3 * Copyright (c) 2003 The ffmpeg Project
5 * This file is part of FFmpeg.
7 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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;
182 /* calculate output size */
183 switch(avctx->codec->id) {
184 case CODEC_ID_ROQ_DPCM:
187 case CODEC_ID_INTERPLAY_DPCM:
188 out = buf_size - 6 - s->channels;
190 case CODEC_ID_XAN_DPCM:
191 out = buf_size - 2 * s->channels;
193 case CODEC_ID_SOL_DPCM:
194 if (avctx->codec_tag != 3)
200 out *= av_get_bytes_per_sample(avctx->sample_fmt);
202 av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
203 return AVERROR(EINVAL);
205 if (*data_size < out) {
206 av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
207 return AVERROR(EINVAL);
210 switch(avctx->codec->id) {
212 case CODEC_ID_ROQ_DPCM:
216 predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8);
217 predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8);
219 predictor[0] = (int16_t)bytestream_get_le16(&buf);
222 /* decode the samples */
223 while (buf < buf_end) {
224 predictor[ch] += s->roq_square_array[*buf++];
225 predictor[ch] = av_clip_int16(predictor[ch]);
226 *output_samples++ = predictor[ch];
233 case CODEC_ID_INTERPLAY_DPCM:
234 buf += 6; /* skip over the stream mask and stream length */
236 for (ch = 0; ch < s->channels; ch++) {
237 predictor[ch] = (int16_t)bytestream_get_le16(&buf);
238 *output_samples++ = predictor[ch];
242 while (buf < buf_end) {
243 predictor[ch] += interplay_delta_table[*buf++];
244 predictor[ch] = av_clip_int16(predictor[ch]);
245 *output_samples++ = predictor[ch];
252 case CODEC_ID_XAN_DPCM:
254 int shift[2] = { 4, 4 };
256 for (ch = 0; ch < s->channels; ch++)
257 predictor[ch] = (int16_t)bytestream_get_le16(&buf);
260 while (buf < buf_end) {
262 int16_t diff = (n & 0xFC) << 8;
266 shift[ch] -= (2 * (n & 3));
267 /* saturate the shifter to a lower limit of 0 */
272 predictor[ch] += diff;
274 predictor[ch] = av_clip_int16(predictor[ch]);
275 *output_samples++ = predictor[ch];
282 case CODEC_ID_SOL_DPCM:
283 if (avctx->codec_tag != 3) {
284 uint8_t *output_samples_u8 = data;
285 while (buf < buf_end) {
288 s->sample[0] += s->sol_table[n >> 4];
289 s->sample[0] = av_clip_uint8(s->sample[0]);
290 *output_samples_u8++ = s->sample[0];
292 s->sample[stereo] += s->sol_table[n & 0x0F];
293 s->sample[stereo] = av_clip_uint8(s->sample[stereo]);
294 *output_samples_u8++ = s->sample[stereo];
297 while (buf < buf_end) {
299 if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F];
300 else s->sample[ch] += sol_table_16[n & 0x7F];
301 s->sample[ch] = av_clip_int16(s->sample[ch]);
302 *output_samples++ = s->sample[ch];
314 #define DPCM_DECODER(id_, name_, long_name_) \
315 AVCodec ff_ ## name_ ## _decoder = { \
317 .type = AVMEDIA_TYPE_AUDIO, \
319 .priv_data_size = sizeof(DPCMContext), \
320 .init = dpcm_decode_init, \
321 .decode = dpcm_decode_frame, \
322 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
325 DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay");
326 DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ");
327 DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol");
328 DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan");