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https://github.com/FFmpeg/FFmpeg.git
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3ef8be2bfc
Originally committed as revision 2195 to svn://svn.ffmpeg.org/ffmpeg/trunk
201 lines
6.7 KiB
C
201 lines
6.7 KiB
C
/*
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* Assorted DPCM codecs
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* Copyright (c) 2003 The ffmpeg Project.
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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/**
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* @file: dpcm.c
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* Assorted DPCM (differential pulse code modulation) audio codecs
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* by Mike Melanson (melanson@pcisys.net)
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* for more information on the specific data formats, visit:
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* http://www.pcisys.net/~melanson/codecs/simpleaudio.html
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*/
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#include "avcodec.h"
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typedef struct DPCMContext {
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int channels;
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short roq_square_array[256];
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int last_delta[2];
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} DPCMContext;
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#define SATURATE_S16(x) if (x < -32768) x = -32768; \
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else if (x > 32767) x = 32767;
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#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000;
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#define LE_16(x) ((((uint8_t*)(x))[1] << 8) | ((uint8_t*)(x))[0])
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#define LE_32(x) ((((uint8_t*)(x))[3] << 24) | \
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(((uint8_t*)(x))[2] << 16) | \
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(((uint8_t*)(x))[1] << 8) | \
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((uint8_t*)(x))[0])
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static int interplay_delta_table[] = {
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0, 1, 2, 3, 4, 5, 6, 7,
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8, 9, 10, 11, 12, 13, 14, 15,
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16, 17, 18, 19, 20, 21, 22, 23,
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24, 25, 26, 27, 28, 29, 30, 31,
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32, 33, 34, 35, 36, 37, 38, 39,
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40, 41, 42, 43, 47, 51, 56, 61,
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66, 72, 79, 86, 94, 102, 112, 122,
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133, 145, 158, 173, 189, 206, 225, 245,
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267, 292, 318, 348, 379, 414, 452, 493,
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538, 587, 640, 699, 763, 832, 908, 991,
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1081, 1180, 1288, 1405, 1534, 1673, 1826, 1993,
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2175, 2373, 2590, 2826, 3084, 3365, 3672, 4008,
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4373, 4772, 5208, 5683, 6202, 6767, 7385, 8059,
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8794, 9597, 10472, 11428, 12471, 13609, 14851, 16206,
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17685, 19298, 21060, 22981, 25078, 27367, 29864, 32589,
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-29973, -26728, -23186, -19322, -15105, -10503, -5481, -1,
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1, 1, 5481, 10503, 15105, 19322, 23186, 26728,
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29973, -32589, -29864, -27367, -25078, -22981, -21060, -19298,
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-17685, -16206, -14851, -13609, -12471, -11428, -10472, -9597,
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-8794, -8059, -7385, -6767, -6202, -5683, -5208, -4772,
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-4373, -4008, -3672, -3365, -3084, -2826, -2590, -2373,
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-2175, -1993, -1826, -1673, -1534, -1405, -1288, -1180,
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-1081, -991, -908, -832, -763, -699, -640, -587,
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-538, -493, -452, -414, -379, -348, -318, -292,
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-267, -245, -225, -206, -189, -173, -158, -145,
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-133, -122, -112, -102, -94, -86, -79, -72,
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-66, -61, -56, -51, -47, -43, -42, -41,
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-40, -39, -38, -37, -36, -35, -34, -33,
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-32, -31, -30, -29, -28, -27, -26, -25,
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-24, -23, -22, -21, -20, -19, -18, -17,
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-16, -15, -14, -13, -12, -11, -10, -9,
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-8, -7, -6, -5, -4, -3, -2, -1
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};
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static int dpcm_decode_init(AVCodecContext *avctx)
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{
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DPCMContext *s = avctx->priv_data;
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int i;
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short square;
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s->channels = avctx->channels;
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switch(avctx->codec->id) {
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case CODEC_ID_ROQ_DPCM:
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/* initialize square table */
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for (i = 0; i < 128; i++) {
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square = i * i;
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s->roq_square_array[i] = square;
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s->roq_square_array[i + 128] = -square;
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}
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break;
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default:
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break;
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}
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return 0;
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}
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static int dpcm_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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uint8_t *buf, int buf_size)
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{
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DPCMContext *s = avctx->priv_data;
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int in, out = 0;
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int i;
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int predictor[2];
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int channel_number = 0;
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short *output_samples = data;
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int sequence_number;
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switch(avctx->codec->id) {
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case CODEC_ID_ROQ_DPCM:
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if (s->channels == 1)
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predictor[0] = LE_16(&buf[6]);
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else {
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predictor[0] = buf[7] << 8;
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predictor[1] = buf[6] << 8;
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}
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SE_16BIT(predictor[0]);
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SE_16BIT(predictor[1]);
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/* decode the samples */
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for (in = 8, out = 0; in < buf_size; in++, out++) {
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predictor[channel_number] += s->roq_square_array[buf[in]];
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SATURATE_S16(predictor[channel_number]);
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output_samples[out] = predictor[channel_number];
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/* toggle channel */
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channel_number ^= s->channels - 1;
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}
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break;
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case CODEC_ID_INTERPLAY_DPCM:
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in = 0;
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sequence_number = LE_16(&buf[in]);
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in += 6; /* skip over the stream mask and stream length */
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if (sequence_number == 1) {
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predictor[0] = LE_16(&buf[in]);
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in += 2;
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SE_16BIT(predictor[0])
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if (s->channels == 2) {
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predictor[1] = LE_16(&buf[in]);
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SE_16BIT(predictor[1])
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in += 2;
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}
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} else {
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for (i = 0; i < s->channels; i++)
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predictor[i] = s->last_delta[i];
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}
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while (in < buf_size) {
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predictor[channel_number] += interplay_delta_table[buf[in++]];
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SATURATE_S16(predictor[channel_number]);
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output_samples[out++] = predictor[channel_number];
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/* toggle channel */
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channel_number ^= s->channels - 1;
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}
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/* save predictors for next round */
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for (i = 0; i < s->channels; i++)
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s->last_delta[i] = predictor[i];
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break;
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}
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*data_size = out * sizeof(short);
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return buf_size;
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}
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AVCodec roq_dpcm_decoder = {
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"roq_dpcm",
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CODEC_TYPE_AUDIO,
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CODEC_ID_ROQ_DPCM,
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sizeof(DPCMContext),
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dpcm_decode_init,
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NULL,
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NULL,
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dpcm_decode_frame,
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};
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AVCodec interplay_dpcm_decoder = {
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"interplay_dpcm",
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CODEC_TYPE_AUDIO,
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CODEC_ID_INTERPLAY_DPCM,
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sizeof(DPCMContext),
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dpcm_decode_init,
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NULL,
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NULL,
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dpcm_decode_frame,
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};
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