, -47, -43, -42, -41,
79 -40, -39, -38, -37, -36, -35, -34, -33,
80 -32, -31, -30, -29, -28, -27, -26, -25,
81 -24, -23, -22, -21, -20, -19, -18, -17,
82 -16, -15, -14, -13, -12, -11, -10, -9,
83 -8, -7, -6, -5, -4, -3, -2, -1
87 static const int8_t sol_table_old[16] = {
88 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
89 -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0
92 static const int8_t sol_table_new[16] = {
93 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15,
94 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15
97 static const int16_t sol_table_16[128] = {
98 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080,
99 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120,
100 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0,
101 0x1D0, 0x1E0, 0x1F0, 0x200, 0x208, 0x210, 0x218, 0x220, 0x228, 0x230,
102 0x238, 0x240, 0x248, 0x250, 0x258, 0x260, 0x268, 0x270, 0x278, 0x280,
103 0x288, 0x290, 0x298, 0x2A0, 0x2A8, 0x2B0, 0x2B8, 0x2C0, 0x2C8, 0x2D0,
104 0x2D8, 0x2E0, 0x2E8, 0x2F0, 0x2F8, 0x300, 0x308, 0x310, 0x318, 0x320,
105 0x328, 0x330, 0x338, 0x340, 0x348, 0x350, 0x358, 0x360, 0x368, 0x370,
106 0x378, 0x380, 0x388, 0x390, 0x398, 0x3A0, 0x3A8, 0x3B0, 0x3B8, 0x3C0,
107 0x3C8, 0x3D0, 0x3D8, 0x3E0, 0x3E8, 0x3F0, 0x3F8, 0x400, 0x440, 0x480,
108 0x4C0, 0x500, 0x540, 0x580, 0x5C0, 0x600, 0x640, 0x680, 0x6C0, 0x700,
109 0x740, 0x780, 0x7C0, 0x800, 0x900, 0xA00, 0xB00, 0xC00, 0xD00, 0xE00,
110 0xF00, 0x1000, 0x1400, 0x1800, 0x1C00, 0x2000, 0x3000, 0x4000
114 static av_cold int dpcm_decode_init(AVCodecContext *avctx)
116 DPCMContext *s = avctx->priv_data;
119 if (avctx->channels < 1 || avctx->channels > 2) {
120 av_log(avctx, AV_LOG_INFO, "invalid number of channels\n");
121 return AVERROR(EINVAL);
124 s->channels = avctx->channels;
125 s->sample[0] = s->sample[1] = 0;
127 switch(avctx->codec->id) {
129 case CODEC_ID_ROQ_DPCM:
130 /* initialize square table */
131 for (i = 0; i < 128; i++) {
132 int16_t square = i * i;
133 s->roq_square_array[i ] = square;
134 s->roq_square_array[i + 128] = -square;
138 case CODEC_ID_SOL_DPCM:
139 switch(avctx->codec_tag){
141 s->sol_table = sol_table_old;
142 s->sample[0] = s->sample[1] = 0x80;
145 s->sol_table = sol_table_new;
146 s->sample[0] = s->sample[1] = 0x80;
151 av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n");
160 if (avctx->codec->id == CODEC_ID_SOL_DPCM && avctx->codec_tag != 3)
161 avctx->sample_fmt = AV_SAMPLE_FMT_U8;
163 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
169 static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
172 const uint8_t *buf = avpkt->data;
173 int buf_size = avpkt->size;
174 const uint8_t *buf_end = buf + buf_size;
175 DPCMContext *s = avctx->priv_data;
179 int stereo = s->channels - 1;
180 int16_t *output_samples = data;
185 /* calculate output size */
186 switch(avctx->codec->id) {
187 case CODEC_ID_ROQ_DPCM:
190 case CODEC_ID_INTERPLAY_DPCM:
191 out = buf_size - 6 - s->channels;
193 case CODEC_ID_XAN_DPCM:
194 out = buf_size - 2 * s->channels;
196 case CODEC_ID_SOL_DPCM:
197 if (avctx->codec_tag != 3)
203 out *= av_get_bytes_per_sample(avctx->sample_fmt);
205 av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
206 return AVERROR(EINVAL);
208 if (*data_size < out) {
209 av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
210 return AVERROR(EINVAL);
213 switch(avctx->codec->id) {
215 case CODEC_ID_ROQ_DPCM:
219 predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8);
220 predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8);
222 predictor[0] = (int16_t)bytestream_get_le16(&buf);
225 /* decode the samples */
226 while (buf < buf_end) {
227 predictor[ch] += s->roq_square_array[*buf++];
228 predictor[ch] = av_clip_int16(predictor[ch]);
229 *output_samples++ = predictor[ch];
236 case CODEC_ID_INTERPLAY_DPCM:
237 buf += 6; /* skip over the stream mask and stream length */
239 for (ch = 0; ch < s->channels; ch++) {
240 predictor[ch] = (int16_t)bytestream_get_le16(&buf);
241 *output_samples++ = predictor[ch];
245 while (buf < buf_end) {
246 predictor[ch] += interplay_delta_table[*buf++];
247 predictor[ch] = av_clip_int16(predictor[ch]);
248 *output_samples++ = predictor[ch];
255 case CODEC_ID_XAN_DPCM:
257 int shift[2] = { 4, 4 };
259 for (ch = 0; ch < s->channels; ch++)
260 predictor[ch] = (int16_t)bytestream_get_le16(&buf);
263 while (buf < buf_end) {
265 int16_t diff = (n & 0xFC) << 8;
269 shift[ch] -= (2 * (n & 3));
270 /* saturate the shifter to a lower limit of 0 */
275 predictor[ch] += diff;
277 predictor[ch] = av_clip_int16(predictor[ch]);
278 *output_samples++ = predictor[ch];
285 case CODEC_ID_SOL_DPCM:
286 if (avctx->codec_tag != 3) {
287 uint8_t *output_samples_u8 = data;
288 while (buf < buf_end) {
291 s->sample[0] += s->sol_table[n >> 4];
292 s->sample[0] = av_clip_uint8(s->sample[0]);
293 *output_samples_u8++ = s->sample[0];
295 s->sample[stereo] += s->sol_table[n & 0x0F];
296 s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
297 *output_samples_u8++ = s->sample[stereo];
300 while (buf < buf_end) {
302 if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
303 else s->sample[ch] += sol_table_16[n & 0x7F];
304 s->sample[ch] = av_clip_int16(s->sample[ch]);
305 *output_samples++ = s->sample[ch];
317 #define DPCM_DECODER(id_, name_, long_name_) \
318 AVCodec ff_ ## name_ ## _decoder = { \
320 .type = AVMEDIA_TYPE_AUDIO, \
322 .priv_data_size = sizeof(DPCMContext), \
323 .init = dpcm_decode_init, \
324 .decode = dpcm_decode_frame, \
325 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
328 DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
329 DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
330 DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
331 DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");